Re: [asterisk-dev] IAX2 still broken
Thanks to Tim Panton, the following were found in 1.4svn IAX, and submitted to bugtracker http://bugs.digium.com/view.php?id=8273 --FROM TIM- IAX2 calls go silent in 1.4 It looks like a meta (trunk) frame, but it doesn't have Meta Command or Cmd data set to valid values. The length is plausible for a Meta frame carrying a single (G.711) packet. Details in the bugtracker. On 2 November 2006 14:10, Tim Panton wrote: On 2 Nov 2006, at 04:31, Anton wrote: Again, the OLD issue - after a while - IAX becomes 1way-or-no-audio operation. Any suggestion or anyone wants to take a look? I'd like to work with you on this, have you ever managed to capture a packet trace with ethereal whilst it is silent ? Are there voice mini-frames flowing? iax2 debug doesn't cut it as it ignores mini-frames and I suspect it is miniframes that are the problem. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
RE: [asterisk-dev] dynamic queue members get stuck with status 'onhold'
It is set as peer, as below. I have tried with friend and that did not work either. Is there anything I missed below? I was using the agentcallback with 1.2 but that crashes with in 1.4 beta3 (and 2) with a seg fault when you try and transfer a call that has come in from a queue. Transferring seems to be fine with dynamic members except for the on hold status afterwards. [7002] type=peer context=staffextraacc secret=7002 host=dynamic dtmfmode=rfc2833 canreinvite=no callerid=Agent 1 7002 disallow=all allow=g729 Thanks, Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: 02 November 2006 13:43 To: Asterisk Developers Mailing List Subject: Re: [asterisk-dev] dynamic queue members get stuck with status 'onhold' On 11/2/06, Dean Bath [EMAIL PROTECTED] wrote: I have an issue with 1.4 beta3 with dynamic queue members, I also pulled the latest SVN and that did the same. If a member takes a call from the queue and either places the call on hold or transfer it to another phone, the status of that member changes to 'On Hold'. So when the next call comes in, it will not call that member as it thinks they have a call on hold. If they do not place the call on hold or transfer all is fine and they can accept the next call. The only way I have found to remove the 'on hold' status is to completely restart asterisk, but it goes back to on hold the next time they transfer a call. I'm running Asterisk 1.4 beta3 running on Debian 2.6.17.3 with Polycom 501 phones. Are your SIP peers setup as type=friend or type=peer? If not type=peer, please set them to that and try again. We're discovering type=friend with SIP UA's that will eventually become members of the queue is becoming very problematic and we'll have to do something more to address that before 1.4 is released. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] asterisk disappeared! any thoughts?
Hi Folks,All of sudden asterisk disappeared(running asterisk process died) and it didn't generate the core. We are very sure that we started asterisk with -g option. When we looked at the logs of asterisk nothing unusal except the messages those will get when we stop using stop now command. We looked at history of the linux, nobody didn't connect to the asterisk with remote option(asterisk -r).Did anybody experience this type of situation? any ideas/thoughts will be appreciated.Thanks in advance. -- Ramu Yadav ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Re: oej: branch 1.2 r46899 - /branches/1.2/channels/chan_sip.c
In article [EMAIL PROTECTED], svn-commits@lists.digium.com wrote: Author: oej Date: Thu Nov 2 09:15:06 2006 New Revision: 46899 URL: http://svn.digium.com/view/asterisk?rev=46899view=rev Log: Don't overwrite flags in the packet Modified: branches/1.2/channels/chan_sip.c Modified: branches/1.2/channels/chan_sip.c URL: http://svn.digium.com/view/asterisk/branches/1.2/channels/chan_sip.c?rev=46899r1=46898r2=46899view=diff == --- branches/1.2/channels/chan_sip.c (original) +++ branches/1.2/channels/chan_sip.c Thu Nov 2 09:15:06 2006 @@ -1289,7 +1289,8 @@ pkt-next = p-packets; pkt-owner = p; pkt-seqno = seqno; - pkt-flags = resp; + if (resp) + ast_set_flag(pkt-flags, FLAG_RESPONSE); pkt-data[len] = '\0'; pkt-timer_t1 = p-timer_t1;/* Set SIP timer T1 */ if (fatal) This doesn't compile on 1.2: chan_sip.c: In function `__sip_reliable_xmit': chan_sip.c:1293: error: invalid type argument of `-' chan_sip.c:1293: warning: type defaults to `int' in declaration of `__p' chan_sip.c:1293: error: invalid type argument of `-' chan_sip.c:1293: warning: comparison of distinct pointer types lacks a cast chan_sip.c:1293: error: invalid type argument of `-' Haven't tried it on 1.4 or trunk, but my first thought is: does ast_set_flag exist in 1.2? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: Re: [asterisk-dev] Re: why 'o' (preserve original callerid) is not default in app_dial.c ?
