LinuxMCE Forums

General => Users => Topic started by: microbrain on October 14, 2012, 07:48:42 pm

Title: FreePbx
Post by: microbrain on October 14, 2012, 07:48:42 pm
I downloaded the stable addition of 8.10 and installed it on a test box to try and get a grip on an actual working LMCE. In playing with it I noticed that FreePBX was a part of that version. I know that I can download and install FreePBX but I'm not sure if the current version of FreePBX is what works with 8.10 or 10.04-26551. Is there a script, add-on a known way that I can install FreePBX on 26551 of 10.04 and where would I need to change the link in LMCE admin page to get to FreePBX once I install it? Maybe someone has install it on 10.04 and would like to post how they did it. I read in some other posts that it was removed due to people not familiar with FreePBX making changes that caused breaking LMCE.

I would like to take advantage of additional options within the Asterisk PBX that are not accessible within LMCE.

Thanks

microbrain
Title: Re: FreePbx
Post by: maverick0815 on October 14, 2012, 09:23:31 pm
As far as I know freepbx is no longer part of 10.04 and can't be used anymore. Asterisk is beeing configured via the database.
Title: Re: FreePbx
Post by: posde on October 14, 2012, 10:05:37 pm
maverick is correct. We no longer use freepbx. We have our own web frontend which manages phone lines, phones, routing and stuff. If there is anything missing from our web frontend, let us know.
Title: Re: FreePbx
Post by: pw44 on October 16, 2012, 09:14:57 pm
How about a front end takes looks like freepbx, with sections divided, would make live much easier..... there are too many variables and a very simple interface, or better, a no interface make it all a PITA. I'm fighting with it for some weeks, and no results. No even ONE trunk is working, and a had THREE working with 8.10, all configured with freepbx and working like a charm. So, IMHO, it may be very good having asterisk with realtime database, but there is almost no documentation about, and worst, i could not find anyone using it to get a help and discuss how to make my old configs work with this combo.
Title: Re: FreePbx
Post by: posde on October 16, 2012, 10:26:50 pm
pw44,

there are people who just use the web admin, fill out the forms, and have multiple lines working. If you have problems with the web frontend, please detail them.
Title: Re: FreePbx
Post by: pw44 on October 17, 2012, 01:23:49 am
Posde,
1 - Linksys SPA-3102 - no way to make it register, and so, no landline.
2 - voipcheap -  needed to add SIP_ANON yes - calling my DID number from my mobile, my home phone rings, outgoing sound works (i hear on the mobile), incoming voice (from mobile) cannot be heared on the home phone. Calling from home to outside, it takes almost 5 minutes to calling phone (my mobile) to ring, and no sound in or out.
3 - sipgate.de - does not register. Nothing happens.
All three trunks were working on the 8.10 release. Hardware is the same: core/hybrid, cisco phone, linksys spa-3102, router, adsl modem, all exactly the same. The only change was from 8.10 to 10.04.
So, i'm digging for the last 40 days for a solution, but no way to make it work. Too less documentation about. I'm out of knowledge to find out what is wrong, as all worked before.
TIA,
Paulo
Title: Re: FreePbx
Post by: microbrain on October 17, 2012, 08:36:11 am

From what I am observing with LMCE 10.04 and phone line/phones set-up I can see where pw44 is justly frustrated beyond belief.

As it is now if you happen to have a Grandstream or Snome phone that does an FTP boot then setting up a plug and play sip phone probably works well for the common user. But, for someone like pw44 or myself whom has a ATA setting up this box to work can be a hassle unless you know the inner workings (code) of Asterisk.

There are some VoIp providers that you just have to go in and modify certain details of the trunk set-up and by having something like a FreePBX front-end to Asterisk makes this a breeze. Things like Outbound CID, maximum channels and dial rules, user details and peer details not to mention the registration string. In addition, LMCE now does not allow for modifications of certain details of an extension when setting it up. Such as what codecs to allow or deny, what Ip's allow/deny, or if it is a SIP, IAX2, DAHDI, ZAP or Custom device or nat settings.. And, for those of us that are knowledgeable, setting up IVR and voice mail options that meet our needs for around the home.

What do you do in the case of a person who has their own landline, wants to install an FXO/FXS card into the LMCE box and use sip phones through out the house? Just asking for personal knowledge.

I would think, IMHO, that a downloadable add-on GUI like FreePBX or something similar would have been the way to go that would allow those of us knowledgeable a way to set-up Asterisk to meet our needs, or, just setting LMCE code to connect to an external Asterisk Box. Asterisk is like LMCE, it's free and setting up a separate machine is less costly then setting up  a LMCE machine and probably would have made LMCE coders a lot happier not having to mess with the Asterisk portion.

For pw44, he has what sounds like three issues. One, the trunk detail to his SIP provider and two possibly a NAT issue (one way audio is normally caused by NAT issues), and three - as  far as the SPA-3102, if the parameters within it are set properly it should register with the LMCE if not then he needs to check its parameters. For the SPA-3102 he can determine what's going on by running sip debug command on a command line entry on the main server and watch what is going on when it tries to register.

Since I haven't learned all there is to LMCE, the question I have again is is it possible to download FreePBX and install it without totally destroying LMCE? Has anyone tried to do it - is it possible? If not then I guess those of us that need the ability to have Asterisk do what we wish and keep it part of the LMCE idea then people like me will be forever stuck with LMCE 8.10 unless something else comes along.

Asterisk has a lot to offer even the home user or better still the SMB. Not taking advantage of its potential I think leaves LMCE limited in its potential, again just my opinion.

microbrain

Title: Re: FreePbx
Post by: posde on October 17, 2012, 09:49:07 am
pw44,

if your three phonelines did work before out of the box, and they no longer work out of the box, please open a bug ticket.

if your three phonelines did not work before out of the box, and you want them to work again in 1004, please open a feature patch ticket, detailing what you had to do in 810 to get them to work.
Title: Re: FreePbx
Post by: pw44 on October 17, 2012, 12:09:45 pm
So, before beginning to blame the new asterisk combo, i inserted the following in the database:
83    0    18    0    sip.conf    general    alwaysauthreject    yes
85    0    18    0    sip.conf    general    nat                            yes
86    0    60    0    sip.conf    general    externhost            mydyndns.homeunix.org
87    0    5    0    sip.conf    general    externrefresh            5
88    0    60    0    sip.conf    general    localnet                   192.168.80.0/255.255.255.0
89    0    9    0    sip.conf    general    allow                    g729
90    0    10    0    sip.conf    general    allow                    g723
91    0    101    0    sip.conf    general    register                    pwollny:XXXXXXXXX@sip.voipcheap.com/2062036594
Also did set SIP_ANON yes.
As as i told before: All three trunks were working, with trunk selection by dialplan, without glitches.
I'm not an asterisk guru, did research a lot in order to make it work, to understand how to build the dialpan and to have all working, and now all i learned is worth zero.
I don't know if it's a bug, if it's a configuration issue, i'm not finding out how to proceed with config. Is there any at least medium documentation where i can find help? I could not find any.
That's all.
Title: Re: FreePbx
Post by: posde on October 17, 2012, 03:30:40 pm
pw44,

could you answer the implied questions of my previous post, please :)
Title: Re: FreePbx
Post by: pw44 on October 17, 2012, 05:54:49 pm
Answering the question: the three lines worked on my 8.10 release.
I will open a bug ticket.
Thx.
Title: Re: FreePbx
Post by: posde on October 17, 2012, 07:43:31 pm
Did the work ootb or did they work with manual fiddling?
Title: Re: FreePbx
Post by: pw44 on October 17, 2012, 10:27:37 pm
On 8.10 nothing worked for me ootb, all trunks needed to be feeded, because sipgate, voipcheap and spa-3102 did not had the amp_create****, and later i did create amp_create_sipgate and amp_create_voipcheap (and created wiki for it). spa-3102 was manually created, so as the dialplans.
Title: Re: FreePbx
Post by: posde on October 17, 2012, 10:31:37 pm
pw44,

than I would not call this a bug, but a feature request, please. Could you detail what settings were needed to get the phone lines working, please? And if possible, do one for each, that way foxy, or whoever is going to look at it, can tackle one at a time, and put things forward. A working sip.conf for each of the phone lines would be a good thing to attach to the tickets.
Title: Re: FreePbx
Post by: pw44 on October 18, 2012, 01:31:50 am
Posde,
thx for answering:

I would not say it's a feature request, because is expected that any user would be able to have at least one trunk working with the sip provider of choice :). Ok, not all sip providers are supported by the devel group, so, some documentation should be provided, and making the trunk work would add new providers. I did it with sipgate and voipcheap, but i had something to research and digg. spa3102 was done by Seth.
 My 8.10 working config.
Trunks: SPA-3102 - the spa config is the same as found in: http://wiki.linuxmce.org/index.php/Linksys_SPA3102
            Voipcheap wiki: http://wiki.linuxmce.org/index.php/VoIP_with_voipscheap.com
            sipgate.de wiki: i remember that i created it, but it's not there :(

Well, to the asterisk confi files. If the freepbx version is needed, please let me know.

