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nestor at tiendalinux.com
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 Posted: Wed Apr 16, 2008 7:45 am Post subject: [asterisk-users] Two annoying bugs of asterisk ( sip in use |
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Hello Asterisk People,
I have two annoying bugs in asterisk, that i want to know if some of you
have already found a way to fix:
Background: Asterisk 1.4.18.1 Debian package back ported to Debian etch.
1. I use a queue with just on sip device, one call at a time, however
and without reason just after some couple of hours the sip device show
in use and then no calls are transfered from the queue to the sip
device, i do a sip show inuse and this is the result:asterisk -rx "sip
show inuse"
* User name In use Limit
200 0 3
* Peer name In use Limit
200 1/0 3
Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf,
recreate 200 extensions and reload sip.conf
Not so nice thing to do....
2. AgentCallBack
I know i shouldn't have to use this function, since it is deprecated but
lets comment the behavior
Everything works fine, but when there are calls waiting in the queue,
and the agent log in using this function, the agent is able to take the
call , but the system log off immediately after the agent hang up the call.
No solution at the moment, just login in and log in until there are no
waiting calls, for the agent to not be kicked off.
Slds.
--
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-600-5490 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia |
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Guest
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| Posted: Wed Apr 16, 2008 11:26 am Post subject: [asterisk-users] Two annoying bugs of asterisk ( sip in use |
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Nestor A. Diaz wrote:
| Quote: | 1. I use a queue with just on sip device, one call at a time, however
and without reason just after some couple of hours the sip device show
in use and then no calls are transfered from the queue to the sip
device, i do a sip show inuse and this is the result:asterisk -rx "sip
show inuse"
* User name In use Limit
200 0 3
* Peer name In use Limit
200 1/0 3
Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf,
recreate 200 extensions and reload sip.conf
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Does a simple sip reload work, or do you really need to go to all the
trouble of removing the peer definition? |
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nestor at tiendalinux.com
Guest
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| Posted: Wed Apr 16, 2008 12:29 pm Post subject: [asterisk-users] Two annoying bugs of asterisk ( sip in use |
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Mojo with Horan & Company, LLC wrote:
| Quote: | Nestor A. Diaz wrote:
| Quote: | 1. I use a queue with just on sip device, one call at a time, however
and without reason just after some couple of hours the sip device show
in use and then no calls are transfered from the queue to the sip
device, i do a sip show inuse and this is the result:asterisk -rx "sip
show inuse"
* User name In use Limit
200 0 3
* Peer name In use Limit
200 1/0 3
Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf,
recreate 200 extensions and reload sip.conf
|
Does a simple sip reload work, or do you really need to go to all the
trouble of removing the peer definition?
|
sip reload doesn't work, that's what i have to remove the peer
definition, reload, recreate and reload.
slds.
--
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-600-5490 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia |
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rentorbuy at yahoo.com
Guest
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| Posted: Wed Apr 16, 2008 1:28 pm Post subject: [asterisk-users] Two annoying bugs of asterisk ( sip in use |
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--- "Nestor A. Diaz" <nestor at tiendalinux.com> wrote:
| Quote: | Mojo with Horan & Company, LLC wrote:
| Quote: | Nestor A. Diaz wrote:
| Quote: | 1. I use a queue with just on sip device, one
|
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call at a time, however
| Quote: | | Quote: | and without reason just after some couple of
|
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hours the sip device show
| Quote: | | Quote: | in use and then no calls are transfered from the
|
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queue to the sip
| Quote: | | Quote: | device, i do a sip show inuse and this is the
|
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result:asterisk -rx "sip
| Quote: | | Quote: | show inuse"
* User name In use Limit
200 0 3
* Peer name In use Limit
200 1/0 3
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Did you try a "show channels" to see if there were
stale channels for peer 200?
I had the same problem you describe but it was due to
"hung" channels (used * 1.4.18.1 with rtp*timeout and
saw "inuse" peers during the pre-timeout periods even
though the agents weren't on a call).
____________________________________________________________________________________
Be a better friend, newshound, and
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nestor at tiendalinux.com
Guest
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| Posted: Thu Apr 17, 2008 4:55 am Post subject: [asterisk-users] Two annoying bugs of asterisk ( sip in use |
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Vieri wrote:
| Quote: | Did you try a "show channels" to see if there were
stale channels for peer 200?
I had the same problem you describe but it was due to
"hung" channels (used * 1.4.18.1 with rtp*timeout and
saw "inuse" peers during the pre-timeout periods even
though the agents weren't on a call).
|
No, i don't , but how do do you fix this problem ? with rtp timeout ?
Slds.
--
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-600-5490 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia |
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rentorbuy at yahoo.com
Guest
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| Posted: Thu Apr 17, 2008 7:10 am Post subject: [asterisk-users] Two annoying bugs of asterisk ( sip in use |
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--- "Nestor A. Diaz" <nestor at tiendalinux.com> wrote:
| Quote: | Vieri wrote:
| Quote: | Did you try a "show channels" to see if there were
stale channels for peer 200?
I had the same problem you describe but it was due
|
to
| Quote: | "hung" channels (used * 1.4.18.1 with rtp*timeout
|
and
| Quote: | saw "inuse" peers during the pre-timeout periods
|
even
| Quote: | though the agents weren't on a call).
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No, i don't , but how do do you fix this problem ?
with rtp timeout ?
|
rtp*timeout for sip peers is not a fix but a
workaround.
Try to set both values and reload sip.
Then when you witness what you posted try doing a
"core show channels". You can then try to "soft
hangup" a stuck channel or wait for the rtp*timeouts.
____________________________________________________________________________________
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ |
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nestor at tiendalinux.com
Guest
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| Posted: Thu Apr 17, 2008 10:41 am Post subject: [asterisk-users] Two annoying bugs of asterisk ( sip in use |
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ok, thanks, does rtp*timeout work if i have canreinvite=yes ? since rtp
traffic is not passing thought asterisk, or i have to put canreinvite=no ?
slds.
| Quote: | rtp*timeout for sip peers is not a fix but a
workaround.
Try to set both values and reload sip.
Then when you witness what you posted try doing a
"core show channels". You can then try to "soft
hangup" a stuck channel or wait for the rtp*timeouts.
____________________________________________________________________________________
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
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--
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-600-5490 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia |
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rentorbuy at yahoo.com
Guest
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| Posted: Thu Apr 17, 2008 1:54 pm Post subject: [asterisk-users] Two annoying bugs of asterisk ( sip in use |
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--- "Nestor A. Diaz" <nestor at tiendalinux.com> wrote:
| Quote: | ok, thanks, does rtp*timeout work if i have
canreinvite=yes ? since rtp
traffic is not passing thought asterisk, or i have
to put canreinvite=no ?
|
In my setup it doesn't really matter since calls are
coming in through PSTN->IVR->QUEUE->SIP
AGENT->TRANSFERS THROUGH ZAP PRI TO ANOTHER PBX.
____________________________________________________________________________________
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ |
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