gmane.comp.telephony.pbx.asterisk.devel

Re: res_jabber call backs

John Todd — 2008-08-21T16:07:35

I believe there is some work being done on Asterisk receiving messages and processing them, but perhaps not exactly what you're looking for (yet?) Maybe you could add those features for presence or event callback. It sounds like you're talking about using XMPP as an API - or am I misunderstanding your question? As a side note, take a look here at the work that Philippe Sultan is doing: http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/ http://lists.digium.com/pipermail/asterisk-users/2008-June/213525.html This is _really_ interesting stuff, and creative use of XMPP combined with Asterisk should allow all sorts of IM-based call transactions. I'd encourage anyone with interest in UC work to take a look at this branch, test the apps, and report your results so this can be merged into mainline if it seems stable and worthwhile. Next thing I'd like to see: XMPP events being received unsolicited and starting a dialplan event sequence with regexp parsing of the IM strings, send

Re: 56k SS7 using TE122.

Matthew Fredrickson — 2008-08-21T01:52:19

They can be T1 based as well. In any case, this is a question that is being handled on the Asterisk-ss7 list, so please tune in there for more info :-) Matthew Fredrickson Digium, Inc. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: 56k SS7 using TE122.

Alex Balashov — 2008-08-21T01:26:59

I think 56K A-links are DDS, not T1-based. Mark A Jenks wrote:

sip notify to reset cisco 79x1 phones

Pavel Jezek — 2008-08-20T22:38:19

Hi, if anyone have interest to implement into sip_notify mechanism to also send custom sip message bodies to phones... I'm attaching packet dump, that callmanager6 use to reset java based phones (79x1) currently, we are able to put sip header, using sip_notify.conf: Event: service-control Content-Lenght: 82 but callmanager also use message body, where is placed actual action to reset or restart phones... action=reset\n _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

56k SS7 using TE122.

Mark A Jenks — 2008-08-20T21:17:06

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Re: specific storage directory for CDR data instead of LOG_DIR

Caio Begotti — 2008-08-20T16:05:40

On Wed, 20 Aug 2008 10:13:25 -0500 Russell Bryant digium.com> wrote: That's even better, awesome :)

Re: specific storage directory for CDR data instead of LOG_DIR

Russell Bryant — 2008-08-20T15:13:25

http://bugs.digium.com/view.php?id=12876 Ready for testing. :)

Re: n_way conference call

Sean Bright — 2008-08-20T15:13:27

This mailing list is for topics related to the development of Asterisk. Your question is more appropriate for the asterisk-users mailing list and has a higher probability of getting answered there. Good luck,

Re: n_way conference call

Russell Bryant — 2008-08-20T15:12:12

Please try your question on the asterisk-users mailing list, as that is the most appropriate place to ask this type of question. Good luck,

Re: specific storage directory for CDR data insteadof LOG_DIR

David A. Bandel — 2008-08-20T15:08:18

Can you pass the CDR logging to syslog? That way, the log can be on another machine. Most embedded devices are real estate poor. Seems to me that would be the best way to handle it. Allow CDR to log to syslog so it could log to say local6 on another machine. Ciao, David A. Bandel

specific storage directory for CDR data instead ofLOG_DIR

Caio Begotti — 2008-08-20T14:37:16

Morning! Someone at #asterisk-br on IRC asked how to storage CDR data somewhere else because he was running Asterisk in an embedded device with no persistent logging (on RAM). At first I thought about using the spool dir, but it seemed hackish because you might care about CDR and still don't use monitoring or voicemail etc. I've added a new ast_config_AST_CDR_DIR for a quick test and it did the trick. If it's not found then Asterisk will still use ast_config_AST_LOG_DIR for that as a fallback and no, it's not a default value in my test even for new instalattions, it's only an alternative path inside asterisk.conf. Is that acceptable or does anyone have anything against it? I still need to write a patch for cdr/cdr_sqlite*.c and cdr/cdr_custom.c. I only got it working for the simple cdr-csv backend, that's why I wanted to ask for input before going ahead. []s

n_way conference call

ameur brahim — 2008-08-20T14:24:20

Re: Possible Deadlock?