The sip.conf from the working config:
Code: [Select]
[general]
#include sip_general_additional.conf

bindport = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
alwaysauthreject=yes ; required by fail2ban
disallow=all
allow=ulaw
allow=alaw
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; Reported as required for Asterisk 1.4
notifyringing=yes
notifyhold=yes
limitonpeers=yes

; enable and force the sip jitterbuffer. If these settings are desired
; they should be set in the sip_general_custom.conf file as this file
; will get overwritten during reloads and upgrades.
;
; jbenable=yes
; jbforce=yes

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_general_custom.conf
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
#include sip_custom.conf
#include sip_additional.conf


sip_additional.conf
Code: [Select]
[sip.voipcheap.com]
type=friend
qualify=yes
insecure=invite,port
host=sip.voipcheap.com
dtmfmode=auto
disallow=all
context=from-pstn
allow=ulaw
allow=alaw
allow=g729

[sipgate]
username=username
type=peer
secret=xxxxxxxxxxxxxxxxxxx
qualify=yes
port=5060
nat=yes
insecure=invite,port
host=sipgate.de
fromuser=username
fromdomain=sipgate.de
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=yes
authuser=username
allow=alaw
allow=ulaw
allow=g726
allow=gsm
allow=g729
call-limit=50

[sipgate_de]
username=username
type=friend
secret=xxxxxxxxxxxxxxxxxx
qualify=yes
port=5060
insecure=invite,port
host=sipgate.de
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=yes
allow=alaw
allow=ulaw
allow=g726
allow=gsm
allow=g729

[spa3102]
username=spa3102
type=friend
secret=lmce
qualify=yes
port=5061
nat=never
incominglimit=1
host=dynamic
dtmfmode=auto
context=from-trunk
canreinvite=no
allow=ulaw
call-limit=50

[voipcheap]
username=username
type=friend
sendrpid=yes
secret=xxxxxxxxxxxxxxx
qualify=yes
port=5060
nat=yes
insecure=invite,port
host=sip.voipcheap.com
fromuser=username
fromdomain=sip.voipcheap.com
dtmfmode=auto
disallow=all
context=from-pstn
canreinvite=yes
authuser=username
allow=ulaw
allow=ulaw
allow=g729
call-limit=50

sip_registrations.conf
Code: [Select]
register=usname:xxxxxxxxxx@sip.voipcheap.com/2062036594
register=username:xxxxxxxxxx@sipgate.de/054138594676

sip_nat.conf
Code: [Select]
nat=yes
externip=myhostdyndns.homeunix.org
externrefresh=10
localnet=192.168.80.0/255.255.255.0

localprefixes.conf
Code: [Select]
[trunk-4]
rule1=00+XXXXXXX.

[trunk-2]
rule1=XXXXXXXX
rule2=08+08|00XXXXX.
rule3=005521|XXXXXXXX
rule4=031+0055|XXXXXXXXXX
rule5=031+0|XXXXXXXXXX
rule6=031+XXXXXXXXXX
rule7=031+011XXXXXXXXX

[trunk-3]
rule1=00+XXXXXXX.


If there is any additional configuration file you need, please let me know.

TIA,

Paulo

Title: Re: FreePbx
Post by: pw44 on October 18, 2012, 01:35:28 am
For pw44, he has what sounds like three issues. One, the trunk detail to his SIP provider and two possibly a NAT issue (one way audio is normally caused by NAT issues), and three - as  far as the SPA-3102, if the parameters within it are set properly it should register with the LMCE if not then he needs to check its parameters. For the SPA-3102 he can determine what's going on by running sip debug command on a command line entry on the main server and watch what is going on when it tries to register.

Sorry to disagree, all the spa3102 configs are the same that were working on 8.10. spa3102 was not touched. And sip debug shows that it does not register in asterisk 1.8.11.
Thx!
Title: Re: FreePbx
Post by: microbrain on October 18, 2012, 02:19:34 am
Pw44,
Before you do anything, turn the firewall off and try all your registrations and see if they work. The process of elimination so to speak. Then

Do a "sip set debug on" then do a "sip show peers". Post the results here or send to my email I will try to see what's happening with SPA.

Also, I see that you have listed sip.conf and sip_additional_conf. Have you actually modified these files and added what you needed on the new machine with 10.04? On newer versions of * sip.conf and others like it are rewritten when you reload *, older versions didn't do that. Check to make sure that your sip.con doesn't say at the beginning like "to modify another conf file as this file is overwritten when reloading * . I will read through you conf files posted tonight when I get free and see if I can tell what's going on.

I am planning tonight on seeing if it is just possible to set LMCE up as a client to a external * machine and be done with trying to make * in LMCE work.

microbrain
Title: Re: FreePbx
Post by: pw44 on October 18, 2012, 03:07:52 am
The spa3102 in located in the 192.168.80.0 network, no firewall, no nat, nothing. I do have firewall in the 192.168.0.0, which is the external network....
Spa3102 simple does not register. I will try to put it again to work and will debug all, from spa3102 and asterisk, and post the results.
The problem with one way voice is with voipcheap trunk - for this one i will disable firewall and see what happens.
Sipgate i did not give i try, because i don't want to add noise to what is not working.
Regarding the files, they are from the 8.10 release, and i sent to see how to make all this work with asterisk realtime in 10.04.
Title: Re: FreePbx
Post by: sambuca on October 18, 2012, 08:48:37 am
I use the SPA3102 in 1004 as a phone line (PSTN -> IP adapter). I configured it using the phone lines page in the web admin. You have to select the SPA protocol. What took the most work was to set up the SPA itself.

And for the provider setups, if the setups had been integrated into LinuxMCE, they would probably have been considered when changing the Asterisk setup for 1004. Because they were only in a wiki page, they most probably were not.

br,
sambuca
Title: Re: FreePbx
Post by: posde on October 18, 2012, 10:01:00 am
Paulo,

to make things easier for everybody, please put the needed information into trac tickets. Forum posts get lost/ignored/whatever. A trac ticket that keeps all the information (and not links to other places) can easily be worked with.

Thanks.
Title: Re: FreePbx
Post by: pw44 on October 18, 2012, 09:39:44 pm
I use the SPA3102 in 1004 as a phone line (PSTN -> IP adapter). I configured it using the phone lines page in the web admin. You have to select the SPA protocol. What took the most work was to set up the SPA itself.

And for the provider setups, if the setups had been integrated into LinuxMCE, they would probably have been considered when changing the Asterisk setup for 1004. Because they were only in a wiki page, they most probably were not.

br,
sambuca
The one in the mentioned wiki is the right one to make spa3102 acts like a trunk for pstn lines. It's all i need. And worked on 8.10.
On 10.04, defining as SPA, actually it will use SIP in the config databank and should use udp port 5061 instead of 5060,
Title: Re: FreePbx
Post by: pw44 on October 18, 2012, 09:41:50 pm
Paulo,

to make things easier for everybody, please put the needed information into trac tickets. Forum posts get lost/ignored/whatever. A trac ticket that keeps all the information (and not links to other places) can easily be worked with.

Thanks.

