LinuxMCE Forums
General => Users => Topic started by: seth on May 16, 2012, 07:54:40 pm
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???
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
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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
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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
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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.
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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? ???
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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 ?
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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?
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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.
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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...
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to add phones you need to lmce-admin -> wizard ->devices ->phones
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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... :$
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you can configure on Telecom | Call routing
what phones need to ring
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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
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Seth, did you get the spa3102 working with lmce 10.04?
BR,
Paulo
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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
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My own asterisk server elsewhere.
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Well, it seams that for a fast fix, maybe building a smal asterisk server and making it serve lmce as the sip provider is a dirty but reliable solution :) I will think about, as for the last two weeks i'm trying to get at least voipcheap to work, but no luck :)
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Or you could try to get on IRC and catch foxy.