Re: [asterisk-dev] PRI Cause codes arriving during dial are lost
On Fri, 17 Feb 2006, Stephen Davies wrote: > So far as I can see, detailed PRI cause codes that arrive "during" a > dial attempt are lost. That turned out to be my mistake. My app was using the wrong ast_ call to get HANGUPCAUSE; its not a "real" variable. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Calls not queued
On Fri, 17 Feb 2006, md wrote: > Hello, > i'm log dinamically the agent (SIP/1001) to the queue using AddQueueMember. > That add the member to the queue and allow that to answer queue calls. The > problem is that a second queued call ring in SIP/1001 when this agent is > busy with another queue Call. > The scenario is: > - SIP/1001 is added dinamically to queue Queue1 with AddQueueMember > - SIP/1002 calls to queue Queue1 > - The call rings in SIP/1001 > - SIP/1001 answer that call and begin the conversation between SIP/1002 and > SIP/1001 > - SIP/1003 calls to queue Queue1 > - The call rings to SIP/1001 --> I think that this isn't correct. I think > that that call must remain in the queue Queue1. Hi, If SIP/1001 only wants one call to be offered, it must send back BUSY when the second is offered. So you need to configure I for one call at a time. An alternative is to set a "call-limit" for the peer in your sip.conf Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] SVN trunk - Zaptel - version.h
Yesterday I blew away my branch 1.2 version of Asterisk. (Luckily I moved it to another drive 8-) And then checked out the trunk version. First try - zaptel won't compile: ZAPTEL_VERSION undefined Fine - I'll just grab it from the old directory... Funny - it says it is automatically generated Tried again tonight - still no update on version.h I miss something or is svn messed up? Brett I been wrong before... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: Repost: Re: [asterisk-dev] How does RFC2833 get indicated to the SIP peer
On 2/17/06, Ed Greenberg <[EMAIL PROTECTED]> wrote: > Can somebody who understands chan_sip.c please explain this to me? THanks. > > --On Thursday, February 16, 2006 6:20 AM -0800 Ed Greenberg > <[EMAIL PROTECTED]> wrote: > > > Back in Asterisk 1.0.5, when we sent our SDP packet to the distant end, > > we sent > > m=audio RTP/AVP 101 > > where the 101 which indicated that we wanted to get RFC2833 DTMF from our > > other end. > > > > Now it's missing, and my peer (level3) is sending me inband DTMF. > > > > It's not obvious to me from reading channels/chan_sip.c (in either the > > old 1.0.5 or the current 1.2.4) how this 101 gets on the end of my Media > > Description line or how else the peer is supposed to know that I need > > rfc2833 DTMF. > > > > Can somebody please explain? Do you have dtmfmode=rfc2833 in sip.conf for this peer? If so, let's get a sip debug and open a bug on bugs.digium.com. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: Repost: Re: [asterisk-dev] How does RFC2833 get indicated to the SIP peer
> > > Back in Asterisk 1.0.5, when we sent our SDP packet to the distant end, > > > we sent > > > m=audio RTP/AVP 101 > > > where the 101 which indicated that we wanted to get RFC2833 DTMF from our > > > other end. > > > > > > Now it's missing, and my peer (level3) is sending me inband DTMF. > > > > > > It's not obvious to me from reading channels/chan_sip.c (in either the > > > old 1.0.5 or the current 1.2.4) how this 101 gets on the end of my Media > > > Description line or how else the peer is supposed to know that I need > > > rfc2833 DTMF. > > > > > > Can somebody please explain? > > Do you have dtmfmode=rfc2833 in sip.conf for this peer? If so, let's > get a sip debug and open a bug on bugs.digium.com. Might also do another update as that was removed by Olle about a week ago, and then restored a few hours later. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