All in one ticket or one ticket for each trunk?
Title: Re: FreePbx
Post by: posde on October 18, 2012, 10:10:57 pm
What ever you think would make most sense to the person who is going to look at the tickets. I personally would make three tickets, but that's just because I love closing tickets
Title: Re: FreePbx
Post by: pw44 on October 19, 2012, 02:36:24 am
Ticket created. If something is needed, please inform and i will provide.
TIA.
Title: Re: FreePbx
Post by: davegravy on October 19, 2012, 07:32:59 am
I personally would make three tickets, but that's just because I love closing tickets
/me creates millions of empty tickets for possy.
Title: Re: FreePbx
Post by: posde on October 19, 2012, 01:31:35 pm
Ticket created. If something is needed, please inform and i will provide.

Just revisit the ticket every now and then, to see if there is an update. Thanks again.
Title: Re: FreePbx
Post by: mcefan on October 20, 2012, 09:54:36 am
Please post a link to the ticket or wherever the discussions move so those who are interested can follow along.
Thank you
Title: Re: FreePbx
Post by: pw44 on October 20, 2012, 05:16:22 pm
http://svn.linuxmce.org/trac.cgi/ticket/1595
Title: Re: FreePbx
Post by: mcefan on October 20, 2012, 06:19:48 pm
Thank you!
Please make it SOP.
Title: Re: FreePbx
Post by: mcefan on October 21, 2012, 07:56:48 am
http://svn.linuxmce.org/trac.cgi/ticket/1595

I looked at the post, I did not see any reference to this thread. It might be useful to do so, in case someone starts from there, as a way of introduction. It helps understand the issue better.
Also, please complete this thread with updates when you have them.

Thank you.
Title: Re: FreePbx
Post by: posde on October 21, 2012, 10:38:23 am
mcefan, if you find it useful to add a link to this thread, why didn't you
Title: Re: FreePbx
Post by: pw44 on October 21, 2012, 04:49:32 pm
Pw44,
Before you do anything, turn the firewall off and try all your registrations and see if they work. The process of elimination so to speak. Then

Do a "sip set debug on" then do a "sip show peers". Post the results here or send to my email I will try to see what's happening with SPA.

Also, I see that you have listed sip.conf and sip_additional_conf. Have you actually modified these files and added what you needed on the new machine with 10.04? On newer versions of * sip.conf and others like it are rewritten when you reload *, older versions didn't do that. Check to make sure that your sip.con doesn't say at the beginning like "to modify another conf file as this file is overwritten when reloading * . I will read through you conf files posted tonight when I get free and see if I can tell what's going on.

I am planning tonight on seeing if it is just possible to set LMCE up as a client to a external * machine and be done with trying to make * in LMCE work.

microbrain

Hi Microbrain,
thx for yor offer in analyze the debug.
As told, i do have: 2 sccp devices (cisco 7970), extensions 201 and 203, two media directors, extensions 200 and 202 and a spa-3102, where line 1 is set as extension 204. spa-3102 is connected in the inside network (192.168.80.0), with not access from the external network. Is only a pstn to voip gateway to enable the use of the old pstn.
As sip trunks, i have voipcheap, but sound only one way, and the pstn line from spa-3102 does not registers on asterisk.
I did enable nat, and the needed parameters at the database tables, disabled the firewall, but no way to have it working.
sip set debug on gave the following results, and i hope that some can see what i'm not being able to.
Best regards and thx again.
Paulo

Code: [Select]
dcerouter*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
[2012-10-21 12:31:35] NOTICE[13062]: chan_sip.c:13059 sip_reregister:    -- Re-registration for  pwollny@sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK14bcecd4;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as4bbc681b
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1580 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2083991781", response="ecf5a4df5c2bfe63effba1a4d47aca3f"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK14bcecd4;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as4bbc681b
To: <sip:pwollny@sip.voipcheap.com>
Contact: sip:77.72.169.134:5060
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1580 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm="sip.voipcheap.com",nonce="2084097281",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4ae0babf;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as7af43c56
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1581 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084097281", response="3a495d3194a17245a7c7a27c1715cf7b"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4ae0babf;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as7af43c56
To: <sip:pwollny@sip.voipcheap.com>
Contact: <sip:2062036594@192.168.80.1:5060>;expires=120
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1581 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' in 32000 ms (Method: REGISTER)
[2012-10-21 12:31:35] NOTICE[13062]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for sip.voipcheap.com is 120 sec (Scheduling reregistration in 105 s)
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->

<--- SIP read from UDP:192.168.80.30:5060 --->
REGISTER sip:192.168.80.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.30:5060;branch=z9hG4bK-67afa790
From: Line 1 <sip:204@192.168.80.1>;tag=aa9309622215e998o0
To: Line 1 <sip:204@192.168.80.1>
Call-ID: 73534f17-489cd6b0@192.168.80.30
CSeq: 52204 REGISTER
Max-Forwards: 70
Contact: Line 1 <sip:204@192.168.80.30:5060>;expires=600
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.80.30:5060 (NAT)

<--- Transmitting (NAT) to 192.168.80.30:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.30:5060;branch=z9hG4bK-67afa790;received=192.168.80.30;rport=5060
From: Line 1 <sip:204@192.168.80.1>;tag=aa9309622215e998o0
To: Line 1 <sip:204@192.168.80.1>;tag=as45e5cbbc
Call-ID: 73534f17-489cd6b0@192.168.80.30
CSeq: 52204 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="532a4994"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '73534f17-489cd6b0@192.168.80.30' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.30:5061 --->
REGISTER sip:192.168.80.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.30:5061;branch=z9hG4bK-b84734c9
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=daea7b1299b5a3a8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>
Call-ID: 59e95fec-ea4373df@192.168.80.30
CSeq: 47241 REGISTER
Max-Forwards: 70
Contact: Unknown Caller <sip:spa3102@192.168.80.30:5061>;expires=300
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.80.30:5061 (NAT)

<--- Transmitting (NAT) to 192.168.80.30:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.30:5061;branch=z9hG4bK-b84734c9;received=192.168.80.30;rport=5061
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=daea7b1299b5a3a8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>;tag=as00018c3a
Call-ID: 59e95fec-ea4373df@192.168.80.30
CSeq: 47241 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="77ef4f55"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '59e95fec-ea4373df@192.168.80.30' in 32000 ms (Method: REGISTER)
[2012-10-21 12:31:53] NOTICE[13062]: chan_sip.c:24933 handle_request_register: Registration from 'Unknown Caller <sip:spa3102@192.168.80.1>' failed for '192.168.80.30:5061' - No matching peer found
Scheduling destruction of SIP dialog '59e95fec-ea4373df@192.168.80.30' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.30:5060 --->
REGISTER sip:192.168.80.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.30:5060;branch=z9hG4bK-a2c02b85
From: Line 1 <sip:204@192.168.80.1>;tag=aa9309622215e998o0
To: Line 1 <sip:204@192.168.80.1>
Call-ID: 73534f17-489cd6b0@192.168.80.30
CSeq: 52205 REGISTER
Max-Forwards: 70
Authorization: Digest username="204",realm="asterisk",nonce="532a4994",uri="sip:192.168.80.1",algorithm=MD5,response="aa826026d9f08657d89505d667fdd596"
Contact: Line 1 <sip:204@192.168.80.30:5060>;expires=600
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.80.30:5060 (NAT)
    -- Registered SIP '204' at 192.168.80.30:5060

<--- Transmitting (NAT) to 192.168.80.30:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.80.30:5060;branch=z9hG4bK-a2c02b85;received=192.168.80.30;rport=5060
From: Line 1 <sip:204@192.168.80.1>;tag=aa9309622215e998o0
To: Line 1 <sip:204@192.168.80.1>;tag=as45e5cbbc
Call-ID: 73534f17-489cd6b0@192.168.80.30
CSeq: 52205 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 600
Contact: <sip:204@192.168.80.30:5060>;expires=600
Date: Sun, 21 Oct 2012 14:31:53 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '73534f17-489cd6b0@192.168.80.30' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.30:5061 --->
REGISTER sip:192.168.80.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.30:5061;branch=z9hG4bK-359ccda2
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=daea7b1299b5a3a8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>
Call-ID: 59e95fec-ea4373df@192.168.80.30
CSeq: 47242 REGISTER
Max-Forwards: 70
Authorization: Digest username="spa3102",realm="asterisk",nonce="77ef4f55",uri="sip:192.168.80.1",algorithm=MD5,response="01a7d37fe11ebbe27edc873f8e69200e"
Contact: Unknown Caller <sip:spa3102@192.168.80.30:5061>;expires=300
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.80.30:5061 (NAT)

