search for: zalex

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

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

gtk12-config --version 1.2.10 glib12-config --version 1.2.10 AZ -----Original Message----- From: Tilghman Lesher [mailto:tilghman@mail.jeffandtilghman.com] Sent: Wednesday, May 21, 2003 2:27 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] gastman segmentation fault when pressing 'enter' in a command win dow On Wednesday 21 May 2003 01:22 pm, Alex Zarubin wrote:

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: z Type: application/octet-stream Size: 24104 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030528/cce0a468/z.obj

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

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 --------------

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

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: p-c.gz Type: application/octet-stream Size: 11364 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030716/396b706e/p-c.obj

Hello, Caller -> PRI -> SIP -> application. If application goes offhook right away, dtmf/audio works fine in both directions. If application, before going offhook (sending OK) plays a message and wants dtmf/voice from the caller, then caller can hear this message but his dtmf/voice don't reach application. Any way to configure it? Thank you. Alex Zarubin -------------- next

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

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

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, It is important for some applications (voicemail and the like) to control play volume and speed if instructed by the caller. The question is: how to do it the right way with asterisk. As far as volume - adding txgain and rxgain to the call structure[s] and setting gains from there (instead of taking global values from config files) looks like an OK solution. As far as speed - the easy way

Hi, We have no problems sending to HP and Panasonic fax machines in the office. We do have problems when we try to send faxes to services supporting fax, i.e. J2 or our UC platform. The receiving side doesn't recognize fax. To send a fax we drop into /var/spool/asterisk/outgoing: Channel: Zap/g1/<fax number> MaxRetries: 0 WaitTime: 20 Context: webley_txfax Extension: txfax_ext

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

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, 1-2 sec pauses happen while * plays (streams) messages/prompts. We get reports about that from users and experience it ourselves randomly. Cannot reproduce it for debugging though, so need to figure out some other ways to fix it. 1. It's not silence recorded within or pauses between audio files 2. It's not load related - can happen with no load at all 3. We use decent boxes - dual

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

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