IRC log for #asterisk-bugs on 20081204

00:21.28*** part/#asterisk-bugs Deeewayne (n=Deeewayn@nat/digium/x-64c0ca5622bf1b97)
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10:22.34elielM0013998
10:22.36MuffinMan[new] [Asterisk] Channels/chan_iax2 0013998: Bug with iax channel (and perhaps IAXmodem) : randomly crashes asterisk reported by ad (Karma: neutral) http://bugs.digium.com/view.php?id=0013998
10:22.49elielThis issue looks like memory corruption...
11:19.27*** join/#asterisk-bugs caio1982 (i=caio1982@CAcert-br/caio1982)
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14:12.37*** join/#asterisk-bugs lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
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16:25.21shazaumhi file
16:37.09Entomologist*** CLOSED (13952) [Channels/chan_skinny/NewFeature] [patch] Add packet information to debug
16:37.10EntomologistAssigned to: mvanbaak
16:37.10EntomologistReported by: wedhorn  Karma: 18.25
16:37.10Entomologisthttp://bugs.digium.com/view.php?id=13952
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17:09.53*** join/#asterisk-bugs oej (n=olle@ns.webway.se)
17:16.22*** join/#asterisk-bugs flujan (n=flujan@internet.nube.com.br)
17:43.44*** join/#asterisk-bugs Pagautas (n=bigman@ns.voip.ktu.lt)
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18:30.37Entomologist*** CLOSED (13835) [Core/RTP] "RTCP SR transmission error, rtcp halted" logged when SIP call put on hold
18:30.37EntomologistReported by: matt_b  Karma: 0
18:30.38Entomologisthttp://bugs.digium.com/view.php?id=13835
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19:40.20*** part/#asterisk-bugs DagMoller (n=aguirre@unaffiliated/dagmoller)
19:49.59Entomologist*** CLOSED (14009) [Addons/cdr_addon_mysql] [patch] Add static values for db columns on cdr_addon_mysql
19:49.59EntomologistAssigned to: Corydon76
19:49.59EntomologistReported by: markwilkinson  Karma: 2
19:50.00Entomologisthttp://bugs.digium.com/view.php?id=14009
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20:27.02*** join/#asterisk-bugs Skavin (i=kevins@202-180-69-27.me.tranzpeer.net)
20:30.43Skavinhave come across a strange behavoiur of the RPT from asterisk when ptime is set to 30 by the peer
20:31.41Skavinthe peer sends rtp with deltas of 30
20:32.07Skavinasterisk is sending back packets with deltas of 20 then 40 20 40 20 40
20:34.35Skavinpackets are all the correct length its just that asterisk is introducing jitter of 20ms as it sends the packet
20:38.25lmadsenhrmm... interesting
20:38.52lmadsenunfortunately I have nothing in addition to add to your issue other than that
20:40.51Skavinis there somthing ing the config I could have broken or should I log a bug?
20:44.12lmadsenhmmmm... well in the configuration I think you can use the allow=ulaw:30 format to define the amount of ms of audio in each packet
20:44.22lmadsenperhaps if you match it that will cause the issue to be resolved?
21:00.42Skavinno change looks like asterisk is matching the peers ptime because it is initiating the call
21:03.27lmadsenmight be worth it to open a bug then I suppose
21:04.07SkavinOK will do.
21:30.32Entomologist*** CLOSED (12134) [Channels/chan_sip/Video] [patch] add more bandwidth settings to the SDP
21:30.32EntomologistAssigned to: oej
21:30.32EntomologistReported by: klaus3000  Karma: 13.75
21:30.32Entomologisthttp://bugs.digium.com/view.php?id=12134
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21:42.45Entomologist*** CLOSED (12937) [Core/RTP] DTMF package's ssrc number wrong when partical bridge channel
21:42.45EntomologistReported by: gino_he  Karma: 0
21:42.45Entomologisthttp://bugs.digium.com/view.php?id=12937
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21:48.50Entomologist*** CLOSED (11485) [Core/Portability] Alignment issues building on solaris sparc
21:48.50EntomologistAssigned to: kpfleming
21:48.50EntomologistReported by: snuffy  Karma: 63.25
21:48.50Entomologisthttp://bugs.digium.com/view.php?id=11485
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21:54.29lmadsenM12999
21:54.31MuffinMan[new] [Asterisk] Channels/chan_sip/NewFeature 0012999: [patch] SIP channel cannot handle FLASH HOOK INFO message sent by SIP device reported by licedey (Karma: -8) http://bugs.digium.com/view.php?id=12999
21:54.37lmadsenany suggestions on what to do, if anything, with that issue?
