[asterisk-users] Zombie DAHDI FXO channels
Dear listers, I'm facing a puzzling situation with Digium TDM2400 card (12 FXO / 12 FXS). Once a day or so we detect 1 or 2 zombie FXO channels. These can be either outbound or inbound calls. I thought this could be related to obsolete DAHDI or Asterisk versions, so I upgraded to 2.4.0 and 1.6.2.15 respectively (OS: Ubuntu 10.04 64 bits). To no avail; the zombie channels keep showing up. Those zombies appear to be hangup calls not being recognized as such by DAHDI. When we see existing calls with suspicious long durations (20 minutes or more) we spy those channels, and pretty often they show no audio at all. Following are a few logs: - inbound call ending up as zombie: http://pastebin.com/NtS8CiMa - outbound call ending up as zombie: http://pastebin.com/nfY707Ap, you can see that starting at line 145 the channel is being hanged up on the console (hangup request DAHDI/x-y) Any ideas? Alex Saavedra -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi-channels.conf for Digium TDM2400
Hello everyone, I have noticed thar our dahdi-channels.conf has some repeating directives, for instance for channel 2 (FXO) we have these settings: ;;; line=2 WCTDM/0/1 FXSKS signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 2 callerid= group= context=default As you can see, a few directives are repeated (callerid, group, context). This was generated by DAHDI tools, and since it's working I didn't want to change it. Is it safe to remove them? Thanks in advance, Alex Saavedra -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi-channels.conf for Digium TDM2400
Gerald, Thank you for the explanation. Glad I asked... Alex Saavedra On Wed, Dec 22, 2010 at 12:40 PM, Gerald A geraldabli...@gmail.com wrote: Hi, On Wed, Dec 22, 2010 at 9:49 AM, Alex Saavedra a...@masterline-logistics.com wrote: I have noticed thar our dahdi-channels.conf has some repeating directives, for instance for channel 2 (FXO) we have these settings: ;;; line=2 WCTDM/0/1 FXSKS signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 2 callerid= group= context=default As you can see, a few directives are repeated (callerid, group, context). This was generated by DAHDI tools, and since it's working I didn't want to change it. Is it safe to remove them? Short Answer: NO!! Longer Answer: The settings all apply to channels, which are defined by the channel = 2 directive. If I'm remembering correctly, the channel is set at the end of the Stanza, not at the beginning. So, your blank callerid and group would apply to your next channel directive (3?). Now, I remember reading there is a way to flip the channel definition bit (channel = XX) to the top of the stanza, but can't recall. Now, if in between two channel definitions you have repetition, it might be ok to trim things up, as long as it has the right information -- the last setting is the effective one. And the bit that starts ;;; is a comment, which is actually ignored by asterisk. Hope this helps, Gerald. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrading DAHDI and Asterisk
Hello, I just upgraded DAHDI from 2.3.0 to 2.4.0, both dahdi-linux and dahdi-tools (Ubuntu 10.04 64 bits). It took a few minutes, and it was straightforward. Everything is working. Now I'm preparing to upgrade Asterisk from 1.6.2.7 to 1.6.2.15. I made a backup of configuration files, codec licenses and CDR. Is there something else I should be aware of before upgrading? Thank you, Alex Saavedra -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless Desktop VoIP Phone?
Grandstream GXV3140 has a WiFi USB adapter. Alex Saavedra On Fri, Dec 17, 2010 at 11:40 AM, Matt mhop...@gmail.com wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video codecs: H263 H264
Jonas, Sorry for the late response. In fact I don't rely on Asterisk settings for video codecs, I rather prefer controlling these in specific devices. The reason being that I experienced video codecs priorities were not always respected by client devices. What's even funnier, for GXV3140, if I leave H264 only, I can still receive calls using H263+. Thus I guess the answer to your question is yes. It seems like both Asterisk and those devices are only applying pass-through strategy for video traffic. When using softpones (Linphone) I can enable H264, H263+ and H263. The preferred codec seems to be used according to this order at handshake time. HTH, Alex Saavedra On Fri, Dec 10, 2010 at 3:41 AM, Jonas Kellens jonas.kell...@telenet.bewrote: On 12/08/2010 02:48 PM, Alex Saavedra wrote: Jonas, I've been using H.264 and H.263+ with a few Grandstream GVX3140. When using H.264 the image quality was better, and required bandwidth appeared lower compared with H.263+. In fact H.264 is expected to consume less bandwidth for as much as 50% (according to Wikipedia). Note however that when configuring bandwidth limits inside Asterisk (maxcallbitrate directive) I saw no effect on client devices. Only client configuration succeeded (v 1.6.2.7 here). I kept H.263+ only for compatibility reasons when using X-Lite softphone (no support for H.264). Regards, Alex Saavedra Do you use H263 and H264 together on a peer ? Because when I define : allow=h263 allow=h264 there is no video. I can only allow 1 video-codec on a peer... Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video codecs: H263 H264
Jonas, I've been using H.264 and H.263+ with a few Grandstream GVX3140. When using H.264 the image quality was better, and required bandwidth appeared lower compared with H.263+. In fact H.264 is expected to consume less bandwidth for as much as 50% (according to Wikipedia). Note however that when configuring bandwidth limits inside Asterisk (maxcallbitrate directive) I saw no effect on client devices. Only client configuration succeeded (v 1.6.2.7 here). I kept H.263+ only for compatibility reasons when using X-Lite softphone (no support for H.264). Regards, Alex Saavedra On Wed, Dec 8, 2010 at 7:36 AM, Jonas Kellens jonas.kell...@telenet.bewrote: Hello list, what is the difference between these 2 codecs ? What codec to choose if bandwith is an issue ? (like in most cases I guess) Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk segfault in dmesg
Hello everyone, We are using an Asterisk server (Ubuntu 10.04 64 bits, Asterisk 1.6.2.7, TDM2400 - 12 FXS/12 FXO), but I noticed the following dmesg alert: asterisk[4687]: segfault at 60 ip ** sp ** error 4 in asterisk[40+187000] The server is running without troubles. I know I should worry, just don't know where to start. Any advices? Alex Saavedra -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk segfault in dmesg
Paul,
Thank you for your reply. From the backtrace it seems like core restart
when convenient caused the segmentation fault, in which case I shouldn't be
really concerned (...should I?). I'm attaching the backtrace.