<--- Transmitting (NAT) to 192.168.80.30:5061 --->
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 192.168.80.30:5061;branch=z9hG4bK-359ccda2;received=192.168.80.30;rport=5061
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=daea7b1299b5a3a8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>;tag=as00018c3a
Call-ID: 59e95fec-ea4373df@192.168.80.30
CSeq: 47242 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2012-10-21 12:31:53] NOTICE[13062]: chan_sip.c:24933 handle_request_register: Registration from 'Unknown Caller <sip:spa3102@192.168.80.1>' failed for '192.168.80.30:5061' - No matching peer found
Scheduling destruction of SIP dialog '59e95fec-ea4373df@192.168.80.30' in 32000 ms (Method: REGISTER)
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
Really destroying SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' Method: REGISTER

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
Really destroying SIP dialog '73534f17-489cd6b0@192.168.80.30' Method: REGISTER
Really destroying SIP dialog '59e95fec-ea4373df@192.168.80.30' Method: REGISTER

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
REGISTER sip:dcerouter SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5061;rport;branch=z9hG4bK538045933
From: <sip:200@dcerouter>;tag=1389717865
To: <sip:200@dcerouter>
Call-ID: 1582944178
CSeq: 300 REGISTER
Contact: <sip:200@192.168.80.1:5061;line=8b7c960eff2e4e2>
Authorization: Digest username="200", realm="asterisk", nonce="4dfdcb82", uri="sip:dcerouter", response="45322023e5915e9823507c25531e1e80", algorithm=MD5
Max-Forwards: 70
User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
Expires: 600
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.80.1:5061 (NAT)

<--- Transmitting (NAT) to 192.168.80.1:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.160:5061;branch=z9hG4bK538045933;received=192.168.80.1;rport=5061
From: <sip:200@dcerouter>;tag=1389717865
To: <sip:200@dcerouter>;tag=as3f131c59
Call-ID: 1582944178
CSeq: 300 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05b0c029"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1582944178' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.1:5061 --->
REGISTER sip:dcerouter SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5061;rport;branch=z9hG4bK372606350
From: <sip:200@dcerouter>;tag=1389717865
To: <sip:200@dcerouter>
Call-ID: 1582944178
CSeq: 301 REGISTER
Contact: <sip:200@192.168.80.1:5061;line=8b7c960eff2e4e2>
Authorization: Digest username="200", realm="asterisk", nonce="05b0c029", uri="sip:dcerouter", response="e707cd6421b91229218a42c6b76b6235", algorithm=MD5
Max-Forwards: 70
User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
Expires: 600
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.80.1:5061 (NAT)

<--- Transmitting (NAT) to 192.168.80.1:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.160:5061;branch=z9hG4bK372606350;received=192.168.80.1;rport=5061
From: <sip:200@dcerouter>;tag=1389717865
To: <sip:200@dcerouter>;tag=as3f131c59
Call-ID: 1582944178
CSeq: 301 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 600
Contact: <sip:200@192.168.80.1:5061;line=8b7c960eff2e4e2>;expires=600
Date: Sun, 21 Oct 2012 14:32:56 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1582944178' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
[2012-10-21 12:33:20] NOTICE[13062]: chan_sip.c:13059 sip_reregister:    -- Re-registration for  pwollny@sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6c40f5fe;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as08137794
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1582 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084097281", response="3a495d3194a17245a7c7a27c1715cf7b"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6c40f5fe;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as08137794
To: <sip:pwollny@sip.voipcheap.com>
Contact: sip:77.72.169.134:5060
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1582 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm="sip.voipcheap.com",nonce="2084202765",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4c780ecd;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as2e9816e3
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1583 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084202765", response="4db53d6f0c9fc539adbc8baef3beb139"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4c780ecd;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as2e9816e3
To: <sip:pwollny@sip.voipcheap.com>
Contact: <sip:2062036594@192.168.80.1:5060>;expires=120
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1583 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' in 32000 ms (Method: REGISTER)
[2012-10-21 12:33:21] NOTICE[13062]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for sip.voipcheap.com is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog '1582944178' Method: REGISTER

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
Really destroying SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' Method: REGISTER
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
[2012-10-21 12:35:06] NOTICE[13062]: chan_sip.c:13059 sip_reregister:    -- Re-registration for  pwollny@sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4d847b9f;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as7cfc2f3b
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1584 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084202765", response="4db53d6f0c9fc539adbc8baef3beb139"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4d847b9f;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as7cfc2f3b
To: <sip:pwollny@sip.voipcheap.com>
Contact: sip:77.72.169.134:5060
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1584 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm="sip.voipcheap.com",nonce="2084308234",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK137e616f;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as3018dd41
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1585 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084308234", response="0d2c10f7fb91def4d10a03d2a95009b0"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK137e616f;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as3018dd41
To: <sip:pwollny@sip.voipcheap.com>
Contact: <sip:2062036594@192.168.80.1:5060>;expires=120
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1585 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' in 32000 ms (Method: REGISTER)
[2012-10-21 12:35:06] NOTICE[13062]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for sip.voipcheap.com is 120 sec (Scheduling reregistration in 105 s)
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
Really destroying SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' Method: REGISTER

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
dcerouter*CLI>
dcerouter*CLI>
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
[2012-10-21 12:36:51] NOTICE[13062]: chan_sip.c:13059 sip_reregister:    -- Re-registration for  pwollny@sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK5b478817;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as769b2582
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1586 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084308234", response="0d2c10f7fb91def4d10a03d2a95009b0"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK5b478817;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as769b2582
To: <sip:pwollny@sip.voipcheap.com>
Contact: sip:77.72.169.134:5060
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1586 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm="sip.voipcheap.com",nonce="2084413718",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK5c7fc405;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as0ce7d75e
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1587 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084413718", response="1366aec33526c429e259e206276a95be"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK5c7fc405;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as0ce7d75e
To: <sip:pwollny@sip.voipcheap.com>
Contact: <sip:2062036594@192.168.80.1:5060>;expires=120
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1587 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' in 32000 ms (Method: REGISTER)
[2012-10-21 12:36:52] NOTICE[13062]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for sip.voipcheap.com is 120 sec (Scheduling reregistration in 105 s)
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
Really destroying SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' Method: REGISTER
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
dcerouter*CLI> sip show peers
Name/username              Host                                    Dyn Forcerport ACL Port     Status     Realtime
200/200                    192.168.80.1                             D   N             5061     Unmonitored Cached RT
204/204                    192.168.80.30                            D   N             5060     Unmonitored Cached RT
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
Title: Re: FreePbx
Post by: microbrain on October 21, 2012, 10:41:11 pm
Pw44,

The first thing you need to do is get all the "401" & "403" response codes fixed.

I'm seeing that your VoIp provider is not rejecting(401 code) then accepting. Educate me again on some things:

How does you machine communicate with the "outside" world? From LMCE to a router, to a DSL/Cable modem and are you using a static IP? If dynamic IP have you registered with someone like dyndns.org or are you just dependant on your Internet Service Provider?

On the spa, if I am correct I'm seeing 401 & 403s, you need to check that the user name & password are the same as what LMCE has entered in its database. It could be that it is and the database is not storing the passwords the same. I have had this issue before in the past, a corrupt database.....

I'm also seeing that the spa is trying to communicate on port 5061, is this the only port in the spa that is assigned, if so you need to have it at beginning at 5060 and end with blank or at least 5061.
I'm assuming that it is set for 5061, asterisk normally starts with 5060....

Had to look your post over pretty quick as I have to leave for church but will check it again when I return but in the passing time check for all your 401 & 403 and try to correct them, as I say it's a authentication issue. I have a website that will give you the sip responses at: http://en.wikipedia.org/wiki/List_of_SIP_response_codes and you can always go to asterisk web site for additional info.

microbrain
Title: Re: FreePbx
Post by: pw44 on October 21, 2012, 11:25:49 pm
Microbrain,
thx for the answer, and now to the details:

Pw44,

The first thing you need to do is get all the "401" & "403" response codes fixed.

