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 port RTP/AVP codec 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 port RTP/AVP codec 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
