search for: zarubin

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2003 Sep 16
3
Adpcm, 6KHz codec
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
2003 Jun 09
3
Setting local IP address for the RTP port
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
2003 May 22
3
nfas on T400P?
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
2004 Mar 30
2
SoftFAX/spandsp - txfax
...ne 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040330/5735e68b/attachment.htm
2003 Jun 09
7
Dual T400P, SMP, performance issues
...RR: 0 MIS: 0 Not sure what went wrong. Any suggestions on how to work with 2 T400P in a box (without hurting performance) and how to get advantage of SMP for Asterisk would be appreciated. Any known Linux kernel related issues (2.4.20-13.7smp #1 SMP for BOX3 )? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030609/11a3bfd0/attachment.htm
2003 Jun 10
1
SIP sdp o= and c= fields
...elds 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030610/27f5d2a1/attachment.htm
2004 May 13
1
poll vs select in channel.c
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: http://lists.digium.com/pipermail/asterisk-users/attachments/20040513/be61bfc0/attachment.htm
2003 Jul 31
3
Mutex problem in sip?
...ice or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy Thank you. Alex Zarubin Webley Systems, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030731/3c7f4565/attachment.htm
2003 Apr 02
12
segmentation fault
...: Unknown error 500 Apr 1 23:30:24 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: Read on 86 failed: Unknown error 500 Question: What do I do to give you more info? Should I issue 'ulimit -c unlimited' to get a coredump? Are there any flags/modes to set? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030402/c8261f5f/attachment.htm
2023 Apr 10
1
Setting PJSIP header from AMI
...gt;setVariable('__SIPADDHEADER51',"Identity: $identity"); // $identity contains generated by 3rd party header Is there anything similar for $action = new OriginateAction("PJSIP/...."); ??? that would work for PJSIP? Any suggestions are appreciated. Alex Zarubin/TTH -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230410/9629d57a/attachment.html>
2003 Apr 15
1
dialplan problems cvs 04-15-03
...es 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 changing contexts etc. didn't help. 'Show dialplan' outputs the right stuff. Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030415/5404c9c6/attachment.htm
2003 May 28
1
SIP INVITE and ACK go to different ports
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2003 Jun 18
1
Integration with external ASR engines
...ine (for example, main ASR players Nuance and Speechworks will be happy with RTP) - can be done via RTP forking. The other part is stopping the prompt when speech is detected (barge-in implementation). One more thing is adding the related commands to agi. Any comments are welcome. Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030618/4dc30c09/attachment.htm
2003 Jul 16
1
Back-to-back connected boards load test
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2003 Dec 01
2
PRI maintenance commands
...sk platform uses SERVICE commands for this purpose to put B-channels into 'out-of-service'/'maintenance' or 'in-service' state. What is (would be) the asterisk solution? The current libpri doesn't send any SERVICE commands, just acks the incoming ones. Thank you. Alex Zarubin Webley Systems, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/2454f69a/attachment.htm
2005 Feb 16
1
Inter-asterisk conferencing delays - IAX2 configuration problem?
...rence? [general] port=5036 disallow=all allow=ulaw tos=lowdelay [iaxcontext] type=user context=default_iax deny=0.0.0.0/0.0.0.0 permit=10.50.4.0/255.255.255.0 ;trunk=yes ; Use IAX2 trunking with this host trunk=no ; Do not use IAX2 trunking with this host Thanks. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050216/6695e9c4/attachment.htm
2013 Oct 21
0
Asterisk 1.8.24.0 Now Available
...g is a sample of the issues resolved in this release: * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- Fix Not Storing Current Incoming Recv Address (Closes issue ASTERISK-22071. Reported by Alex Zarubin) * --- Fix Segfault When Syntax Of A Line Under [applicationmap] Is Invalid (Closes issue ASTERISK-22416. Reported by CGI.NET) * --- Tolerate presence of RFC2965 Cookie2 header by ignoring it (Closes issue ASTERISK-21789. Reported by Stuart Henderson) * --- Fix Not Storing Current Inco...
2013 Oct 21
0
Asterisk 1.8.24.0 Now Available
...g is a sample of the issues resolved in this release: * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- Fix Not Storing Current Incoming Recv Address (Closes issue ASTERISK-22071. Reported by Alex Zarubin) * --- Fix Segfault When Syntax Of A Line Under [applicationmap] Is Invalid (Closes issue ASTERISK-22416. Reported by CGI.NET) * --- Tolerate presence of RFC2965 Cookie2 header by ignoring it (Closes issue ASTERISK-21789. Reported by Stuart Henderson) * --- Fix Not Storing Current Inco...
2003 May 21
1
gastman segmentation fault when pressing 'enter' in a command win dow
...at gastman.c:1545 #33 0x403b1507 in __libc_start_main (main=0x8050a88 <main>, argc=2, ubp_av=0xbffff874, init=0x804bc14 <_init>, fini=0x8058d68 <_fini>, rtld_fini=0x4000dc14 <_dl_fini>, stack_end=0xbffff86c) at ../sysdeps/generic/libc-start.c:129 (gdb) Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030521/164c416e/attachment.htm
2004 Mar 16
24
Softfax/spandsp
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