I'm seeing that your VoIp provider is not rejecting(401 code) then accepting. Educate me again on some things:

How does you machine communicate with the "outside" world? From LMCE to a router, to a DSL/Cable modem and are you using a static IP? If dynamic IP have you registered with someone like dyndns.org or are you just dependant on your Internet Service Provider?

LMCE box - external nic -> router -> adsl modem
IP setting dynamic and i also have dyndns setted on the external router.
LMCE firewall:
tcp    ipv4    443                  core_input       Delete
tcp    ipv4    2000          core_input       Delete
udp    ipv4    2000          core_input       Delete
udp    ipv4    4569          core_input       Delete
udp    ipv4    5060          core_input       Delete
udp    ipv4    10001 to 20000    core_input       Delete

Both external sip providers (sipgate and voipcheap) uses udp 5060.

Quote

On the spa, if I am correct I'm seeing 401 & 403s, you need to check that the user name & password are the same as what LMCE has entered in its database. It could be that it is and the database is not storing the passwords the same. I have had this issue before in the past, a corrupt database.....

I'm also seeing that the spa is trying to communicate on port 5061, is this the only port in the spa that is assigned, if so you need to have it at beginning at 5060 and end with blank or at least 5061.
I'm assuming that it is set for 5061, asterisk normally starts with 5060....


The spa3102 is connected in the internal network, as the log shows (192.168.80.30).
The spa configuration has two parts: pstn and line 1.
Line 1 is defined as extension, and registers as a sip phone.
The pstn is defined with port 5061 and user and password were checked and are the same. The whole configuration of the spa3102 is the same as the wiki, http://wiki.linuxmce.org/index.php/Linksys_SPA3102, including username and password
Subscriber Settings

    Display Name - Unknown Caller (this is what is displayed when a call comes in without caller ID info)
    UserID - spa3102 (this is what you set when creating the Phone Line in the LinuxMCE webmin)
    Password - lmce (this is what you set when creating the Phone Line in the LinuxMCE webmin)

In the 8.10 release, it worked well, with the new release, it does not authenticates, and i'm not finding out the reason, as userid, password, port and settings are the same.

Quote

Had to look your post over pretty quick as I have to leave for church but will check it again when I return but in the passing time check for all your 401 & 403 and try to correct them, as I say it's a authentication issue. I have a website that will give you the sip responses at: http://en.wikipedia.org/wiki/List_of_SIP_response_codes and you can always go to asterisk web site for additional info.

microbrain

The other issues are:
cisco sccp extension 203 calling sip on spa3102 extension 204: 204 rings, i can hear from 203, but 203 does not hears from 204 :(
sip on spa3102 extension 204 calling cisco sccp extension 203: always busy.

Well, any help is welcome in order to solve it all.

BTW, where should i define the dialplans according to the trunk?

Best regards and thx again.

Paulo
Title: Re: FreePbx
Post by: pw44 on October 21, 2012, 11:53:45 pm
Well, something new to report:
having 2 sccp extensions and 1 sip extension, the results are:
calling from outside to my voipcheap trunk, sip extension rings but not the sccp (yes, all are configured to ring)
calling from outside to my sipgate trunk, sip extension rings, but not the sccp (yes, all are configured to ring).
Answering the call from the sip extension, vice both ways (perfect).
Calling from any sccp extension to the sip, voice one way only.
Anyone with a mixed environment (sccp and sip extensions)?
And i did find out that the /usr/share/asterisk/sounds/pluto are wrong (no voicemail)......
Title: Re: FreePbx
Post by: microbrain on October 22, 2012, 01:59:02 am
Pw44,

"BTW, where should i define the dialplans according to the trunk?", that and other isses from what I am seeing with the way LMCE setup files with Asterisk this is the problem.

Where we had (under FreePBX) the ability to set the following:

General Settings, Dialed Rules, Outgoing Peer Details (including the ability to list what codecs to use), User Context & Details plus the Registration String....

we don't under the way LMCE has it setup now. Not all VoIp providers have a "Standard" of connection. It would be nice if they did as it would make life easier. That's where something like FreePBX comes in so that you can modify necessary information. Just like under "Phones" we don't have the ability to choose the codecs, the Outbound CID Number Alias, and a host of other options.

If LMCE could find it in their design to at least code the phone line page & phones page to resemble the same as is under FreePBX it would be great.

I, under the circumstances, don't feel I can help you any further as I see the problems being all within LMCE. It will take a rewrite of some code to fix.

Sorry I couldn't be of better help

microbrain
Title: Re: FreePbx
Post by: pw44 on October 22, 2012, 03:03:41 am
Microbrain,
you are right. On the previous version, 8.10, with freepbx, i got all running as you said.
With the new one, i'm not finding where and how to make it..... and maybe that's the problem: my ignorance and lack of documentation :)
Anyway, thx again for trying to help.
Best regards,
Paulo
Title: Re: FreePbx
Post by: mcefan on October 22, 2012, 03:55:19 am
mcefan, if you find it useful to add a link to this thread, why didn't you

Because I had no clue where that was! Unfortunately, people assume others know. I for one, don't usually work with these things, so, this is a learning experience for me, and in the process,  I made up my mind to try to lower the entry curve for others. Not everyone interested in the project is a developer, and I really believe that a larger user base will help the project a great deal, and that's my goal. The entry level learning curve is ridiculously steep, so, every time you see me asking or commenting, I'm not necessarily thinking about myself.

When I do know, I do post the links.

Hope this helps clear things a bit.
Title: Re: FreePbx
Post by: posde on October 22, 2012, 12:17:51 pm
In trac just add a comment
Title: Re: FreePbx
Post by: mcefan on October 22, 2012, 07:06:51 pm
In trac just add a comment
I don't know what trac is...
but since the subject came up, I looked on the main site and found the link "Tracker" under "Developer", which takes me here (http://svn.linuxmce.org/).
Seeing that I am not a developer, I would not have ventured under that heading.
I suggest we add a simple paragraph to the wiki that will point people to it, with a simple explanation of when to do so, and links to the most important pages there.
Title: Re: FreePbx
Post by: pw44 on October 23, 2012, 10:52:22 pm
The most frustrating is that is almost impossible to get help and support....  no one knows and the few that knows are silent... in the asterisk forum, etc... If there was any decent documentation... but this is also a no show.
my spa3102 is there, as nmap shows:
Code: [Select]
dcerouter_1031272:/home/paulo#  nmap -p 5061 -sU 192.168.80.30

Starting Nmap 5.00 ( http://nmap.org ) at 2012-10-23 18:19 BRST
Interesting ports on 192.168.80.30:
PORT     STATE         SERVICE
5061/udp open|filtered sip-tls
MAC Address: 00:0E:08:C0:97:55 (Cisco Linksys)

Nmap done: 1 IP address (1 host up) scanned in 7.05 seconds
dcerouter_1031272:/home/paulo#  nmap -p 5060 -sU 192.168.80.30

Starting Nmap 5.00 ( http://nmap.org ) at 2012-10-23 18:20 BRST
Interesting ports on 192.168.80.30:
PORT     STATE         SERVICE
5060/udp open|filtered sip
MAC Address: 00:0E:08:C0:97:55 (Cisco Linksys)

Nmap done: 1 IP address (1 host up) scanned in 7.01 seconds
dcerouter_1031272:/home/paulo# nmap -p 5061 -sU 192.168.80.1

Starting Nmap 5.00 ( http://nmap.org ) at 2012-10-23 18:20 BRST
Interesting ports on dcerouter.localdomain (192.168.80.1):
PORT     STATE         SERVICE
5061/udp open|filtered sip-tls

Nmap done: 1 IP address (1 host up) scanned in 2.13 seconds

But asterisk claims no matching peer, but knows the peer is located at 192.168.80.30
Code: [Select]
[2012-10-23 18:12:14] NOTICE[2494]: chan_sip.c:24933 handle_request_register: Registration from 'Unknown Caller <sip:spa3102@192.168.80.1>' failed for '192.168.80.30:5061' - No matching peer found
    -- Registered SIP '204' at 192.168.80.30:5060
       > Saved useragent "Linksys/SPA3102-5.1.7(GW)" for peer 204
[2012-10-23 18:12:14] NOTICE[2494]: chan_sip.c:24933 handle_request_register: Registration from 'Unknown Caller <sip:spa3102@192.168.80.1>' failed for '192.168.80.30:5061' - No matching peer found

spa3102 = username = trunkname.
Password checked and rechecked.
What could be wrong? Me? I'm considering myself too stupid to understand what's going on  :'(
Title: Re: FreePbx
Post by: posde on October 23, 2012, 11:11:34 pm
see if you can find anything useful using sip debug on:

sip set debug {on|off|ip|peer} Enable/Disable SIP debugging
Title: Re: FreePbx
Post by: microbrain on October 24, 2012, 12:33:38 am
Pw44,

As I said earlier I think it's going to be a bug within LMCE and it's db. Please run the following command from a command line on the LMCE machine:

sip set debug ip 192.168.0.3  (put in your IP address of you LMCE machine that the SPA is trying to communicate on)

The post here so I can see what it is doing. Since there is no peer found running sip show peer won't show what's going on between the SPA and LMCE, using the "IP" will.

