Re: [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive

2013-04-06 Thread Duane Larson
Looks like version 11.3 did not fix my issue. http://pastebin.com/gd291Bqz On Thu, Apr 4, 2013 at 1:23 PM, Duane Larson duane.lar...@gmail.com wrote: Thanks Jim. Searched through the change log for deadlock but nothing really stuck out. I'll upgrade to 11.3 and see if that makes

Re: [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive

2013-04-04 Thread Duane Larson
Thanks Jim. Searched through the change log for deadlock but nothing really stuck out. I'll upgrade to 11.3 and see if that makes a difference. On Thu, Apr 4, 2013 at 10:59 AM, Jim Lucas li...@cmsws.com wrote: On 04/03/2013 08:15 PM, Duane Larson wrote: So it just happened again on both

Re: [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive

2013-04-03 Thread Duane Larson
It just happened again on the 11.0.1 box and I was able to grab a debug. I am hoping someone can tell me if this is a bug or something wrong with my config. gdb asterisk-bin/sbin/asterisk 29048 Go here for the debug output http://pastebin.com/DGXx0BSk On Tue, Apr 2, 2013 at 7:42 PM, Duane

Re: [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive

2013-04-03 Thread Duane Larson
with version 11.2.1 http://pastebin.com/mbjSSAWM This has to be a bug right? I am thinking of opening an issue on the Asterisk JIRA system On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson duane.lar...@gmail.com wrote: It just happened again on the 11.0.1 box and I was able to grab a debug. I am

[asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive

2013-04-02 Thread Duane Larson
I am currently running two different versions of Asterisk 11.0.1 11.2.1 I have noticed the bug occur on both servers. The issue is that when I try to dial a phone number sometimes the call will never go out. I will check the Asterisk server with NGREP and see that the SIP messages are making

Re: [asterisk-users] Queue not sending call to Agent

2011-06-16 Thread Duane Larson
? On Tue, Jun 14, 2011 at 5:18 PM, Duane Larson duane.lar...@gmail.comwrote: One more piece to add. I had mentioned before that I could get a call from a PSTN user to work the first time. So here is all the output of a Good call from a PSTN user after I have performed a RELOAD on asterisks CLI

Re: [asterisk-users] Queue not sending call to Agent

2011-06-14 Thread Duane Larson
or not. [SATISH] On Mon, Jun 13, 2011 at 9:12 PM, Duane Larson duane.lar...@gmail.comwrote: Yesterday I rebooted the server and it seems to be working again. Not sure what the reboot might have changed. Hopefully it doesn't happen again but I can't be sure. To answer your question I have

Re: [asterisk-users] Queue not sending call to Agent

2011-06-14 Thread Duane Larson
all calls from the PSTN afterward get put in the queue automatically and the agent never gets called. On Tue, Jun 14, 2011 at 4:37 PM, Duane Larson duane.lar...@gmail.comwrote: Ok. Something isn't right. With a user that is local to my SIP user database calls the queue phone number everything

Re: [asterisk-users] Queue not sending call to Agent

2011-06-13 Thread Duane Larson
Yesterday I rebooted the server and it seems to be working again. Not sure what the reboot might have changed. Hopefully it doesn't happen again but I can't be sure. To answer your question I have the sip.conf in my mysql database and in MySQL I have callcounter set to yes. I don't have a

[asterisk-users] Queue not sending call to Agent

2011-06-10 Thread duane . larson
Queue not sending call to Agent I am having an issue and i am not sure if it is a bug or a config issue. I was originally running Asterisk 1.8.1.1 when I noticed this issue. I upgraded to 1.8.4.2 to see if that would fix it but it didn't. The issue is that I have a call queue and the

Re: [asterisk-users] Forward voicemail not working

2011-01-10 Thread Duane Larson
Thanks Chad. I will try the patch. On Mon, Jan 10, 2011 at 2:27 PM, Chad Wallace cwall...@lodgingcompany.comwrote: On Sun, 02 Jan 2011 17:44:19 + duane.lar...@gmail.com wrote: I have asterisk 1.8.0 installed and I am not able to forward a voicemail from one users mailbox to another

