|Summary:||ASTERISK-08771: Call queues do not work when configured via web interface|
|Reporter:||Christopher Marino (marinoc)||Labels:|
|Date Opened:||2007-02-11 01:04:08.000-0600||Date Closed:||2007-05-16 18:56:51|
|Environment:||Attachments:||( 0) sip.patch|
|Description:||Using the latest version of AsteriskNOW I set up a call queue via web interface. Handsets do not answer until the queues.conf file is maually edited and member entries are changed|
From: member = Agent/extn
To: member = SIP/extn
Would it be possible to have the SIP option available in the web interface?
Please excuse my ignorance if there is some other feature that I have not enabled. I am new to this.
|Comments:||By: Serge Vecher (serge-v) 2007-03-15 11:08:30|
By: Phillip (killfill) 2007-04-04 10:31:28
Asterisk-gui seem to have hardcoded the folowing in queues.html, line 257:
uri += build_action('append', p, $('name').value,"member", "Agent/"+$(selectedagent).value);
If we change it to "SIP" guess what happends... :)
I think having an option for directly assign sip/iax/whatever to call queues could be very good. Maybe there could be done something like this:
User Agent IAX SIP
thouse having asterisk-gui write queues.conf like
member = Agent/03
member = SIP/04
just an idea..
In the meaning time i'll just hardcode to SIP.. :)
By: Brandon Kruse (bkruse) 2007-04-10 17:13:26
I have done some investigation into this, and you are right in the fact that it doesnt work the way it is, but you also made a good point by pointing out it will not always be a sIP device.
I will look into this further, by any chance does it work now with trunk killfill?
By: Phillip (killfill) 2007-04-13 22:55:29
"I will look into this further, by any chance does it work now with trunk killfill?"
Well i dont know what you mean, becouse i see not changes in trunk about this..
anyway, in using this (sip.patch)
By: Brandon Kruse (bkruse) 2007-04-14 15:03:59
what version of the gui?
and can you test it in trunk, as a reference point
By: Phillip (killfill) 2007-04-14 16:10:23
Sure, it trunk as of yesterday ;-)
By: Brandon Kruse (bkruse) 2007-04-16 14:37:52
does this fix it?
By: Phillip (killfill) 2007-04-16 18:53:01
"does this fix it?"
Well if you mean 'this' by the patch, yes.
It writes the queues.conf files with SIP/ extensions instead of Agent,/ so queues work withouth requireing users (agents) to login. Their sip phones just need to be connected.
By: Amaia Lesta (axolass) 2007-05-09 10:07:35
I need to use this patch in Asterisk, I have tried to apply it to the chan_sip.c file in the versions 1.4.0 and 1.4.2 of Asterisk but I get some errors. What I am doing wrong?
[root@asterisk ~]# patch /home/admin/chansip/chan_sip.c /home/admin/chansip/chan_sip.c.patch
patching file /home/admin/chansip/chan_sip.c
Hunk #1 succeeded at 229 with fuzz 2 (offset 81 lines).
Hunk #2 FAILED at 410.
Hunk #3 FAILED at 4672.
Hunk #4 FAILED at 12767.
Hunk ASTERISK-1 succeeded at 16016 with fuzz 1 (offset 3232 lines).
3 out of 5 hunks FAILED -- saving rejects to file /home/admin/chansip/chan_sip.c.rej
[root@asterisk ~]# patch /usr/src/asterisk-1.4.2/channels/chan_sip.c /usr/src/asterisk-1.4.2/channels/chan_sip.c.patch
patching file /usr/src/asterisk-1.4.2/channels/chan_sip.c
Reversed (or previously applied) patch detected! Assume -R? [n] y
Hunk #1 succeeded at 224 with fuzz 2 (offset 76 lines).
Hunk #2 FAILED at 404.
Hunk #3 FAILED at 4665.
Hunk #4 FAILED at 12757.
Hunk ASTERISK-1 succeeded at 16219 with fuzz 1 (offset 3441 lines).
3 out of 5 hunks FAILED -- saving rejects to file /usr/src/asterisk-1.4.2/channels/chan_sip.c.rej
By: Brandon Kruse (bkruse) 2007-05-09 10:53:20
That patch is not applied to 1.4.0
I tried with trunk gui and trunk asterisk, WITHOUT the patch and it works fine.
Any new news?
By: Arseny Chernov (ars888) 2007-05-10 03:01:01
We have a set-up with both IAX2/ and SIP/ agents in the queue, and we have to manually edit queues.conf each time. Imagine this ;-)
As far as I understand this is something from next releases, that would allow applications such as queues and etc. operate with superposition of extension, regardless the protocol one's device is using currently. If this is case, what is the most likely solution time for using 'Agent' ideology?
By: Pari Nannapaneni (pari) 2007-05-16 17:39:26
Agentlogin and agentcallbacklogin fatures were added to the gui
recently and we tested the queues with SIP and Analog phones
we can not reproduce the problem and everything worked fine with the