[asterisk-users] Manager Originate Action and Cancel
I'm using the Originate Action on the Asterisk Manager to place calls between two extensions in async mode. Is there any way to cancel the Originate Action before I get the OriginateResponse action? I'm unable to perform a Hangup because I can't know the channel name before I get the response... thanks in advance! -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel. +(598 2) 707 7687 Fax. +(598 2) 709 4866 / http://www.netlabs.com.uy ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.14 - Chanspy, sound issues.
I upgraded my Asterisk system to version 1.2.14 to check if the sound quality issues I was having with Chanspy in 1.2.7 remained. I'm still getting them, and I'm honestly out of ideas except from RTFS. The called party sounds normally fine, but it's impossible to hear the caller. Sometimes, when the called party is talking, the caller can also be heard. The conversation sounds broken, to the point is almost useless. We don't have any other quality problems beside this. Sound is quite good when making a call or accessing other asterisk services. My setup is as follows: All calls are performed inside a LAN (NOT fully switched...), using SIP and g711. I use SJPhone v1.60 at agents and AT-530 VoIP Phones for the spies. * Intel(R) Pentium(R) 4 CPU 3.00GHz, 1GB RAM, Broadcom Corporation NetXtreme BCM5705_2 Gigabit Ethernet. * Linux foo.bar.com 2.6.9-34.0.2.ELsmp #1 SMP Fri Jul 7 19:52:49 CDT 2006 i686 i686 i386 GNU/Linux * Asterisk 1.2.14-BRIstuffed-0.3.0-PRE-1w built by bachbuilder @ octopus.physik.fu-berlin.de on a i686 running Linux on 2006-12-19 00:11:55 UTC. Is someone else getting this kind of behaviour? Is Chanspy used normally under this conditions on other installations? Any ideas? saludos, -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel. +(598 2) 707 7687 Fax. +(598 2) 709 4866 / http://www.netlabs.com.uy begin:vcard fn:Santiago Aguiar n:Aguiar;Santiago org:;Desarrollo adr:;;Palmar 2548;Montevideo;Montevideo;11600;Uruguay email;internet:[EMAIL PROTECTED] title:NetLabs tel;work:+598 2 7077687 tel;fax:+598 2 7094866 tel;home:+598 2 7075079 tel;cell:+598 99 579739 x-mozilla-html:TRUE url:http://www.netlabs.com.uy/ version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chanspy severe sound problems
Hi everyone! I'm using Asterisk 1.2.7.1 on a CentOS 4 server with 5 - 9 agents and I'm having some issues with the Chanspy application. All the agents are on SIP channels with g711 and all the communications are inside a LAN. When I'm spying a SIP channel, the audio from one of the ends (normally the caller) sounds *extremely* (unusable) choppy, as if it was losing some frames. Sometimes the called party is heard almost perfectly, but there are ALWAYS sound quality issues. The agents do not report any problem, and the audio recorded with the Monitor applications sounds reasonably fine. I'm able to reproduce the problem with any amount of load and it happened also while doing tests with my computer as an Asterisk server. Additional Information: * Asterisk 1.2.7.1 built by test @ ast3 on a i686 running Linux on 2006-04-24 10:52:49 UTC * Linux foo.bar.com 2.6.9-34.0.2.ELsmp #1 SMP Fri Jul 7 19:52:49 CDT 2006 i686 i686 i386 GNU/Linux * Intel(R) Pentium(R) 4 CPU 3.00GHz, 1GB RAM. did anyone encountered the same situation? Google only reported one similar problem without a solution (http://bugs.digium.com/print_bug_page.php?bug_id=7340) any ideas are welcome! thanks a lot! saludos, -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel. +(598 2) 707 7687 Fax. +(598 2) 709 4866 / http://www.netlabs.com.uy begin:vcard fn:Santiago Aguiar n:Aguiar;Santiago org:;Desarrollo adr:;;Palmar 2548;Montevideo;Montevideo;11600;Uruguay email;internet:[EMAIL PROTECTED] title:NetLabs tel;work:+598 2 7077687 tel;fax:+598 2 7094866 tel;home:+598 2 7075079 tel;cell:+598 99 579739 x-mozilla-html:TRUE url:http://www.netlabs.com.uy/ version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF and SIP
hi! I'm having the same problem, I'm connecting through a Planet VIP-450 ITG, and when I send a DTMF code I get a: WARNING: codec_ilbc.c:141 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? I tried using different dtmf settings in sip.conf, but the message is still there. I don't have problems using a softphone... any ideas??? saludos! santiago. Lee Norvall wrote: Hi Just tried that, and still the same with the same error! The spec for the phones includes rfc2833, so I don't think that is it. Rgds -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Justin Carlson Sent: 02 June 2004 19:23 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DTMF and SIP have you tried commenting out the dtmf lines in your sip.conf we had similar problems with our snom 200's and after commenting out the dtmf lines in sip.conf asterisk reload they worked great :-) On Wed, 2004-06-02 at 11:36, Lee Norvall wrote: Hi I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also tried inband) and I get the following error: june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? This means that I cannot get access to voicemail from the handsets !!! Any clues??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Questions
hi everyone! Two days ago we installed asterisk in our labs to do some testing and try the product with a couple of ITGs. Overall, we really loved it! We found it easy to configure and manage, and with good debugging options. There are a couple of questions I would like to ask: a) We had some authentication issues trying to register a Planet ITG with asterisk. Apparently, asterisk ignored the username attribute on the sip.conf entry: [10] type=friend username=foo secret=foosec host=dynamic context=sip-call The ITG was connecting as 'sip:10@ITG-IP' and its md5 was calculated using the specified user 'foo'. However, asterisk was using '10' to calculate the md5, and therefore authentication failed. We don't know if we found a bug or we are doing something wrong ;) (the code in question is in channels/chan_sip.c:3812, were it looks it sends peer-name instead of peer-username, on v0.9.0). b) Is it possible to make asterisk play a file in a codec supported by the client?? We tried to play tt_monkeys, but we got an error when passing from GSM to g723, which is ok, but the client supported g711 also, and I suppose it could be used by asterisk. We added allow=g711 to sip.conf and it worked (however, we had an error if we used allow=all since it tried sending in gsm, which wasn't supported by the ITG). c) We are getting some NOTICE: sched.c:218 sched_settime: Request to schedule in the past?!?! on the CLI, we don't know yet its cause or what it means. thanks a lot for the support! saludos! santiago. netlabs Palmar 2548 Montevideo, Uruguay +(598 2) 707-7687