Asterisk is not finding the peer name you are trying to register with the SPA. There is either a user name or password not matching up with what Asterisk is looking for based on what Asterisk is pulling from it's db, normally Asterisk pulls this info from it's AstDB.

Also while your at it, copy and paste your "sip.conf" file here, you can find it on the LMCE server /etc/asterisk/sip.conf, oh wait a minute, there is no sip.conf on the LMCE machine I have under etc/asterisk..... Either it is located somewhere else and I'm not aware of or found yet, or, it's been renamed to something else, or, maybe someone forgot to include it in the build....

I will be tied up till after ten tonight so I won't be able to get right back to you, but I will look at it tonight. Stop pulling your hair out, I'm sure there is a simple answer to it all.

microbrain


Title: Re: FreePbx
Post by: pw44 on October 24, 2012, 07:15:56 pm
sip debug for asp3102 ata ip:
Please note that the spa3102 is serving fro two purposes:
1 - pstn line as a pots trunk - no way no register.
2 - line 1 as a sip extension 204 - this one registers and works.

Code: [Select]
dcerouter*CLI> sip set debug ip 192.168.80.30
SIP Debugging Enabled for IP: 192.168.80.30
[2012-10-24 15:08:34] NOTICE[2494]: chan_sip.c:13059 sip_reregister:    -- Re-registration for  pwollny@sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
[2012-10-24 15:08:34] NOTICE[2494]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for sip.voipcheap.com is 120 sec (Scheduling reregistration in 105 s)


<--- SIP read from UDP:192.168.80.30:5060 --->
REGISTER sip:192.168.80.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.30:5060;branch=z9hG4bK-30f57ef2
From: Line 1 <sip:204@192.168.80.1>;tag=a085e89ca30903a8o0
To: Line 1 <sip:204@192.168.80.1>
Call-ID: e37c19f1-dc99f9a0@192.168.80.30
CSeq: 34238 REGISTER
Max-Forwards: 70
Contact: Line 1 <sip:204@192.168.80.30:5060>;expires=600
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.80.30:5060 (NAT)

<--- Transmitting (NAT) to 192.168.80.30:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.30:5060;branch=z9hG4bK-30f57ef2;received=192.168.80.30;rport=5060
From: Line 1 <sip:204@192.168.80.1>;tag=a085e89ca30903a8o0
To: Line 1 <sip:204@192.168.80.1>;tag=as2d7251a9
Call-ID: e37c19f1-dc99f9a0@192.168.80.30
CSeq: 34238 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48da0ee2"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'e37c19f1-dc99f9a0@192.168.80.30' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.30:5061 --->
REGISTER sip:192.168.80.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.30:5061;branch=z9hG4bK-fc1374d1
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=77c8a0fc9a3ea6c8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>
Call-ID: 7fe3a044-958a6f69@192.168.80.30
CSeq: 57993 REGISTER
Max-Forwards: 70
Contact: Unknown Caller <sip:spa3102@192.168.80.30:5061>;expires=300
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.80.30:5061 (NAT)

<--- Transmitting (NAT) to 192.168.80.30:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.30:5061;branch=z9hG4bK-fc1374d1;received=192.168.80.30;rport=5061
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=77c8a0fc9a3ea6c8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>;tag=as20bcbad4
Call-ID: 7fe3a044-958a6f69@192.168.80.30
CSeq: 57993 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0534bc94"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7fe3a044-958a6f69@192.168.80.30' in 32000 ms (Method: REGISTER)
[2012-10-24 15:08:43] NOTICE[2494]: chan_sip.c:24933 handle_request_register: Registration from 'Unknown Caller <sip:spa3102@192.168.80.1>' failed for '192.168.80.30:5061' - No matching peer found
Scheduling destruction of SIP dialog '7fe3a044-958a6f69@192.168.80.30' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.30:5060 --->
REGISTER sip:192.168.80.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.30:5060;branch=z9hG4bK-268220b5
From: Line 1 <sip:204@192.168.80.1>;tag=a085e89ca30903a8o0
To: Line 1 <sip:204@192.168.80.1>
Call-ID: e37c19f1-dc99f9a0@192.168.80.30
CSeq: 34239 REGISTER
Max-Forwards: 70
Authorization: Digest username="204",realm="asterisk",nonce="48da0ee2",uri="sip:192.168.80.1",algorithm=MD5,response="b46a6784f5334d0ea28b614c869a1d13"
Contact: Line 1 <sip:204@192.168.80.30:5060>;expires=600
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.80.30:5060 (NAT)
    -- Registered SIP '204' at 192.168.80.30:5060

<--- Transmitting (NAT) to 192.168.80.30:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.80.30:5060;branch=z9hG4bK-268220b5;received=192.168.80.30;rport=5060
From: Line 1 <sip:204@192.168.80.1>;tag=a085e89ca30903a8o0
To: Line 1 <sip:204@192.168.80.1>;tag=as2d7251a9
Call-ID: e37c19f1-dc99f9a0@192.168.80.30
CSeq: 34239 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 600
Contact: <sip:204@192.168.80.30:5060>;expires=600
Date: Wed, 24 Oct 2012 17:08:43 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'e37c19f1-dc99f9a0@192.168.80.30' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.30:5061 --->
REGISTER sip:192.168.80.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.30:5061;branch=z9hG4bK-5349da3a
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=77c8a0fc9a3ea6c8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>
Call-ID: 7fe3a044-958a6f69@192.168.80.30
CSeq: 57994 REGISTER
Max-Forwards: 70
Authorization: Digest username="spa3102",realm="asterisk",nonce="0534bc94",uri="sip:192.168.80.1",algorithm=MD5,response="6daeb6df71cbff698c5bce7e1f3c98a3"
Contact: Unknown Caller <sip:spa3102@192.168.80.30:5061>;expires=300
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.80.30:5061 (NAT)

<--- Transmitting (NAT) to 192.168.80.30:5061 --->
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 192.168.80.30:5061;branch=z9hG4bK-5349da3a;received=192.168.80.30;rport=5061
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=77c8a0fc9a3ea6c8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>;tag=as20bcbad4
Call-ID: 7fe3a044-958a6f69@192.168.80.30
CSeq: 57994 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2012-10-24 15:08:43] NOTICE[2494]: chan_sip.c:24933 handle_request_register: Registration from 'Unknown Caller <sip:spa3102@192.168.80.1>' failed for '192.168.80.30:5061' - No matching peer found
Scheduling destruction of SIP dialog '7fe3a044-958a6f69@192.168.80.30' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog 'e37c19f1-dc99f9a0@192.168.80.30' Method: REGISTER
Really destroying SIP dialog '7fe3a044-958a6f69@192.168.80.30' Method: REGISTER
dcerouter*CLI> sip set debug off
SIP Debugging Disabled
dcerouter*CLI> quit
Executing last minute cleanups
dcerouter_1031272:

Quote
Also while your at it, copy and paste your "sip.conf" file here, you can find it on the LMCE server /etc/asterisk/sip.conf, oh wait a minute, there is no sip.conf on the LMCE machine I have under etc/asterisk..... Either it is located somewhere else and I'm not aware of or found yet, or, it's been renamed to something else, or, maybe someone forgot to include it in the build....

sip.conf is stored in the mysql database.

