Re: [asterisk-users] Problem with Cisco Phones
If I remember correctly, 9.x firmware dropped UDP support altogether. On Thu, Jan 22, 2015 at 4:31 AM, Jordan Cook - Gyron Networks < jordan.c...@gyron.net> wrote: > > Apparently this is a known problem past v8 firmware: > > > http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update- > > version-9/ > > I've done some more playing about and what I've noticed is that even when > using TCP SIP on the 8.x Firmware conferencing doesn’t work - making it use > UDP fixes this. > > So has anyone managed to get the 9.x firmware working with UDP? Possibly > worth a try to see if this resolves the issue? > > > This message may be private and confidential. If you have received this > message in error, please notify us and remove it from your system. > > Gyron may monitor email traffic data and the content of email for the > purposes of security and staff training. > > Gyron Internet Ltd is a limited company registered in England and Wales. > Registered number: 4239332. Registered office: 3 Centro, Boundary Way, > Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered > trademark. > > Gyron is a Deloitte Technology Fast 50 ranked company. > -- > _ > -- 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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org -- _ -- 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
Re: [asterisk-users] Problem with Cisco Phones
> Apparently this is a known problem past v8 firmware: > http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update- > version-9/ I've done some more playing about and what I've noticed is that even when using TCP SIP on the 8.x Firmware conferencing doesn’t work - making it use UDP fixes this. So has anyone managed to get the 9.x firmware working with UDP? Possibly worth a try to see if this resolves the issue? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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
Re: [asterisk-users] Problem with Cisco Phones
Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update-version-9/ On Tue, Jan 20, 2015 at 11:16 AM, Jordan Cook - Gyron Networks < jordan.c...@gyron.net> wrote: > > Next step is packet capture to see if there is a clue as to the cause of > the > > failure in the SIP signalling. > > Right, I see the following when running SIP Debug. Looks to me like the > phones are expecting the server to do the conference mixing, which I guess > it would do in CallManager? > > <--- SIP read from TCP:xxx.xxx.xxx.xxx:50604 ---> > REFER sip:xxx.xxx.xxx.xxx SIP/2.0 > Via: SIP/2.0/TCP xxx.xxx.xxx.xxx:50604;branch=z9hG4bK48c7492c > From: "4005" >;tag=203a07fceb4b00eff1377deb-da93e2ee > To: > Call-ID: outofdialog--001e-67a906f5-5333c...@xxx.xxx.xxx.xxx > Max-Forwards: 70 > Date: Tue, 20 Jan 2015 17:10:19 GMT > CSeq: 101 REFER > User-Agent: Cisco-CP7945G/9.4.2 > Contact: > Referred-By: "4005" > Refer-To: cid:9a2a9191@xxx.xxx.xxx.xxx > Content-Length: 963 > Content-Type: application/x-cisco-remotecc-request+xml > Content-Disposition: session;handling=required > Content-Id: <9a2a9...@xxx.xxx.xxx.xxx> > > > > Conference > 203a07fc-eb4b001c-1bf7ad61-614d3...@xxx.xxx.xxx.xxx > 203a07fceb4b00ed3e4e2321-d9cb1581 > as4a087ee2 0 > 0 > 203a07fc-eb4b001d-14750420-d3d10...@xxx.xxx.xxx.xxx > 203a07fceb4b00ee46f74fd6-4ed3acbd > as18747c6d false > > > explicit > 0 > 0 > > <-> > --- (16 headers 3 lines) --- > Sending to xxx.xxx.xxx.xxx:50604 (no NAT) > Call outofdialog--001e-67a906f5-5333c...@xxx.xxx.xxx.xxx got a SIP call > transfer from caller: (REFER)! > > <--- Transmitting (no NAT) to xxx.xxx.xxx.xxx:50604 ---> > SIP/2.0 603 Declined (No dialog) > Via: SIP/2.0/TCP > xxx.xxx.xxx.xxx:50604;branch=z9hG4bK48c7492c;received=xxx.xxx.xxx.xxx > From: "4005" >;tag=203a07fceb4b00eff1377deb-da93e2ee > To: ;tag=as141fffdd > Call-ID: outofdialog--001e-67a906f5-5333c...@xxx.xxx.xxx.xxx > CSeq: 101 REFER > Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Content-Length: 0 > > > <> > > > This message may be private and confidential. If you have received this > message in error, please notify us and remove it from your system. > > Gyron may monitor email traffic data and the content of email for the > purposes of security and staff training. > > Gyron Internet Ltd is a limited company registered in England and Wales. > Registered number: 4239332. Registered office: 3 Centro, Boundary Way, > Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered > trademark. > > Gyron is a Deloitte Technology Fast 50 ranked company. > -- > _ > -- 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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org -- _ -- 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
Re: [asterisk-users] Problem with Cisco Phones
