[asterisk-users] Asterisk 16.23.0 strange issue where Answer request succeeds and able to perform actions but Asterisk never sent 200 OK to answer call

Some background... We use AMI and AsyncAGI to be able to receive events about calls (and other Asterisk details) and control it from our application. Works great and have about 100 sites (some newer, some older) without issues. I was notified this morning about a customer who had something

Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

On 21-11-16 17:20, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens > wrote: On 21-11-16 15:17, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens

Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

On 21-11-16 19:14, Jonas Kellens wrote: On 21-11-16 17:20, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens > wrote: On 21-11-16 15:17, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 7:05 AM, Jonas

Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

On 21-11-16 17:20, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens > wrote: On 21-11-16 15:17, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens

Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens wrote: > On 21-11-16 15:17, Matthew Jordan wrote: > > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > wrote: > >> Hello >> >> when using Asterisk version 13.12.2 I notice that it takes up to

Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

On 21-11-16 15:17, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > wrote: Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue

Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens wrote: > Hello > > when using Asterisk version 13.12.2 I notice that it takes up to 30 > seconds (sometimes even longer) for a call queue to call its members. > > Example 1 : > > [Nov 21 08:17:57] pbx.c: Executing

[asterisk-users] Asterisk 13.12.2 : strange queue behaviour

Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members. Example 1 : [Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack [Nov 21

[asterisk-users] Dahdi 2.5.0.2 - Strange Warning

Hi, On a recently updated system , I'm now reading lines as this one (never noticed them before): [Dec 19 19:01:52] WARNING[10828]: chan_dahdi.c:11302 pri_dchannel: Ring requested on unconfigured channel 0/0 span 4 My setup is: Asterisk 1.6.1.18 Libpri 1.4.12 Dahadi 2.5.0.2 My card is a

Re: [asterisk-users] Dahdi 2.5.0.2 - Strange Warning

On a recently updated system , I'm now reading lines as this one (never noticed them before): [Dec 19 19:01:52] WARNING[10828]: chan_dahdi.c:11302 pri_dchannel: Ring requested on unconfigured channel 0/0 span 4 My setup is: Asterisk 1.6.1.18 Libpri 1.4.12 Dahadi 2.5.0.2 My card is a

[asterisk-users] Version 1.8 strange expression error

This expression that worked fine in 1.6.2 is returning an error: exten =_X.,n,Set(i=$[${i} + 1]) It needs to add 1 to the value if i. Did I miss something? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Version 1.8 strange expression error

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of CDR Sent: Monday, August 08, 2011 9:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Version 1.8 strange expression error

[asterisk-users] [Fwd: Re: Strange network issue]

Forwarded Message From: Ishfaq Malik i...@pack-net.co.uk To: Mike Diehl mdi...@diehlnet.com Subject: Re: [asterisk-users] Strange network issue Date: Fri, 22 Jul 2011 09:55:53 +0100 On Fri, 2011-07-22 at 02:53 -0600, Mike Diehl wrote: On Friday 22 July 2011 2:42:12 am Ishfaq

[asterisk-users] 1.4.x Strange Vocemail delay

Hi there, I've got the following code (for remote enquiry of the answering machine) in my dialplan: [mailbox] exten = m,1,Set(TIMEOUT(digit)=4) exten = m,2,Set(TIMEOUT(response)=0) exten = m,3,Set(LANGUAGE()=de) exten = m,4,Read(Pin,unavail,4) exten = m,5,capicommand(echosquelch|no) exten =

[asterisk-users] loading misdn.conf strange error regarding out of range

Hello all! I just noticed, that since installing the latest SVN branch (152803), I receive the following error, when loading/reloading the misdn.conf file misdn reload [...] [Nov 3 16:20:37] WARNING[5267]: misdn_config.c:938 _build_general_config: misdn.conf: misdn_init=/etc/misdn-init.conf

[asterisk-users] [SOLVED + EXPLANATION]: Strange ISDN-problem with incoming calls out of the same city

Hi, in december last year I posted the following problem: QUOTE When I dial the number of our client, located in another town, I get a connection to the asterisk server, I can talk to my client or listen to his mailbox. If someone in the town of this client calls him, he gets the ISDN error

Re: [asterisk-users] [SOLVED + EXPLANATION]: Strange ISDN-problem with incoming calls out of the same city

Stefan Guenther wrote: QUOTE When I dial the number of our client, located in another town, I get a connection to the asterisk server, I can talk to my client or listen to his mailbox. If someone in the town of this client calls him, he gets the ISDN error service not available. With

Re: [asterisk-users] Help on strange problem...

Stuart, First update to the latest version of the asterisk you are using. When reporting problems, always mention which version of Asterisk you are using. If the problem still exists, please file a bug report with proper logging as described in the bug guidelines. Thank you. /Olle 16 nov

[asterisk-users] Help on strange problem...