Thx Microbrain for your offer.

BR,

Paulo
Title: Re: FreePbx
Post by: microbrain on October 25, 2012, 02:45:51 am
Pw44,

Your problem is within the authentication of your PSTN line., but you know that already.

Three things I would look at: (make note of any changes you make)

1: Did you set the spa for dhcp or assign it a dynamic ip? If you have it as dhcp, change it to static and try that then change the #2 next, if it still don't work put it back.

2: in the spa under "Proxy and Registration" is it set for yes, if so try to set it to off and then check if it works. If it is set to on it tries to tell LMCE what your spa address is, and since you already assigned an ip you don't need to register.

3: as much as I hate to say this, go back to LMCE 8.10 because the current 10.04 will not work without the ability to setup this box with something like (here it comes, wait, wait,) FREEPBX...... and don't feel too bad as you won't be the only one having to do it.

I myself was going to try and set LMCE up to be an extension of my asterisk box, but after studying how LMCE communicates with asterisk I can't do that either nor can I make the asterisk box work as a in/out trunk to the LMCE because of the changes they made in how the LMCE deals with asterisk. I'm still not sure if I fully understand why the powers to be had to change up asterisk/lmce. If it truly was an issue (as I have read on some other post) that too many people were breaking LMCE using FreePbx I would have thought it would have been easier to just make FreePBX an add-on and not offered help when they screwed it up, just my opinion.

microbrain
Title: Re: FreePbx
Post by: tschak909 on October 25, 2012, 04:05:10 am
Spoken like a know it all, who understands nothing of the reasons and decisions as to why things were done the way they were. Way to go.

We got rid of FreePBX, because for 99% of what people needed to do, needed to be done in the web admin, under a single user interface, and so that uninformed changing of the system, wouldn't break the telecom part of the system in half. If there is a feature missing from the web admin that you need, please work with us to add it! I'm tired of all this incessant "I don't understand why they did this!" talk. You're acting like a whiny bitch.

-Thom
Title: Re: FreePbx
Post by: gbutters on October 25, 2012, 06:33:34 am
sip debug for asp3102 ata ip:
Please note that the spa3102 is serving fro two purposes:
1 - pstn line as a pots trunk - no way no register.
2 - line 1 as a sip extension 204 - this one registers and works.

For the PSTN line setup a phone line in lmce as spa with the phone number (1234567890) as the user and in spa3102 config under pstn line

Line Enable:   yes      
 
SIP Settings
SIP Port:   5061      
 
Proxy and Registration
Proxy:   192.168.80.1
Register:   yes   Make Call Without Reg:   yes
Register Expires:   300   Ans Call Without Reg:   yes
 
Subscriber Information
Display Name:   Unknown  Caller   User ID:   phone number
Password:   password   Use Auth ID:   yes
Auth ID:   phone number   


With those settings the spa3102 pstn line registers for me.
Title: Re: FreePbx
Post by: tschak909 on October 25, 2012, 03:47:10 pm
I am pasting this, because microbrain felt it important to block users private messages, while spamming me with his own:

Your response to my web post today displays the type of attitude that has the attendances to make a person like me not want to return to this website.

Quote
You sound like a 15 year old BOY that's still living at home and momma still wiping your ass.

Like anyone else, not fully understanding why the changes were done the way they were, I was hoping for a somewhat reasonable explanation behind these changes. Your explanation would have done well had you left out the condescending, degrading and name calling remarks. Most 15 year old BOYS don't understand that as they haven't lived long enough to learn it.

I was simply trying to make an attempt to help someone whom was beyond frustration to get their problem worked out since it was apparent he wasn't getting much assistance and remarks such as yours not only turn me sour toward you but also the other ton of new people whom may happen across this post.

Where I come from if you can't say something constructive in a somewhat decent attitude where it benefits all to the good then it is usually best to keep your shit to yourself, or, send it to me in a private message so you don't show others your yuppie ignorance, BITCH.


No, you misunderstand, completely understandable, as I was very fried and frazzled after working three contracts in a row for now, and the foreseeable future, and I lashed out.

I am angry, because you've basically not worked with us at ALL over what features you are DESPERATELY needing from FreePBX that can be folded into the Web Admin. Nada, nothing, you just whine and bitch and complain, as do many users on this forum, maybe because they feel entitled to do so, maybe because they feel helpless for some reason. If it's the latter, anyone who has worked with the team directly can attest that we BEND OVER BACKWARDS to help those who help themselves.

This system became a volunteer effort, and the changes that we're trying to make are the result of trying to make the system predictable and maintainable. Neither I, nor foxi352, who made the changes, nor the rest of the development team will step back from these assertions, because we believe that given the REALITY of the development team that we have, the most common use cases used by the telephone system, and the fact that we wanted to present a single user interface for configuring everything, doing what we did was and still is, the best choice.

-Thom
Title: Re: FreePbx
Post by: pw44 on October 25, 2012, 07:50:56 pm
Pw44,

Your problem is within the authentication of your PSTN line., but you know that already.

Three things I would look at: (make note of any changes you make)

1: Did you set the spa for dhcp or assign it a dynamic ip? If you have it as dhcp, change it to static and try that then change the #2 next, if it still don't work put it back.


spa is setted to a fixed ip address 192.168.80.30

Quote

2: in the spa under "Proxy and Registration" is it set for yes, if so try to set it to off and then check if it works. If it is set to on it tries to tell LMCE what your spa address is, and since you already assigned an ip you don't need to register.


Proxy and Registration: Register is set to YES. I will give a try with NO.

Quote
3: as much as I hate to say this, go back to LMCE 8.10 because the current 10.04 will not work without the ability to setup this box with something like (here it comes, wait, wait,) FREEPBX...... and don't feel too bad as you won't be the only one having to do it.

I myself was going to try and set LMCE up to be an extension of my asterisk box, but after studying how LMCE communicates with asterisk I can't do that either nor can I make the asterisk box work as a in/out trunk to the LMCE because of the changes they made in how the LMCE deals with asterisk. I'm still not sure if I fully understand why the powers to be had to change up asterisk/lmce. If it truly was an issue (as I have read on some other post) that too many people were breaking LMCE using FreePbx I would have thought it would have been easier to just make FreePBX an add-on and not offered help when they screwed it up, just my opinion.

microbrain


Yes, would be prefereable do have something like freepbx, but it's gone, so learning all again from ground zero (where and what, beside the criptic syntax, which was hidden by freepbx). But, as said, it's gone.

Thx for your trying to help :) Let's see if we get it, mostly by trial and error, due lack of documentation.
Title: Re: FreePbx
Post by: pw44 on October 25, 2012, 07:57:01 pm
For the PSTN line setup a phone line in lmce as spa with the phone number (1234567890) as the user and in spa3102 config under pstn line

Line Enable:   yes      
 
SIP Settings
SIP Port:   5061      
 
Proxy and Registration
Proxy:   192.168.80.1
Register:   yes   Make Call Without Reg:   yes
Register Expires:   300   Ans Call Without Reg:   yes
 
Subscriber Information
Display Name:   Unknown  Caller   User ID:   phone number
Password:   password   Use Auth ID:   yes
Auth ID:   phone number   


With those settings the spa3102 pstn line registers for me.

Gbutters,

thank you for your email. I will try this configuration later and will report back with results..

Best regards,

Paulo
Title: Re: FreePbx
Post by: pw44 on October 25, 2012, 08:13:05 pm
I'm not blaming not having freepbx, but the lack of documentation and information, and that's not your fault. I know it's a volunteer effort and i also understand it's not easy to make it easy for end users.
I would gladly help, if i had the time and the knowledge for it. Some very little contribution i gave (fail2ban, and some voip trunk settings).  

Well, if i can suggest, how about a little tutorial about, comparing the freepbx settings (trunk - peer detail, user detail), outbound route, dialplan according to trunk and register, or a hidden panel where we could tune it.