On Thu, 2006-11-02 at 08:16 +0200, Stephen Davies wrote: On 01/11/06, John Lange [EMAIL PROTECTED] wrote: I think Asterisk could stand to have a (much?) expanded set of internal variables accessible in the dialplan. Just one simple example would be the username of the calling party (when available). Doing dialplan logic based on callerid is sometimes very problematic. I agree with both your points. Nevertheless, it is easy to get channel variables set on the channel for things like the username. For instance, the setvar option in a SIP peer/user/friend entry allows you to pass on the peer that the call came from. (And/or, accountcode). Would you mind expanding on that a little? Perhaps I'm missing something obvious? Please post an example if you can. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] asterisk disappeared! any thoughts?
On Thursday 02 November 2006 10:09, Ramu Yadav wrote: All of sudden asterisk disappeared(running asterisk process died) and it didn't generate the core. We are very sure that we started asterisk with -g option. You also need to run 'ulimit -c unlimited' prior to running asterisk. Asking Asterisk to dump core on failure doesn't help much when the size of core files are limited to 0. -- Tilghman ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Bug id 8233
The recent flurry of housekeeping (which is good) keeps pushing this bug down the list, possibly keeping it from getting noticed. The fix is trivial and restores the process of adding the a=ptime: attribute to the SIP SDP. I did not attach a patch for two reasons: 1. It is a 2-liner 2. My deveploment platform is tied up testing 1.4b3 I hate to use the -dev list to get a bump, but I'd hate more for such a trivial and obvious fix to be overlooked for 1.4b4 or worse the 1.4.0 release. Could a bug marshall or developer with commit access give it a quick peek? Thanks, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Bug marshals - won't you take a look at 8188
On Thu, 2 Nov 2006, Stephen Davies wrote: I posted up a change that gets chan_iax2 to log jitter buffer stats into the logs regularly during active calls.I've also responded to the bug (#8188), but I'll repost my comments below as they're likely to get a greater audience of developers: Interesting... John Todd and I were discussing something similar at AstriCon, but in a more generic sense. What we'd like to see written is CQDR -- Call Quality Detail Records. Think of them as sitting side-by-side the CDR records, but showing information about the Call Quality. In the case of IAX, we get all kinds of useful information, which we don't currently capture (and your patch logs to the verbose log). In the case of SIP, we have RTCP reports (as well as call quality reports in the BYE messages of some endpoints). Wouldn't it be cool to see all this information logged to a CQDR table?-Jared ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] auto hangup problem
This is a -users, not -dev question, but On Thu, 2 Nov 2006 19:54:03 +0800, Ma Zhiyong wrote I dial out then goto a macro to send fax to callee. And I limit the call to 2 minites. But the connnection can't hangup after 2 minites. It is hungup after the fax has sent completely. I want to use auto hangup for prepaid purpose. So how can I resolve this? Thank you. Dial (CAPI/g1/, 30, L(6)M(sendfax^xx.sff)) You should set the timeout inside the Macro as it is running at the called channel SET(TIMEOUT(absolute)=60) Note: the timeout is in seconds here using L() with a single parameter is converted to S() actualy, but still not checked before the Macro is completed Asterisk 1.2.10 capi_chan-0.7.1 diva server ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Bug marshals - won't you take a look at 8188
