Where are the Asterisk Files?

[seth]

 
I have been playing around lately with the Telecom in LMCE 10.04. I understand now that everything asterisk is stored in a database.
However, I would like to know where it is getting it's information from. The problem is, in order for my SIP softphone ( 3CX from the android market ) to work, I have to ensure that the 'host' setting remains set to 'dynamic'. I go in through phpMyadmin, and I can set the field, but something is whacking my settings, and setting them back to 'Null'

Where is the script that changes asterisk located, and where are the files like sip.conf and such stored. I looked in /etc/asterisk, but they are not there, and something is filling in the defaults after a period of undisclosed time. I can change it by hand, and it works, but then when I use it later, the SIP phone fails to register, and sure enough, if I go back into the phpMyadmin for asterisk, it is right back to 'Null'

I also have a feeling this is why my MD phones are not working. Plus, if I had a sip.conf file, I could get my SPA3102 to register, by adding the appropriate line to it.

Thanks in advance.

Best Regards,

Seth

[seth]

Bump.

Is anyone familiar with asterisk in 10.04. I grow weary of having to use phpmyadmin to change my SIP clients every day.

Thanks.

Best Regards,

Seth

[twodogs]

I got so weary of dealing with LMCE telecom that I yanked the ATA out of the LMCE network. Now at least I have a phone that works. I actually bought a second ebay ATA for the day when I try to get LMCE telecom working again.

twodogs

[Marie.O]

If there are problems in 1004's asterisk they will get fixed, if usable trac tickets are created, and people revisit those tickets to answer questions, and/or visit IRC to talk to foxi directly about it. But first, open tickets for each and every asterisk problem you encounter.

btw: Asterisk is working nicely for me under LinuxMCE 1004 with some grandstream phones and a single SIP channel to a provider.

[brononius]

Hey,

I can't figure out how i must add phone (sip phone cisco 7940) in 1004.
Is there something like the admin page of asterisk? When i go to Advanced > Configuration > Phones setup, i've got an "The requested URL /admin/ was not found on this server."
When i check the folder /var/www, i don't have anything related asterisk.

Is this a small bug? Or am i overlooking something? 

[cfernandes]

i have create a script to configure  cisco 7940 as a sccp 

but  the correct url  is not /admin/   is /lmce-admin/ 


your  10.04  us updated ?

[brononius]

I have create a script to configure  cisco 7940 as a sccp 
but  the correct url  is not /admin/   is /lmce-admin/ 
your  10.04  us updated ?

Hey,

I'm running the latest 1004, and in there, i don't have /admin anymore. 
Before, this was the admin page of asterisk.

Today (1004), i've got pluto-admin and lmce-admin. Both are the adminpages of LinuxMCE.

But beside to limited pages for adding a phone, i can't find anything else.
Like for example, where can i check my phone status or change the SIP password... Where can i add phone groups?

[cfernandes]

well , 
to manage  password can you go to  phones  on lmce-admin  you can view all of yor phones ,

for now no control panel to show phones, status.

[brononius]

to manage  password can you go to  phones  on lmce-admin

Ahh,

I thought this was controlled by 'asterisk', so that it was best not to touch the phoneconfig in lmce...

I hope that the rest will follow soon. So we can start creating phonegroups and so on...

[cfernandes]

to add phones  you need to   lmce-admin  -> wizard ->devices ->phones

[brononius]

I've tried that one, and the SIP files are automaticlly generated in the /tftpboot/ folder.

But my phones aren't ringing when i've got an incoming call, nor can i dial out. In the "call details records", i can see calls coming in, and the are answered automaticlly. So at least the server is recieving the correct phone number.


Is it normal that it takes a very long time to refresh the page once i added a phone? Takes about 3 minutes before i can add a second phone...



ps pitty that you can't assign own numbers. Today, it jumps from 201 towards 315 and so. Not very easy to remember... :$

[cfernandes]

you can configure on  Telecom | Call routing

what phones need to ring

[brononius]

Can you somewhere define the outgoing routes?

When i call now from a sip phone, i got a nice voice telling me that all circuits are busy.
Or maybe it has something to do with the amount of numbers? For example i personally don't like an extra 0 to dial outside. And in Belgium the amount of numbers is rarther short:

• fixed line: 01 234 56 78
• cell phone: 0123 45 67 89

The phones are trying to go outside, the 'line' is trying to come in. But somewhere in between, they don't meet...

Some logging:
Call Date   File   Src Channel   Source   Application   Destination   Dst Channel   Disposition   Duration   Userfield   Account      
2012-07-24 20:01:02      SIP   201   Playback   00479112233   SIP   FAILED   00:03            
2012-07-24 20:00:53      SIP   201   Playback   0479112233   SIP   FAILED   00:02            
2012-07-24 19:53:29      SIP   201   Playback   1207   SIP   FAILED   00:02            
2012-07-24 19:53:22      SIP   201   Playback   01207   SIP   FAILED   00:03            
2012-07-18 20:34:02      SIP   201   Playback   209   SIP   FAILED   00:02            
2012-07-18 20:20:04      SIP   201   ResetCDR   0479112233   SIP   FAILED   00:05            
2012-07-18 20:19:55      SIP   201   Playback   00479112233   SIP   FAILED   00:03            
2012-07-18 20:07:33      SIP   201   Playback   00479112233   SIP   FAILED   00:03            
2012-07-18 20:00:14      SIP   201   Playback   0479112233   SIP   FAILED   00:03            
2012-07-18 19:55:59      Local   003293112233   Dial   201   SIP   NO ANSWER   00:15            
2012-07-18 19:55:59      SIP   003293112233   Set   003293112233   Local   ANSWERED   00:20            
2012-07-18 19:55:05      Local   003293112233   Dial   201   SIP   NO ANSWER   00:15            
2012-07-18 19:55:05      SIP   003293112233   Set   003293112233   Local   ANSWERED   00:29            
2012-07-18 19:52:42      SIP   201   Playback   00479112233   SIP   FAILED   00:02            
2012-07-18 19:52:21      SIP   0479112233   Congestion   s      ANSWERED   00:11            
2012-07-18 19:48:46      SIP   201   Playback   0479112233   SIP   FAILED   00:03            

[pw44]

Seth, did you get the spa3102 working with lmce 10.04?
BR,
Paulo

[pw44]

If there are problems in 1004's asterisk they will get fixed, if usable trac tickets are created, and people revisit those tickets to answer questions, and/or visit IRC to talk to foxi directly about it. But first, open tickets for each and every asterisk problem you encounter.

btw: Asterisk is working nicely for me under LinuxMCE 1004 with some grandstream phones and a single SIP channel to a provider.

Posde,
which sip provider ar you using?
BR
Paulo