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1

[asterisk-users] unwanted chanspy: strange behaviour

Hi, I have an Asterisk box with a Sangoma PRI on a Debian distro. It may happen that user A tries to call B but for some reason the call drops and A is connected to another call between C and D. User A can only hear what C and D are saying, like a sort of unwanted chanspy. Is there anybody

[asterisk-users] Line drops strange problem(got event On hook)

Hello to all, I have a strange problem with my asterisk. Line drops while i am in a call and without a reason.The line drops no matter if it is a incoming or outgoing call and it happen while i am talking on the phone (no silence detection problem). I tried to debug the situation and the only

[asterisk-users] Cisco 7970 strange Xml , but upgrade success.

When I try to upgrade 7970 phone to sip 8.0.4SR1, Im getting this error all time: Read request for file .loads. Mode octet [16/10 15:14:12.187] File .loads : error 2 in system call CreateFile The system cannot find the file specified. [16/10 15:14:12.187] But I found this inside

[asterisk-users] Beronet card strange log messages

Hi, I have an Asterisk (1.2.9.1) box with a Beronet card and chan-misdn-queue version 0.3.1-rc23. This morning I found these messages inside my asterisk log (never got them before!!): Oct 9 10:08:49 localhost -- MARK -- Oct 9 10:23:39 localhost kernel: mISDN: prim 280 addr 100 not

SV: [asterisk-users] Help debugging strange asterisk behaviour

] Skickat: den 2 augusti 2006 13:25 Till: asterisk-users@lists.digium.com Ämne: SV: [asterisk-users] Help debugging strange asterisk behaviour I'm thinking this could be a queue problem? But I still don't understand why the hell it just flips out after a few hours. Now it all ran for about 12 hours

SV: [asterisk-users] Help debugging strange asterisk behaviour (update)

] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 4 augusti 2006 12:20 Till: asterisk-users@lists.digium.com Ämne: SV: [asterisk-users] Help debugging strange asterisk behaviour Ok. I have an update! When all the problems begin (described below) the 'show queues' command doesn't work

SV: [asterisk-users] Help debugging strange asterisk behaviour

] [mailto:[EMAIL PROTECTED] För Mojo with Horan Company, LLC Skickat: den 1 augusti 2006 23:20 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] Help debugging strange asterisk behaviour Are you using mpg123 for MoH or native? What's in your musiconhold.conf

SV: [asterisk-users] Help debugging strange asterisk behaviour

:52 Till: asterisk-users@lists.digium.com Ämne: SV: [asterisk-users] Help debugging strange asterisk behaviour I think I'm using native since I don't recall installing anything else (except lame codec). How do I check which I am using? I'm unfortunately no asterisk expert that's why I need your

[asterisk-users] Help debugging strange asterisk behaviour

Hi, I'm one of those types who want to know what the heck is wrong when something is wrong. I just installed a new server (see config below) and it all works fine for a few hours. But after 3-5 hours asterisk starts behaving VERY strangely for no apparent reason... 1) MoH stops playing 2) Some

Re: [asterisk-users] Help debugging strange asterisk behaviour

Are you using mpg123 for MoH or native? What's in your musiconhold.conf? [EMAIL PROTECTED] wrote: Hi, I'm one of those types who want to know what the heck is wrong when something is wrong. I just installed a new server (see config below) and it all works fine for a few hours. But after

[Asterisk-Users] Garbled speech - strange problem.

I have two Asterisk boxes connected via IAX2 trunking so that calls made on ServerA to extensions on ServerB connect over a Cable TV connection. As we are close, the latency is only 16ms and this has worked very well for weeks to connect these offices together. Suddenly all of the calls from

[Asterisk-Users] Asterisk Queue strange behaviour

Hi. I have a problem which I assume would be easy to fix, but I can't find anything about it... I wish to have people dialing my phone, and if it is busy, they are put into a queue. And then I am dialed back when the previous call is finished, and connected to the waiting caller. Easy enough?

Re: [Asterisk-Users] Asterisk Queue strange behaviour

: Tuesday, March 15, 2005 6:19 AM Subject: [Asterisk-Users] Asterisk Queue strange behaviour Hi. I have a problem which I assume would be easy to fix, but I can't find anything about it... I wish to have people dialing my phone, and if it is busy, they are put into a queue. And then I am dialed back

Re: [Asterisk-Users] Asterisk Queue strange behaviour

person in the queue. - Original Message - From: Jan Marius Evang [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, March 15, 2005 6:19 AM Subject: [Asterisk-Users] Asterisk Queue strange behaviour Hi. I have a problem which I assume would be easy to fix, but I

Re: [Asterisk-Users] Asterisk Queue strange behaviour

Try adding an agent and logging into that agent (rather than making the SIP phone the member, directly). /edg --On Tuesday, March 15, 2005 1:19 PM +0100 Jan Marius Evang [EMAIL PROTECTED] wrote: Hi. I have a problem which I assume would be easy to fix, but I can't find anything about it... I

Re: [Asterisk-Users] Asterisk Queue strange behaviour

-Users] Asterisk Queue strange behaviour Hi. Is this different depending on wether I am a Member or Agent? I have used Memeber and it does not work... Yours Jan Marius Evang I just did this for a customer. All I did was create a queue just for him, he is the only agent in the queue. * acts just

[Asterisk-Users] Hmm something strange.