Moises Silva — 2008-08-19T22:30:07

oops, sorry about that, Russell is correct, DEBUG_THREADS is the required option to compile with "core show channels" support. Select it by doing: make menuselect 10. Compiler Options -> 3. DEBUG_THREADS Moy

res_jabber call backs

Ruddy Gbaguidi — 2008-08-19T18:24:18

Hi all, As I see, in the asterisk 1.4, jabber is designed to send messages, receive answers and display them in the cli. I think we should add the feature that any asterisk module can register some callbacks for a specific jabber user. Then, from any module, we can request the client using his username and register callbacks on message, on presence etc... So, as soon as a message will be receive, the call back will be fire. What do you think about that ? _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: Possible Deadlock?

Moises Silva — 2008-08-19T17:20:19

If I were you, I'd compile Asterisk with DEBUG_CHANNEL_LOCKS, reproduce the issue and execute command "core show locks", this will likely be easier to debug. Moy On Tue, Aug 19, 2008 at 11:46 AM, Norman Franke myasd.com> wrote:

Re: Possible Deadlock?

Russell Bryant — 2008-08-19T17:25:34

Minor correction: DEBUG_THREADS, not DEBUG_CHANNEL_LOCKS

Possible Deadlock?

Norman Franke — 2008-08-19T16:46:10

I had a deadlock today using 1.4.21.1. I had asterisk dump core to explore in GDB. Thread 32 is waiting for the "channels" lock held by 21 while holding a channel lock. Thread 21 is then waiting for a lock on a channel held by 32 while holding a lock on "channels". However, I can't figure out how thread 32 got the lock in the first place. Perhaps from this in the log? [Aug 19 10:55:26] VERBOSE[25324] logger.c: -- Got SIP response 480 "Temporarily Unavailable" back from 172.16.22.30 And it never released the lock? I see in chan_sip.c:12936 /* XXX Locking issues?? XXX */ This has only happened once recently. Thread 32 (process 25324): #0 0xffffe410 in __kernel_vsyscall () #1 0xb7edd56e in __lll_mutex_lock_wait () from /lib/tls/i686/cmov/ libpthread.so.0 #2 0xb7eda179 in _L_mutex_lock_141 () from /lib/tls/i686/cmov/ libpthread.so.0 #3 0xb786a918 in ?? () #4 0x0807d514 in ast_cdr_start (cdr=0x815a148) at cdr.c:689 #5 0x0807fd3f in ast_mutex_lock (pmute

Re: sip_devicestate and queue problems

Russell Bryant — 2008-08-19T16:07:32

It's waiting on the next release of Zaptel and the release of DAHDI, which are currently being tested. It should be this week unless something goes wrong ...

Re: sip_devicestate and queue problems

Daniel Ferrer — 2008-08-19T16:02:41

Ok, I'm testing version SVN-branch-1.4-r138685, at the moment all is ok. I see that there is no DNS query anymore, because function ast_setstate now breaks SIP channel name string with '-' separator, so never called with entire channel name. Waiting for 1.4.22 to came up... How long do you estimate ? Thanks! bye daniel Russell Bryant escribió: _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: ExtensionStatus

Sean Bright — 2008-08-19T15:20:44

Feel free to post them wherever, just saying that they are more likely to stay on the radar in the bug tracker. Philipp Kempgen wrote:

Re: ExtensionStatus

Philipp Kempgen — 2008-08-19T14:38:43

Sean Bright schrieb: When it's about minor issues like this one I'm not sure whether to post it on the mailing list or on the bug tracker or not to report them at all. But ok, Mantis - next time. Grüße, Philipp Kempgen