> Next step is packet capture to see if there is a clue as to the cause of the > failure in the SIP signalling. Right, I see the following when running SIP Debug. Looks to me like the phones are expecting the server to do the conference mixing, which I guess it would do in CallManager? <--- SIP read from TCP:xxx.xxx.xxx.xxx:50604 ---> REFER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/TCP xxx.xxx.xxx.xxx:50604;branch=z9hG4bK48c7492c From: "4005" ;tag=203a07fceb4b00eff1377deb-da93e2ee To: Call-ID: outofdialog--001e-67a906f5-5333c...@xxx.xxx.xxx.xxx Max-Forwards: 70 Date: Tue, 20 Jan 2015 17:10:19 GMT CSeq: 101 REFER User-Agent: Cisco-CP7945G/9.4.2 Contact: Referred-By: "4005" Refer-To: cid:9a2a9191@xxx.xxx.xxx.xxx Content-Length: 963 Content-Type: application/x-cisco-remotecc-request+xml Content-Disposition: session;handling=required Content-Id: <9a2a9...@xxx.xxx.xxx.xxx> Conference 203a07fc-eb4b001c-1bf7ad61-614d3...@xxx.xxx.xxx.xxx 203a07fceb4b00ed3e4e2321-d9cb1581 as4a087ee2 0 0 203a07fc-eb4b001d-14750420-d3d10...@xxx.xxx.xxx.xxx 203a07fceb4b00ee46f74fd6-4ed3acbd as18747c6d false explicit 0 0 <-> --- (16 headers 3 lines) --- Sending to xxx.xxx.xxx.xxx:50604 (no NAT) Call outofdialog--001e-67a906f5-5333c...@xxx.xxx.xxx.xxx got a SIP call transfer from caller: (REFER)! <--- Transmitting (no NAT) to xxx.xxx.xxx.xxx:50604 ---> SIP/2.0 603 Declined (No dialog) Via: SIP/2.0/TCP xxx.xxx.xxx.xxx:50604;branch=z9hG4bK48c7492c;received=xxx.xxx.xxx.xxx From: "4005" ;tag=203a07fceb4b00eff1377deb-da93e2ee To: ;tag=as141fffdd Call-ID: outofdialog--001e-67a906f5-5333c...@xxx.xxx.xxx.xxx CSeq: 101 REFER Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <> This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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
Re: [asterisk-users] Problem with Cisco Phones
Next step is packet capture to see if there is a clue as to the cause of the failure in the SIP signalling. On Tue, Jan 20, 2015 at 10:41 AM, Jordan Cook - Gyron Networks < jordan.c...@gyron.net> wrote: > We were using G722 - I thought similarly and tried a call with alaw. Same > problem occurred, any other ideas? > > > I'm willing to bet you are forcing the use of G729. 7940 and 7960 > phones can > > only do a single G729 channel, and if you require G729 for the second > leg of a > > conference, it will fail. > > > > This message may be private and confidential. If you have received this > message in error, please notify us and remove it from your system. > > Gyron may monitor email traffic data and the content of email for the > purposes of security and staff training. > > Gyron Internet Ltd is a limited company registered in England and Wales. > Registered number: 4239332. Registered office: 3 Centro, Boundary Way, > Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered > trademark. > > Gyron is a Deloitte Technology Fast 50 ranked company. > -- > _ > -- 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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org -- _ -- 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
Re: [asterisk-users] Problem with Cisco Phones
We were using G722 - I thought similarly and tried a call with alaw. Same problem occurred, any other ideas? > I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can > only do a single G729 channel, and if you require G729 for the second leg of a > conference, it will fail. This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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
Re: [asterisk-users] Problem with Cisco Phones
I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can only do a single G729 channel, and if you require G729 for the second leg of a conference, it will fail. On Tue, Jan 20, 2015 at 10:03 AM, Jordan Cook - Gyron Networks < jordan.c...@gyron.net> wrote: > Possibly slightly off topic, has anyone ever had Cisco 79xx Series > phones come up with “cannot complete conference” errors when trying to > conference two calls together? > > > This message may be private and confidential. If you have received this > message in error, please notify us and remove it from your system. > > Gyron may monitor email traffic data and the content of email for the > purposes of security and staff training. > > Gyron Internet Ltd is a limited company registered in England and Wales. > Registered number: 4239332. Registered office: 3 Centro, Boundary Way, > Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered > trademark. > > Gyron is a Deloitte Technology Fast 50 ranked company. > > -- > _ > -- 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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org -- _ -- 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