I issue this command from time to time, because our TDM2400 seems to hold
zombie DAHDI channels on FXO ports (DAHDI v 2.3.0), never putting them on
hook again, and resulting in wrong CDR entries reporting extremely long
conversations.
Once you confirm that the segfault error is related to core restart when
convenient I will post a question regarding those zombie DAHDI channels.
Best regards,
Alex Saavedra
On Wed, Dec 8, 2010 at 10:49 AM, Paul Belanger pabelan...@digium.comwrote:
On 10-12-08 09:31 AM, Alex Saavedra wrote:
The server is running without troubles. I know I should worry, just don't
know where to start. Any advices?
If you didn't notice downtime from Asterisk, your likely using the
safe_asterisk script. It has some inline documentation you should read.
Also, once you located your core file, generate a backtrace[1] and
attached it to this message.
[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com http://asterisk.org
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Core was generated by `/usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c'.
Program terminated with signal 11, Segmentation fault.
#0 ast_sched_thread_get_context (st=0x0) at sched.c:132
132 {
#0 ast_sched_thread_get_context (st=0x0) at sched.c:132
#1 0x7f42259b87fe in iax2_destroy_helper (pvt=0x7f4210030698) at
chan_iax2.c:1643
#2 0x7f42259bdff0 in iax2_destroy (callno=1688) at chan_iax2.c:3230
#3 0x7f42259be98e in __unload_module () at chan_iax2.c:13654
#4 0x004a2f21 in ast_module_shutdown () at loader.c:470
#5 0x00434575 in quit_handler (niceness=2, safeshutdown=1, restart=1,
num=value optimized out) at asterisk.c:1597
#6 0x00434f65 in handle_restart_when_convenient (e=value optimized
out, cmd=value optimized out,
a=value optimized out) at asterisk.c:1916
#7 0x00464481 in ast_cli_command_full (uid=value optimized out,
gid=value optimized out, fd=49,
s=0x7f42163e4930 core restart when convenient ) at cli.c:2299
#8 0x004646c0 in ast_cli_command_multiple_full (uid=value optimized
out, gid=value optimized out, fd=49,
size=30, s=0x7f42163e4ba0 core restart when convenient ) at cli.c:2322
#9 0x0043541a in netconsole (vconsole=0x7afca0) at asterisk.c:1231
#10 0x0050d4ca in dummy_start (data=value optimized out) at
utils.c:968
#11 0x7f423454b9ca in start_thread () from /lib/libpthread.so.0
#12 0x7f4234d9c70d in clone () from /lib/libc.so.6
#13 0x in ?? ()
#0 ast_sched_thread_get_context (st=0x0) at sched.c:132
No locals.
#1 0x7f42259b87fe in iax2_destroy_helper (pvt=0x7f4210030698) at
chan_iax2.c:1643
__tmp_context = value optimized out
__PRETTY_FUNCTION__ = iax2_destroy_helper
#2 0x7f42259bdff0 in iax2_destroy (callno=1688) at chan_iax2.c:3230
pvt = 0x7f4210030698
owner = 0x788470
__PRETTY_FUNCTION__ = iax2_destroy
#3 0x7f42259be98e in __unload_module () at chan_iax2.c:13654
con = value optimized out
x = 1688
#4 0x004a2f21 in ast_module_shutdown () at loader.c:470
__list_next = 0x7f422c033c50
__list_prev = 0x7f422c03b300
__new_prev = value optimized out
mod = 0x7f422c05daa0
somethingchanged = -64
#5 0x00434575 in quit_handler (niceness=2, safeshutdown=1, restart=1,
num=value optimized out) at asterisk.c:1597
filename = '\000' repeats 79 times
s = 4589321
e = 49
x = value optimized out
#6 0x00434f65 in handle_restart_when_convenient (e=value optimized
out, cmd=value optimized out,
a=value optimized out) at asterisk.c:1916
No locals.
#7 0x00464481 in ast_cli_command_full (uid=value optimized out,
gid=value optimized out, fd=49,
s
[asterisk-users] DAHDI channels not hanging up FXO
Hello everyone, I've been facing an odd issue with Digium TDM2400 card (12 FXO / 12 FXS). From time to time a FXO port will not hangup, although the extension and POTS line already did. It's like a zombie channel. When this happens we issue hangup request DAHDI/x-x. Anyone faced similar issue? Current configuration: Ubuntu 10.04 64 bits, Asterisk 1.6.2.7, DAHDI 2.3.0. Thank you, Alex Saavedra -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