On 11/2/06, Jared Smith [EMAIL PROTECTED] wrote: On Thu, 2 Nov 2006, Stephen Davies wrote: I posted up a change that gets chan_iax2 to log jitter buffer stats into the logs regularly during active calls. I've also responded to the bug (#8188), but I'll repost my comments below as they're likely to get a greater audience of developers: Interesting... John Todd and I were discussing something similar at AstriCon, but in a more generic sense. What we'd like to see written is CQDR -- Call Quality Detail Records. Think of them as sitting side-by-side the CDR records, but showing information about the Call Quality. In the case of IAX, we get all kinds of useful information, which we don't currently capture (and your patch logs to the verbose log). In the case of SIP, we have RTCP reports (as well as call quality reports in the BYE messages of some endpoints). Wouldn't it be cool to see all this information logged to a CQDR table? Jared - Totally agree with you here. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] VM breakage in trunk r47021
Hi List Im a fairly long time user, first time reporter, so Im unsure of the protocol. Apologise if this is the wrong list for this. However, Im just checking out SVN-trunk-r47021 as digiums G729 drivers dont seem to work with the beta release. Everything seems to be behaving well, apart from voicemail. Individual voicemail is fine I can leave a message on someones phone, and its saved as expected, and emailed correctly. However, I have several group mailboxes, which are essentially used to present a single message for a group of extensions and of course has the benefit of being easily updatable via the usual comedian interface. Typical layout is: In [acontext] in voicemail.conf ; Additional holding box for group message 199 = 199,Group Mailbox,,,delete=1 And then in extensions.conf exten = 199,1,VoiceMail([EMAIL PROTECTED][EMAIL PROTECTED] (etc) |u) exten = 199,2,Hangup So the idea being, the voicemail is temporarily stored in 199, then copied to everyone elses box, then deleted. Works perfectly in everything up to SVN (including beta3). So onto the problem: In this trunk, I get the following: -- x=0, open writing: /var/spool/asterisk/voicemail/acontext/103/tmp/JINAMD format: wav49, 0x9d75670 -- x=1, open writing: /var/spool/asterisk/voicemail/ acontext /103/tmp/JINAMD format: gsm, 0x9d75988 -- x=2, open writing: /var/spool/asterisk/voicemail/ acontext /103/tmp/JINAMD format: wav, 0x9d75c58 -- User hung up [Nov 3 00:22:55] NOTICE[8455]: app_voicemail.c:2541 copy_message: Copying message from 103@ acontext to 101@ acontext [Nov 3 00:22:55] WARNING[8455]: app_voicemail.c:1737 base_encode: Failed to open log file: /var/spool/asterisk/voicemail/acontext /101/INBOX/msg0001.WAV: No such file or directory == Spawn extension (davesextensions, 199, 1) exited non-zero on 'SIP/101-09d6f4b8' following in the log: Everyone in the list gets an email, but with an empty (270 byte) file, and nobodys voicemail box actually has the message! Note that there is only ever one copy message, but several Failed to open log file: (one for each user) messages. I am reporting it here, as its changed behaviour from the beta3, and the system under test has otherwise stayed exactly the same. I was wondering if somehow perhaps the delete instruction for the group box is somehow being executed before the copy has finished? Best Regards, Dave Bath ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] VM breakage in trunk r47021
David Bath wrote: I’m a fairly long time user, first time reporter, so I’m unsure of the protocol. Apologise if this is the wrong list for this. Please include all of these details of your problem on a bug report on http://bugs.digium.com/. That way, we can make sure it gets looked at and fixed. -- Russell Bryant Software Engineer Digium, Inc. begin:vcard fn:Russell Bryant n:Bryant;Russell org:Digium, Inc. adr:;;150 West Park Loop;Huntsville;AL;35806;USA email;internet:[EMAIL PROTECTED] title:Software Engineer tel;work:+1-256-428-6000 x-mozilla-html:FALSE url:http://www.digium.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
