similar to: Asterisk 13.12.2 : strange queue behaviour

Displaying 11 results from an estimated 11 matches similar to: "Asterisk 13.12.2 : strange queue behaviour"

2016 Nov 21
3
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 > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> 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. > >
2016 Sep 02
3
Trouble getting peer variable (sip username) on 302 Moved Temporarily
Hello when setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI : [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 11.22.33.44:40670 [Aug 31 14:59:34] -- Now forwarding Local/myaccount184 at CallFromQueue-000007f4;2 to 'Local/23 at from-internal' (thanks to SIP/myaccount184-00003729)
2016 Nov 10
0
Asterisk 13.12.2 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.12.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.12.2 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in
2015 Mar 26
1
CDR dst value null after attended transfer
I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,
2007 Feb 15
3
Re: Incremental Updates
As an alternative to polling the client, as Ryan describes, you could consider piggy-backing the status updates on the back of other ajax responses. Which way you go depends entirely on the nature of your app, in particular: 1. how frequently it generates ajax traffic anyway 2. how long the server-side process is going to take If the server-side process takes, say, 20 seconds, polling is a
2009 Dec 08
3
botched RAID, now e2fsck or what?
Hi all, Somehow I managed to mess with a RAID array containing an ext3 partition. Parenthesis, if it matters: I disconnected physically a drive while the array was online. Next thing, I lost the right order of the drives in the array. While trying to re-create it, I overwrote the raid superblocks. Luckily, the array was RAID5 degraded, so whenever I re-created it, it didn't go into sync;
2016 Feb 10
2
Unexpected termination of the call when pick up (res_pjsip)
Please help find the cause of strange behavior res_pjsip. Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up! Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d No diagnostic messages why asterisk suddenly decided to hangup i don't found :( There are suggestions or strong belief
2017 Sep 14
3
Realtime pjsip issues
On Thu, Sep 14, 2017, at 05:27 PM, Bryant Zimmerman wrote: > This appears to be some kind of cache issue. > We have been doing caching with earlier versions of asterisk 13 on the > pjsip realtime, but now for some reason > The items only show up the first time we use pjsip list/show and then > they > are wiped. I see a new full cache option and that appears to make a >
2017 Sep 15
3
Realtime pjsip issues
On Fri, Sep 15, 2017, at 10:37 AM, Bryant Zimmerman wrote: > Joshua > > That is the interesting part of it. We took our configs and database > tables from our working 13.12.2 deployments and tried to use them with > our > new 13.17.1 deployments and we are having issues where the tables are not > working. On the new server asterisk keeps saying it can't find the
2016 Nov 22
2
Regression in 13.13.0-RC1
In my setup, which is FreeBSD, using pjsip 2.5.5 as sip backend I am observing a regression when testing the latest Release Candidate. Any calls get refused and the following error shown on console: [Nov 22 10:49:26] WARNING[101105]: res_rtp_asterisk.c:2400 int create_new_socket(const char *, int): Unable to allocate RTP socket: Protocol not supported [Nov 22 10:49:26] WARNING[101105]:
2017 Sep 15
2
Realtime pjsip issues
Joshua We have completed more testing this morning and when we remove the realtime cache options from the sorcery file the endpoints complete registration, but we pjsip show/list does not offer any feed back at all, We also can't send any pjsip send notify commands as they say they don't have an endpoint there. Something has changed in the cache part of the system that is breaking