similar to: asterisk + openBTS

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

Hi I met a problem in asterisk, please see message in the following, the detail debug log is in the attached file. can someone help to point out where to correctly configure asterisk, thanks a lot ! BR/Scott -------> -- Executing [8690 at phones:1] Dial("SIP/IMSI466990004244439-00000014", "SIP/IMSI466974104638690") in new stack Really destroying SIP dialog '

I try to configure that only registered sips can make calls. How can I do that? I was looking in sip.conf but I didn?t found wath opition configure this functionality. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071212/cfed2687/attachment.htm

I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other. I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk I set up two phones in sip.conf and extensions.conf. In my SIP output I see this: WARNING[1689]: app_dial.c:2041 dial_exec_full:

Is a normal functionality? when I do #dahdi_cfg -vvvvvv In my Asterisk console shows this.... [Jul 24 13:39:08] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 If I do this a lot of times...then [Jul 24 13:39:20] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24

Is there any way to do this? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120307/77764e4b/attachment.htm>

I have a server with an asterisk ss7 link connected to a Siemens working well for over a year. A few days ago I started having problems with signaling. I found the following logs in / var / log / messages Sep 12 11:49:25 call3 kernel: [1018427.030959] dahdi: Master changed to TE2/0/2 Sep 12 11:49:25 call3 kernel: [1018427.120740] dahdi: Master changed to TE2/0/1 Sep 12 11:49:26 call3 kernel:

I want the callers don't join in a queue when the agents are busy. I suposse it is easy but i can't get the solution for this. Can you suggest me something? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081229/3545ab8c/attachment.htm

Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100802/4f58fea7/attachment.htm

How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten => 492,1,Answer exten => 492,2,eagi,InfMsg -s 1 exten => 492,3,Hangup() It doesn?t work my * report... -- Executing [92@telpin-112:2] EAGI("Zap/4-1", "InfMsg -s 1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg

<asterisk-users@lists.digium.com>Asterisk 1.4 I have strategy= leastrecent and autofill = yes I have 2 agents, one is answering a call and the other is free and have some calls waiting in the queue. Only when the first agent hangup the second agent receive the first call in the queue. It happends some times. This behavior still happend in 1.4.1 version. Thanks a lot. -------------- next

Hi! I need to use text to speech, what is the best application? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070829/bc69eb9d/attachment.htm

Hi! I have this errors compiling Asterisk 1.4.5 cdr_tds.c:86:2: warning: #warning "You have older TDS, you should upgrade!" cdr_tds.c: In function `tds_log': cdr_tds.c:213: error: too many arguments to function `tds_process_simple_query' cdr_tds.c: In function `mssql_connect': cdr_tds.c:326: error: `TDSCONNECTINFO' undeclared (first use in this function) cdr_tds.c:326:

Hi Dave, >> context=openbts >> callerid=4735202222 >I see you are using OpenBTS. To my understanding, OpenBTS does not >support caller ID, so I don't think it can work. >But as I have the same issue as you, I'd be glad to be wrong ! :D Let me know. Disregard my answer. I just tested the callerid on my OpenBTS and it worked. So the problem you encounter must be

Friends, After listening to Mark Summer's keynote at Astricon (hopefully soon on the Astricon web site) I think we should come back to the discussion he started. Mark talked about using Open Source in general and Asterisk in particular in third world projects as well as in emergencies in other countries. He and Inveneo help groups of people to get a better understanding of how to build

Is there any option in Asterisk distribution of this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120531/8fac6a22/attachment.htm>

Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. Thanks. Ronaldo.

Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 <- call1 queue1 <- call2 queue2 <- call3 queue2 <- call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could

Hi, I'm trying to use Swift tts from eagi, my problem is when I send EXEC SWIFT <*prosody rate*=\'.8\' >Hello World\, this is a test\,</*prosody* >|0|1 Would I use a scape character? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100301/7132be4c/attachment.htm

Hi ! I have a python script that create and move .call files to /var/spool/asterisk/outgoing Sometimes...(in this case after 500 successfull calls) Asterisk don?t make the calls and the .call files are in the "outgoing" forever... Any Ideas? I'm using Asterisk 1.4.22 (in 1.4.36 was the same behavior) In my python script I move .call files using ... import shutil