For me, at least following directives are missing.
* 83    0    18    0    sip.conf    general    alwaysauthreject    yes
* 85    0    18    0    sip.conf    general    nat                            yes
* 86    0    60    0    sip.conf    general    externhost            mydyndns.homeunix.org
* 87    0    5    0    sip.conf    general    externrefresh            5
* 88    0    60    0    sip.conf    general    localnet                   192.168.80.0/255.255.255.0
* 89    0    9    0    sip.conf    general    allow                    g729
* 90    0    10    0    sip.conf    general    allow                    g723
* 91    0    101    0    sip.conf    general    register                    pwollny:XXXXXXXXX@sip.voipcheap.com/2062036594


I don't know if i did it right, because i could not find what are cat_metric and var_metric for.
I'm not asking to have someone doing it for me, but i wish to know where to put what i need, and i'm not finding out :(

Best regards to all,

Paulo
Title: Re: FreePbx
Post by: posde on October 25, 2012, 08:16:43 pm
microbrain,

I wonder what you did to your other asterisk system. It is the exact setup I have here. My main asterisk acts as my ISDN gateway and caters to two offices, and is the outgoing line for my LinuxMCE system. Just added the phoneline details and it worked.
Title: Re: FreePbx (SOLVED).
Post by: pw44 on October 25, 2012, 08:34:14 pm
For the PSTN line setup a phone line in lmce as spa with the phone number (1234567890) as the user and in spa3102 config under pstn line

Line Enable:   yes      
 
SIP Settings
SIP Port:   5061      
 
Proxy and Registration
Proxy:   192.168.80.1
Register:   yes   Make Call Without Reg:   yes
Register Expires:   300   Ans Call Without Reg:   yes
  
Subscriber Information
Display Name:   Unknown  Caller   User ID:   phone number
Password:   password   Use Auth ID:   yes
Auth ID:   phone number   


With those settings the spa3102 pstn line registers for me.

Thank you. It worked. spa3102 registered. This is a difference between the old asterisk (used in release 8.10) and the new one (used in release 10.04).

Again, big THX!!!!!!!!!!!!

Wiki updated with the info.
Title: Re: FreePbx
Post by: posde on October 25, 2012, 08:49:58 pm
pw44,

instead of posting it to the wiki, see if you can create a pnp script for the SPA. Look at the grandstream perl scripts how it is done.
Title: Re: FreePbx
Post by: pw44 on October 25, 2012, 09:44:27 pm
Ja, mein Kommandant! Werde da nachschauen!
Title: Re: FreePbx
Post by: posde on October 25, 2012, 10:12:53 pm
;)
Title: Re: FreePbx
Post by: microbrain on October 26, 2012, 06:26:25 am
posde,

I didn't do anything to my asterisk server, everything on it is and has been working fine for several years without a glich. So lets just accept I'm totally ignorant when it comes to LMCE and start at the beginning.

Since I don't find any info on how to setup 10.04 phonelines maybe you are willing to share how you are connecting to your asterisk box, with sip or aix2?

My original set up (8.10) was a peer/user arrangement using sip that worked just fine. I've tried to set it up as a  "peer asterisk box as an extension" and as a "fried/friend arrangement" and neither is working. I'm not even seeing the LMCE box trying to communicate with the asterisk box through the sip debug on my asterisk box, however I am seeing on the LMCE box the asterisk box trying to communicate with the LMCE box.

So that I may get a better understanding of how you are interfacing the two separate boxes together could you post your asterisk trunk details (without passwords of course) and your LMCE phoneline info you entered into LMCE 10.04 and maybe some info on how LMCE handles the peer details and codecs.

Thank you sir

microbrain

Title: Re: FreePbx
Post by: posde on October 26, 2012, 06:39:17 am
SIP. I merely entered username, password, phonenumber, prefix and host address.

My main asterisk's SIP conf for the LinuxMCE "user" looks like this:
Code: [Select]
[959]
type=friend                     
context=privat          ; Where to start in the dialplan when this phone calls
host=dynamic            ; we have a static but private IP address
defaultip=10.1.2.67
insecure=port,invite
                                ; No registration allowed
nat=no                          ; there is not NAT between phone and Asterisk
canreinvite=yes         ; allow RTP voice traffic to bypass Asterisk
dtmfmode=info                   ; either RFC2833 or INFO for the BudgeTone
username=959
secret=myverysecretpassword

Title: Re: FreePbx
Post by: microbrain on October 26, 2012, 06:35:26 pm
Under "PhoneLines" there is a "Status", is this status showing it's registered, in use, what?

The problem is that the line shows "enabled" but status shows "no". I'm assuming (we know what that means) that the "status" means it's not registering with the other end.

I can call from my asterisk box into LMCE and it rings the extension assigned on incoming route, I can not though dial out to the asterisk box as it continues to show "congestion/busy".

What version of asterisk does LMCE use?

Thanks
microbrain
Title: Re: FreePbx
Post by: microbrain on October 27, 2012, 05:31:02 am
Well I got the "phonelines" as it's call set up and figured out that "Status" is for show if registered or not (no).

All works but audio. What codec is assigned within the LMCE when a phoneline is setup?

In reference to possibly making certain available options for the end user to access, it would sure be nice to have the following (just a request -- not a demand) so as to make it easier to set up trunks and outgoing dial patterns:

Outbound Routes Setting (for setting up different dial patterns based on best price practice)
Outbound Trunk Dial Rules (The current "prefix" lets you select the trunk to use but not dial rules on that trunk)
Outgoing Peer Detail Settings ( allows for fine tuning and or manipulation of trunk/peer info)
Incoming Peer Detail Setting ( same as outgoing peer details)

Since LMCE sip.conf info is stored within the database and not accessible under /etc/asterisk (where it normally resides on most asterisk distros) it makes it hard to modify trunk & extension details along with codecs when necessary. Although I can set all this in my separate asterisk box for someone who doesn't have a separate box and needs to choose different dial patterns on outbound routes or trunks and/or special mods to peer info I could see where this would be beneficial. Just a thought.

Thanks
microbrain

Title: Re: FreePbx
Post by: mcefan on November 02, 2012, 05:31:07 am
microbrain: Could you post a complete description of what you did to make your equipment work please?
Or better, create a tutorial page on the wiki for it, and link it in the thread. We really need things documented for others, so they don't go through what you went through. Please remember that not everyone "knows" asterisk well, so make sure you provide as much detail as you can an don't skip over the "obvious".

Thanks
Title: Re: FreePbx
Post by: microbrain on November 04, 2012, 07:52:35 am
Sir  mcefan;

I sent you a responce to your PM by PM. Basically I have chosen not to use the asterisk portion within LMCE10.04 since it doesn't meet my needs after trying for several hours to make it work.

Thank you for your request and should I at a later date get it up and running I will post how it was completed.

Thank you
microbrain
Title: Re: FreePbx
Post by: pw44 on December 05, 2012, 10:02:10 pm
pw44,

instead of posting it to the wiki, see if you can create a pnp script for the SPA. Look at the grandstream perl scripts how it is done.

Posde, i did not forget the issue. Searching, there is one way, but it's a backup done by saving the webpage of the spa3102 and editing some parameters and then loading the page again.
I'm not sure if this is a reliable solution. I will keep you posted.
BR
Title: Re: FreePbx
Post by: totallymaxed on December 10, 2012, 06:16:38 pm
I downloaded the stable addition of 8.10 and installed it on a test box to try and get a grip on an actual working LMCE. In playing with it I noticed that FreePBX was a part of that version. I know that I can download and install FreePBX but I'm not sure if the current version of FreePBX is what works with 8.10 or 10.04-26551. Is there a script, add-on a known way that I can install FreePBX on 26551 of 10.04 and where would I need to change the link in LMCE admin page to get to FreePBX once I install it? Maybe someone has install it on 10.04 and would like to post how they did it. I read in some other posts that it was removed due to people not familiar with FreePBX making changes that caused breaking LMCE.

I would like to take advantage of additional options within the Asterisk PBX that are not accessible within LMCE.

Thanks

microbrain

Dianemo S on 12.04LTS still uses FreePBX if that's of interest.

All the best

Andrew
Title: Re: FreePbx
Post by: hari on December 10, 2012, 11:40:07 pm
use ago control, you have to configure Asterisk yourself :-)