similar to: idle SIP channels problem

El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. > > Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred

Hello community, I've been running Asterisk on an embedded device for about six months, and my operation has been largely trouble-free. I'm hoping I could get some help with a minor problem: Every week or three, my PBX gets stuck in a state where it can receive calls, but it becomes completely unable to originate outgoing calls until I do a "sip reload". After doing the SIP

On a production system, running 1.4.17 (compiled from bristuff-0.4.0-test6-xr1) we had this strange issue two times in the last weeks: [2009-01-13 13:58:30] WARNING[1213] channel.c: Fixup failed on channel SIP/2332-081d0108<MASQ>, strange things may happen. [2009-01-13 13:58:30] WARNING[1213] channel.c: Hangup failed! Strange things may happen! [2009-01-13 13:58:30] WARNING[1213]

Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try. 2015-04-07 20:33 GMT-03:00 Alex Villac??s Lasso <a_villacis at palosanto.com>: > El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > > I am trying to collect enough information about an

Hello, I have a local Asterisk server that keep loosing its registration to main Asterisk server. The local asterisk server is on the local subnet, it acts as a client with extension 808. Local server Sip.conf register => 808:password at as2.xxxxx.com registertimeout=20 registerattempts=10 Main Asterisk Server sip.conf [808] type=friend context=sip-phones call-limit=99

Hello every one, I just want to add a soft button to make my soft sip client with idle or busy status. Does any one know what's the event action drive Asterisk to be busy or idle in API event list? Thanks, Johnson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100129/8a08f4b5/attachment.htm

After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports

Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing .... when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not

Hi all, I'm having this weird issue that I can't explain. Maybe someone can explain what is happening. This is a Asterisk install that has been in production for 6+ months. It's version 1.2.10. Couple weeks ago one SIP peer started disappearing randomly. And I mean it simply disappears. One second "sip show peers" shows it, and then it's gone. A simple "sip

Does anybody know if idle-url works for Cisco 79xx using Sip?? If it doesn't work is it a Sip vs?SCCP issue or Asterisk vs CallManager issue?? Thanks Paul

Hopefully I'm not just missing some little detail here. We're trying to set the HANGUPCAUSE on SIP channels to have our softswitch play the proper recording instead of answering the call on Asterisk to play the message. It appears that no matter what the HANGUPCAUSE is set to, Asterisk always just sends "603 Declined". I looked through the source code briefly and it appears

I know over time SIP OPTIONS message handling has changed and I've seen some write ups that seem to indicate that an s extension in the default context is needed now to get them to work. It's probably my error in any case. So, what am I doing wrong or what do I need to do to get the sip ping to work? Bruce Ferrell Just for fun, I created a sip peer called ping at a fixed address

Hi, 1) is there a function like "zap destroy channel" to destroy sip channels? Zap/10-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/-081aee08 (pstn-out s 7 ) Ring Dial Zap/g1/0123456789|50|g Zap/8-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/-081aee08 (pstn-out s

I saw the following lines in the log this morning. From my router logs I see that the connection went down as my ISP was doing maintenance for a few minutes last night. I can understand the external registrations timing out, but why do the phones become unreachable. They are on the internal lan within the same subnet as the Asterisk server. Internal DHCP and DNS was functional. If I had a PRI card

Hi everyone! Everyday some channels go to this situation: # asterisk -rx 'pri show channels'| head -n 32 | grep 'Yes No Idle Yes' PRI B Chan Call PRI Channel Span Chan Chan Idle Level Call Name 1 1 Yes No Idle Yes 1 2 Yes No Idle Yes 1 6 Yes No Idle Yes 1 13 Yes No Idle Yes 1

On Thu, 16 Nov 2017, Quan Xu wrote: > On 2017-11-16 16:45, Peter Zijlstra wrote: > > I really have considered this factor, and try my best not to interfere with > scheduler/idle code. > > if irq_timings code is ready, I can use it directly. I think irq_timings > is not an easy task, I'd like to help as much as I can. It's not a question whether irq_timings code is

https://bugs.freedesktop.org/show_bug.cgi?id=79532 Priority: medium Bug ID: 79532 Assignee: nouveau at lists.freedesktop.org Summary: [NV50] errors with DMA PUSHER and idle channel Severity: normal Classification: Unclassified OS: All Reporter: mattia.b89 at gmail.com Hardware: Other Status:

Hi, I'm using the Perl AGI interface for a prepaid card platform. And sometimes (almost twice an hour), asterisk doesn't detect a call has been hung up. The call is so hung up when the time limit for the call is reached (the corresponding prepaid card is then emptied ...). I've tried to look in the asterisk log files to find anything suspect with these calls, and I've found a

Are these attempts to scam SIP calls through my Asterisk server: [May 13 22:50:41] NOTICE[30888]: chan_sip.c:17295 handle_request_invite: Call from '' to extension '084312297134' rejected because extension not found. [May 14 13:36:35] NOTICE[30888]: chan_sip.c:17295 handle_request_invite: Call from '' to extension '0114312297136' rejected because extension not

On 2017-11-16 16:45, Peter Zijlstra wrote: > On Wed, Nov 15, 2017 at 11:03:08PM +0100, Thomas Gleixner wrote: >> If I understand the problem correctly then he wants to avoid the heavy >> lifting in tick_nohz_idle_enter() in the first place, but there is already >> an interesting quirk there which makes it exit early. > Sure. And there are people who want to do the same for