21:57.32putnopvutThat sounds like a feature request without a patch to me.
21:58.11putnopvutUnfortunately, the only patch there is one where a license was rejected for whatever reason.
21:58.54putnopvutAlso, it looks like some sort of non-standard INFO package from Broadsoft.
22:12.31lmadsenok, closing
22:12.34lmadsenall I needed to hear :)
22:12.43lmadsenI hadn't noticed the rejected license
22:13.59Entomologist*** CLOSED (12999) [Channels/chan_sip/NewFeature] [patch] SIP channel cannot handle FLASH HOOK INFO message sent by SIP device
22:13.59EntomologistReported by: licedey  Karma: -8
22:13.59Entomologisthttp://bugs.digium.com/view.php?id=12999
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22:18.09*** join/#asterisk-bugs JunK-Y (n=junky@modemcable156.137-20-96.mc.videotron.ca)
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22:26.59jpeelerputnopvut: quick question: when a caller is bridged to the agent and the agent presses * to disconnect, is that setable or hard coded?
22:27.13putnopvutIt depends on the version of Asterisk.
22:27.21jpeeleryuck, ok for 1.4
22:27.27putnopvutfor 1.4, it's hard-coded.
22:27.45putnopvutI think you have to be using 1.6.1 or newer to have the configurable enddtmf option in agents.conf
22:28.27putnopvutI just double-checked, and I am correct.
22:28.37putnopvut1.6.1 is the first version to have the enddtmf option.
22:29.46jpeelerwell i wonder how bad it would be then to hard code a caller waiting for the agent to be able to disconnect with * as well
22:45.22putnopvutjpeeler: well.....if they're using app_dial to call the agent, then they should be able to press * to disconnect already.
22:45.30putnopvutFor app_queue, on the other hand...I'm not as certain.
22:46.17putnopvutActually it appears that pressing * has the same affect for app_queue, too.
22:46.34putnopvutAnd in both cases, they are hardcoded to the '*' key.
22:46.44jpeeleri'm talking about for a caller that hasn't been answered yet, waiting in the queue
22:46.54jpeeleri had to modify it to get that to work
22:47.04putnopvutoh, and who isn't currently calling a member, either, I take it?
22:47.21putnopvutThat can be done already, but not really in an "obvious" way.
22:48.03putnopvutIf you set the 'context' option in queues.conf, then pressing a DTMF digit while waiting in the queue will take you to the extension pressed in the context specified in queues.conf
22:48.12jpeeleri'm a bit hazy with the terminology, just is calling a queue group i guess
22:48.36jpeelerright, which is why adding * was so easy because the code is already checking for digits
22:55.56*** join/#asterisk-bugs Deeewayne (n=Deeewayn@nat/digium/x-b133150c8aeea363)
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23:23.36putnopvutM13256
23:23.37MuffinMan[acknowledged] [Asterisk] Channels/General 0013256: crash if hangup during wait to read reported by travishein (Karma: +2.75) http://bugs.digium.com/view.php?id=13256
23:23.59putnopvutAll right. Problem is that ast_read_generator_actions crashes because it assumes that chan->generator is non-NULL.
23:24.16putnopvutchan->generatordata is non-NULL, but chan->generator is NULL.
23:24.33putnopvutI've been looking through the code trying to figure out how this could happen, but it's not coming to me.
23:24.38putnopvutDoes anyone have any ideas?
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23:41.30Entomologist*** CLOSED (13815) [General] Missing CallerID in CDR when last app is voicemail
23:41.30EntomologistAssigned to: otherwiseguy
23:41.30EntomologistReported by: ssuehring  Karma: 3.5
23:41.30Entomologisthttp://bugs.digium.com/view.php?id=13815
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