Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
--- "Nestor A. Diaz" <[EMAIL PROTECTED]> wrote: > 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
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. > 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 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
--- "Nestor A. Diaz" <[EMAIL PROTECTED]> wrote: > Vieri wrote: > > 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 ? 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Vieri wrote: > 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:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
--- "Nestor A. Diaz" <[EMAIL PROTECTED]> wrote: > Mojo with Horan & Company, LLC wrote: > > Nestor A. Diaz wrote: > > > >> 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 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 know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Mojo with Horan & Company, LLC wrote: > Nestor A. Diaz wrote: > >> 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:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Nestor A. Diaz wrote: > 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? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
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:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
