similar to: Setting local IP address for the RTP port

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

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'

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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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

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

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,

Hi, I'm using gotoiftime on asterisk, but it seems&nbsp; there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob? Regards, nhadie -------------- next part -------------- An HTML attachment was scrubbed... URL:

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Howdy, How do i perform a lookup from a remote odbc database in the asterisk dialplan? I can do it with mysql but not sure of commands for odbc connection. Cheers!!!

Hi. Is it possible to override the standard DB function in Asterisk? My dialplan contains a lot of calls to "Set(DB(...))" and "${DB(...)} which of course use astdb to store/read data. I would like to stop using astdb and switch to Clustered MySQL (I don't suppose "clustered astdb" exists?). So instead of rewriting extensions.conf and replacing the DB calls with

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

Is there any way to return multiple values from functions defined in func_odbc.conf? It appears that you can only return one value. True? Hope not.... Doug. ____________________________________________________________________________________ Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how. http://overview.mail.yahoo.com/ -------------- next part

Dialplan stopped working after I did cvs update for zaptel-zapata-libpri-asterisk and 'make clean', 'make', 'make install' for the above components on 04-15-03. All the config files are the same as before. Both PRI and SIP calls I am making forward calls to 's' in the default context. Fields 'from' and 'to' look normal. My attempts to fix it by

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