similar to: dialplan problems cvs 04-15-03

With multiple inbound PRIs (and hunting across them) coming to multiple [asterisk] servers it is important to be able to do administration, i.e. control which PRIs in the same hunt group take (and which don't take) calls from telco at any given period of time. Our pre-asterisk platform uses SERVICE commands for this purpose to put B-channels into 'out-of-service'/'maintenance'

Hi We are having a significant (> 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes. All the other legs are PSTN (TE410P). The example configuration Slave box 1 meetme <--- IAX2 ---> Master box meetme <--- IAX2 ---> Slave box 2 meetme The delay is between Slave box 1 and Slave box 2 The primary suspect is our iax configuration

Hi Steve and all, 1. Faxing from asterisk back to the same asterisk (from one Zap channel to another) doesn't work for us. Txfax called with the 'caller' parameter issues CED, while the receiving side needs CNG in order to switch to fax extension with rxfax. 2. This is probably the reason why J2 and our UC don't recognize incoming fax. Thank you. Alex Zarubin Webley Systems

Is there a way to play adpcm, 6KHz in asterisk? If yes, where can we get this codec? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030916/b8be2453/attachment.htm

Hello, The v1-0_stable cvs release doesn't include the recent change ('poll' instead of 'select') in channel.c. Will it end up there any time soon, or we need to use cvs head to pick up this change? Thank you. Alex Zarubin Webley Systems -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ... grep -e "Error" -e "eventually" p-console chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got

Configuration: Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown P4 2.5 GHz, 1 GB RAM T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s). Each call gets transferred (Dial) to the SIP platform and stays for 5 min. Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. Segmentation fault. Case 2. Asterisk built out of CVS Apr. 1. Test was running

Hi, We are trying to validate Asterisk as a media gateway PRI <-> SIP with two T400P (8 T1s) per box. The first experience with BOX1 (Compaq, 2.53 GHz, 1 Gb RAM) and just one T400P was encouraging - on the load test with 3 T1s worth of calls we had on average 75% idle CPU. Not so with BOX2 (Dell, single 2.6 GHz Xeon, 1 Gb RAM, 2 T400P) and BOX3 (Dell, dual 2.6 GHz Xeon, 2 Gb RAM, 2 T400P,

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If there are multiple NICs in the box, how do we specify the local IP address to be used for RTP? Anything in rtp.conf ? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030609/d151f190/attachment.htm

Can T400P be configured for nfas (one d-channel providing signaling for more than one span)? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030522/f2b637a9/attachment.htm

Hello, If I understand it correctly, when sending INVITE, o= and c= sdp fields are built using p->ourip IP address. At this point RTP packets will be coming to the default asterisk IP address. For the machine with multiple interfaces this could be not the right one (not what we want). Could it be configured (in rtp.conf or in sip.conf per context) ? Thank you. Alex Zarubin --------------

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Hello, Question for developers: what is the asterisk way to integrate with ASR (speech recognition)? Question to the community: has someone done anything in this direction? On the first glance that shouldn't be too hard. One part is delivering audio to the engine (for example, main ASR players Nuance and Speechworks will be happy with RTP) - can be done via RTP forking. The other part is

Hello, We are moving from an older asterisk/SIP to a newer (18+) asterisk/PJSIP and trying to figure out how to add [identity] header when originating a call from AMI/PAMI. In the older version we would just set a variable like this: $action = new OriginateAction("SIP/...."); $action->setVariable('__SIPADDHEADER51',"Identity: $identity"); // $identity

Hi all, After a long time having no time, I have finally done some fresh work on my software fax machine. I have replaced the original carrier tracking with something more robust. I have also added 4800, and 2400 bits per second modes, and cleaned up a few bugs in areas like superfine mode operation. I apologise for this update taking so long. At ftp://ftp.opencall.org/pub/spandsp you will

Gastman (cvs 05/21/03) coredumps when entering an empty (or any other) command in a command window. The backtrace follows... ... ---Type <return> to continue, or q <return> to quit--- Reading symbols from /usr/lib/gtk/themes/engines/libraleigh.so...done. Loaded symbols for /usr/lib/gtk/themes/engines/libraleigh.so Reading symbols from

I used: cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds iaxyprov astcc and in the same order I try to compile it. Asterisk ends with the lines below. It complains of a newer libpri, but I just did it a step before! What do I miss? chan_zap.c:62:2: #error "You need newer libpri" chan_zap.c:128: error: parse error before '<<' token chan_zap.c:133:1:

I just rebuilt it and watched this time. What are the ? about ? [root@carol src]# cvs checkout zaptel libpri asterisk ? libpri/libpri.so.1.0 ? libpri/pri.lo ? libpri/prisched.lo ? libpri/q921.lo ? libpri/q931.lo ? asterisk/doc/api cvs server: Updating zaptel cvs server: Updating libpri cvs server: Updating asterisk cvs server: Updating asterisk/agi cvs server: Updating asterisk/apps cvs server:

Please, try exten => 99,1,Wait,1 exten => 99,2,Record,/tmp/pcmfile:pcm exten => 99,3,Wait,1 exten => 99,4,Playback,/tmp/pcmfile exten => 99,5,Wait,1 exten => 99,6,Record,/tmp/voxfile:vox exten => 99,7,Wait,1 exten => 99,8,Playback,/tmp/voxfile (put your own extension). Pcm recording is OK, playback is OK. Adpcm recording is noticeably worse. Adpcm playback is very