After a Good Call from a PSTN phone if I do a "sip prune realtime peer 9013XX9XX8" (9013XX9XX8 being the phone number of the Agent/Member) then I can call the number again and not get the issue. So this has something to do with the stuff that is put in my peer table after a call.
Any ideas? On Tue, Jun 14, 2011 at 5:18 PM, Duane Larson <duane.lar...@gmail.com>wrote: > 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 > > http://pastebin.com/9RSvQsmN > > And when the caller or agent hangs this call up 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.com>wrote: > >> Ok. Something isn't right. With a user that is local to my SIP user >> database calls the queue phone number everything works without issue. It is >> when a remote user (like someone from the PSTN) calls the queue phone number >> that the caller gets put into the queue and the agent/member doesn't receive >> the call. I have captured debugs from OpenSIPS and Asterisk and I can't >> really see any difference. I also executed the commands you told me where I >> could. Here are the debugs >> >> Good call from local SIP user to Queue >> LocalUser -> OpenSIPSProxy -> Asterisk (then asterisk calls the >> agent/member) -> OpenSIPSProxy -> Agent >> http://pastebin.com/Fa9y3CXQ >> >> >> >> Bad call from PSTN Caller to Queue >> PSTN Gatway -> OpenSIPSB2BUA -> OpenSIPSProxy -> Asterisk (then asterisk >> doesn't call Agent/Member for some reason) >> http://pastebin.com/VBA9nGAs >> >> >> Thanks for looking at this. Currently this happens every time. Any call >> from a local user gets put in queue and agent is called right away, but any >> call from PSTN user gets put in queue and agent isn't called but the agent >> shows as >> >> Asterisk18*CLI> queue show >> irock.com has 1 calls (max unlimited) in 'ringall' strategy (6s holdtime, >> 131s talktime), W:0, C:12, A:15, SL:0.0% within 0s >> Members: >> SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls >> (last was 1991 secs ago) >> Callers: >> 1. SIP/9013XX9XX8-0000002d (wait: 0:02, prio: 0) >> >> When it is a good call and I do "queue show" I see this >> Asterisk18*CLI> queue show >> irock.com has 0 calls (max unlimited) in 'ringall' strategy (5s holdtime, >> 131s talktime), W:0, C:12, A:16, SL:0.0% within 0s >> Members: >> SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls >> (last was 2079 secs ago) >> No Callers >> >> *How come with the Bad Call the Agent/Member shows up in a "queue show" >> as being a Member and a Caller???* >> >> >> >> On Mon, Jun 13, 2011 at 11:58 PM, Satish Barot < >> satish4aster...@gmail.com> wrote: >> >>> >>> I am not sure but seems like Agent channel not being released from >>> Asterisk. >>> >>> Next time when this happens, try 'core show channels' to check whether >>> Agent channel is released or not. >>> >>> [SATISH] >>> >>> >>> On Mon, Jun 13, 2011 at 9:12 PM, Duane Larson <duane.lar...@gmail.com>wrote: >>> >>>> 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 >>>> column of 'qualify' in my database for the sip users. For my config I am >>>> using OpenSIPS as the register and proxy. Asterisk is only used for >>>> voicemail and ACD/Hunt groups. >>>> >>>> >>>> On Mon, Jun 13, 2011 at 12:53 AM, Satish Barot < >>>> satish4aster...@gmail.com> wrote: >>>> >>>>> >>>>> Provide the entry for Agent SIP/9013XX9XX8 along with parameters >>>>> 'callcounter' and 'qualify' from sip.conf. >>>>> >>>>> Also provide CLI outputs of 'core show channels',sip show peers' and >>>>> 'queue show' when... >>>>> >>>>> (1)First caller enters the Queue >>>>> (2)First caller gets connected with Agent >>>>> (3)First caller gets disconnected from Agent >>>>> (4)Second caller enters the Queue >>>>> >>>>> You may have sequences changed for step no 3 and 4 in your scenario. >>>>> >>>>> >>>>> [SATISH] >>>> >>>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> -- >> -- >> *--*--*--*--*--* >> Duane >> *--*--*--*--*--* >> -- >> > > > > -- > -- > *--*--*--*--*--* > Duane > *--*--*--*--*--* > -- > -- -- *--*--*--*--*--* Duane *--*--*--*--*--* --
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users