Re: [asterisk-users] Asterisk Cisco calling Name
On Sat, 1 Dec 2007 00:42:43 -0500, John Bittner wrote: Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling are you sure you've got ulaw enabled for that peer in sip.conf ? and the invite trace shows that the cisco is not sending any cname. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Asterisk Cisco calling Name
The fix for this is not to use the normal Cisco IOS. Must use 12.4T version. It is a Cisco bug. John Bittner Simlab.net -Original Message- On Sat, 1 Dec 2007 00:42:43 -0500, John Bittner wrote: Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling are you sure you've got ulaw enabled for that peer in sip.conf ? and the invite trace shows that the cisco is not sending any cname. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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 -- BEGIN-ANTISPAM-VOTING-LINKS -- Teach CanIt if this mail (ID 11581481) is spam: Spam:https://mx1.simlab.net/b.php?i=11581481m=db075974ace6c=s Not spam:https://mx1.simlab.net/b.php?i=11581481m=db075974ace6c=n Forget vote: https://mx1.simlab.net/b.php?i=11581481m=db075974ace6c=f -- END-ANTISPAM-VOTING-LINKS ___ --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] Asterisk Cisco calling Name
On Thu, 2007-12-06 at 10:32 -0500, John Bittner wrote: The fix for this is not to use the normal Cisco IOS. Must use 12.4T version. It is a Cisco bug. I would suggest jumping to greater than 12.4.11T as they introduced all kinds of DTMF fixes there as well.. -Original Message- On Sat, 1 Dec 2007 00:42:43 -0500, John Bittner wrote: Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling are you sure you've got ulaw enabled for that peer in sip.conf ? and the invite trace shows that the cisco is not sending any cname. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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 ___ --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] Asterisk Cisco calling Name
Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling forward unconditional then Asterisk accepts the call but without cname. Below is a trace. Any help is appreciated. Thanks John Bittner Simlab.net voippbx01*CLI -- SIP read from 216.86.35.24:63549: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag=tgrp:PRI-TRUNK-GROUP1;branch=z9hG4bK111A56 From: sip:[EMAIL PROTECTED];tag=4F9EF08-163B To: sip:[EMAIL PROTECTED] Date: Sat, 01 Dec 2007 05:23:25 GMT Call-ID: [EMAIL PROTECTED] Supported: 100rel,timer,replaces Min-SE: 1800 Cisco-Guid: 1613584196-2667844060-2152857615-892193345 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: pending sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1196486605 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Allow-Events: telephone-event MIME-Version: 1.0 Content-Type: multipart/mixed;boundary=uniqueBoundary Content-Length: 680 --uniqueBoundary Content-Type: application/sdp v=0 o=CiscoSystemsSIP-GW-UserAgent 6852 2375 IN IP4 216.86.35.24 s=SIP Call c=IN IP4 216.86.35.24 t=0 0 m=audio 18472 RTP/AVP 0 101 c=IN IP4 216.86.35.24 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, PRN,isdn*,,NI***, USI,rate,c,s,c,1 USI,lay1,ulaw TMR,00 CPN,04,,1,9734333001 CGN,04,,1,y,4,9733901090 CPC,09 FCI,,,y, GCI,602d57449f0411dc8052000f352dca41 UFC,GEN,5,gentf,79 UFC,GEN,5,fachd,9f8b0100 UFC,GEN,5,inpdu,02010106072a8648ce150004 --uniqueBoundary-- --- (21 headers 33 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 216.86.35.24 : 5060 (non-NAT) Found peer '216.86.35.24' Transmitting (no NAT) to 216.86.35.24:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag=tgrp:PRI-TRUNK-GROUP1;branch=z9hG4bK111A56;received=216.86.35.24 From: sip:[EMAIL PROTECTED];tag=4F9EF08-163B To: sip:[EMAIL PROTECTED];tag=as39c359be Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: SimlabVOIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' voippbx01*CLI -- SIP read from 216.86.35.24:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag=tgrp:PRI-TRUNK-GROUP1;branch=z9hG4bK111A56 From: sip:[EMAIL PROTECTED];tag=4F9EF08-163B To: sip:[EMAIL PROTECTED];tag=as39c359be Date: Sat, 01 Dec 2007 05:23:25 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 ___ --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