I have been tring to send email to the List for the last day or so and have not seen it come back on the list and have not seen any reponse's to my email so I am unsure if it is making it to the list. And I have also been seeing some of the same emails over and over again. The list does not

Re: [Asterisk-Users] Hmm something strange.

I haven't been able to get any list messages to come through to my gmail account today, but my main mailserver hasn't had a problem. Ditto on the duplicates. ~Adam David Uzzell wrote: I have been tring to send email to the List for the last day or so and have not seen it come back on the list

Re: [Asterisk-Users] SNOM 190 - strange voice problems

On Tuesday 26 October 2004 02:36, Ulexus wrote: Jeremy Rusnak wrote: Hi all, ...snip... We're running SIP and version 3.46 of the phone firmware. I don't know about this specific problem, but the latest firmware is 3.55. I'd try it. http://www.snom.com/download/share exactly !

[Asterisk-Users] SNOM 190 - strange voice problems

Hi all, We just installed a new Asterisk system with 12 phones and eight phone lines last week. Things have been going well, except for one strange problem. Most of our phones are Grandstream 102s, but we have four SNOM 190 for exec and support phones (for the PC headset support). These phones

Re: [Asterisk-Users] SNOM 190 - strange voice problems

Jeremy Rusnak wrote: Hi all, ...snip... We're running SIP and version 3.46 of the phone firmware. I don't know about this specific problem, but the latest firmware is 3.55. I'd try it. http://www.snom.com/download/share -- Ulexus ___ Asterisk-Users

[Asterisk-Users] ZAP problem / Strange State

Hello, I am receiving an error in my error logs any time I receive a call on the third line in our hunt group. Sep 20 13:15:03 WARNING[1116939584]: Ring/Off-hook in strange state 6 on channel 3 The weird part is that the calls seem to work fine, just this error message is logged. Currently, I

[Asterisk-Users] ISDN Voicemail: Strange Behaviour

Hi, whenever I include a Ringing Line in some Voicemail Extension I get an error when a call from the outside (via ISDN) comes in, but it works when an internal (SIP-phone) calls the extension. Here is my configuration for testing: extensions.conf [isdnext] ; strep

[Asterisk-Users] H.323 strange error

Hi! I have this test configuration: Cisco7940(SIP)-*-GnuGK(H.323)-ATA186(H.323) When I do call from ATA to 7940, everything is OK (exept volume level, but it is not seriously). But when I try to call from 7940 to ATA, I got a strange error: =*= In CreateRealTimeLogicalChannel for

Re: [Asterisk-Users] THIS IS STRANGE

On Tue, 2003-09-23 at 09:44, Bartosz Jozwiak wrote: Right now it works great! Thanks so much. Could you tell me what is that: 'canreinvite=no' in sip.conf ? When SIP initiates the call, the INVITE message contains the information on where to send the media streams. * uses itself as the

Re: [Asterisk-Users] THIS IS STRANGE

On Mon, 2003-09-22 at 13:51, Bartosz Jozwiak wrote: When I call directly to X-Lite from ATA it doesn't work but when I call to X-lie with queue from ATA is works... It is really strange. Try adding 'canreinvite=no' to the sip definition for both the X-lite phone and the ATA.

Re: [Asterisk-Users] THIS IS STRANGE

Right now it works great! Thanks so much. Could you tell me what is that: 'canreinvite=no' in sip.conf ? -- Bart - Original Message - From: Stephen Varga [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 23, 2003 8:53 AM Subject: Re: [Asterisk-Users] THIS IS STRANGE

[Asterisk-Users] THIS IS STRANGE

Hello everyone, I have posted once a message that I had problem with Asterisk, ATA and X-Lite. The problem was: When I called from ATA to X-Lie it did not want to work. The connectuion apper in astersik but I could not hear anything. Right now I updated asterisk from CVS and I still have

[Asterisk-Users] Almost there.... strange error message... anyone???

Hello Everyone - Well, I think I'm getting closer with the asterisk connection. This si my setup and I keep getting this error below in ,my /var/log/asterisk/messages file. I have opened 5060 port on the firewall box Jun 26 23:02:17 WARNING[4101]: File chan_sip.c, Line 385 (__sip_xmit):