Re: [asterisk-users] Forward voicemail not working

2011-01-10 Thread Duane Larson
Patch worked like a charm. Thanks Chad. Thought I had done something wrong when installing. Really appreciate it. On Mon, Jan 10, 2011 at 2:27 PM, Duane Larson duane.lar...@gmail.comwrote: Thanks Chad. I will try the patch. On Mon, Jan 10, 2011 at 2:27 PM, Chad Wallace cwall

Re: [asterisk-users] Forward voicemail not working

2011-01-07 Thread Duane Larson
I still can't figure out why this isn't working. I updated to the latest version of Asterisk 1.8.1 with no luck. I am using Realtime for sipusers and vmusers if that makes any difference. I tested this on a new install and saw the following under the folder where I installed Asterisk I had

[asterisk-users] Forward voicemail not working

2011-01-02 Thread duane . larson
I have asterisk 1.8.0 installed and I am not able to forward a voicemail from one users mailbox to another user. I have the user log into their mailbox press 8 to forward a message enter the extension of the user I wish to forward too I don't prepend a audio message and press # to send the

[asterisk-users] Asterisk 1.8 Realtime Queue not working

2010-12-26 Thread duane . larson
I have configured my mysql database by following this link http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue The only difference is that I am using ODBC instead of MySQL with Realtime. Within extensions.conf I have the following for my queue exten = 9**2**1611,1,Answer exten =

Re: [asterisk-users] Wireless Desktop VoIP Phone?

2010-12-17 Thread Duane Larson
Snom Sent from Droid On Dec 17, 2010 12:36 PM, 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

Re: [asterisk-users] Asterisk with MySQL Cluster

2010-12-01 Thread Duane Larson
Awesome. Didn't notice that, but that is my fault for not reading the changelog or the updated sample configs. I will try this out. Thanks all for the comments. On Wed, Dec 1, 2010 at 10:09 AM, Tilghman Lesher tles...@digium.com wrote: On Tuesday 30 November 2010 18:34:17 Duane Larson wrote

[asterisk-users] Asterisk with MySQL Cluster

2010-11-30 Thread Duane Larson
I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant High-Availability. I was wondering if it's possible for Asterisk to also use multiple database servers for Realtime? Currently with Realtime I am only able to point to a single IP address for a database. If that

Re: [asterisk-users] Asterisk with MySQL Cluster

2010-11-30 Thread Duane Larson
automated failovers in MySQL in the 1-2 second range. Singer On Tue, Nov 30, 2010 at 19:51, David Backeberg dbackeb...@gmail.comwrote: On Tue, Nov 30, 2010 at 7:34 PM, Duane Larson duane.lar...@gmail.com wrote: I have MySQL Cluster set up for OpenSIPS which allows for the best

Re: [asterisk-users] Asterisk with MySQL Cluster

2010-11-30 Thread Duane Larson
Thats sounds interesting too. I will look into that also. On Wed, Dec 1, 2010 at 1:30 AM, Stefan Schmidt s...@sil.at wrote: Am 01.12.10 05:10, schrieb Duane Larson: For me OpenSIPS will do most of the work. Asterisk will only handle Hunt Groups/Queues, IVRs, and Voicemail when OpenSIPS

Re: [asterisk-users] ADSL Load Balancing

2010-11-02 Thread Duane Larson
Your router would have to do per-destination when it came to load balancing between the two dsl circuits. That way a single call could only use one dsl path. On Nov 2, 2010 7:36 PM, Dan Journo d...@keshercommunications.com wrote: Hi, I've got a client with two ADSL connections for

[asterisk-users] Queue Group not forwaring calls to agents

2010-11-01 Thread Duane Larson
I am trying to set up Hunt Groups and I am having some issues. Here is what I am trying to do. All my users actually register with OpenSIPS. Asterisk is using Realtime and I have set up a MySQL View Table so that Asterisk see's all the SIP users info that OpenSIPS has. This is what I have