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jordan.cook at gyron.net Guest
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Posted: Tue Jan 20, 2015 11:04 am Post subject: [asterisk-users] Problem with Cisco Phones |
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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?
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sgriepentrog at digium... Guest
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Posted: Tue Jan 20, 2015 11:32 am Post subject: [asterisk-users] Problem with Cisco Phones |
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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.cook@gyron.net (jordan.cook@gyron.net)> wrote:
Quote: |
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?
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Gyron is a Deloitte Technology Fast 50 ranked company.
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jordan.cook at gyron.net Guest
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Posted: Tue Jan 20, 2015 11:41 am Post subject: [asterisk-users] Problem with Cisco Phones |
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We were using G722 - I thought similarly and tried a call with alaw. Same problem occurred, any other ideas?
Quote: | 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.
|
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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.
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sgriepentrog at digium... Guest
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Posted: Tue Jan 20, 2015 12:05 pm Post subject: [asterisk-users] Problem with Cisco Phones |
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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.cook@gyron.net (jordan.cook@gyron.net)> wrote:
Quote: | We were using G722 - I thought similarly and tried a call with alaw. Same problem occurred, any other ideas?
Quote: | 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.
|
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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.
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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 |
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jordan.cook at gyron.net Guest
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Posted: Tue Jan 20, 2015 12:16 pm Post subject: [asterisk-users] Problem with Cisco Phones |
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Quote: | Next step is packet capture to see if there is a clue as to the cause of the
failure in the SIP signalling.
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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" <sip:4005@xxx.xxx.xxx.xxx>;tag=203a07fceb4b00eff1377deb-da93e2ee
To: <sip:4004@xxx.xxx.xxx.xxx>
Call-ID: OutOfDialog--001e-67a906f5-5333c2b8@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: <sip:4005@xxx.xxx.xxx.xxx:50604;transport=tcp>
Referred-By: "4005" <sip:4005@xxx.xxx.xxx.xxx>
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: <9a2a9191@xxx.xxx.xxx.xxx>
<?xml version="1.0" encoding="UTF-8"?>
<x-cisco-remotecc-request> <softkeyeventmsg> <softkeyevent>Conference</softkeyevent> <dialogid> <callid>203a07fc-eb4b001c-1bf7ad61-614d38c1@xxx.xxx.xxx.xxx</callid> <localtag>203a07fceb4b00ed3e4e2321-d9cb1581</localtag> <remotetag>as4a087ee2</remotetag> </dialogid> <linenumber>0</linenumber> <participantnum>0</participantnum> <consultdialogid> <callid>203a07fc-eb4b001d-14750420-d3d10a57@xxx.xxx.xxx.xxx</callid> <localtag>203a07fceb4b00ee46f74fd6-4ed3acbd</localtag> <remotetag>as18747c6d</remotetag> </consultdialogid> <state>false</state> <joindialogid> <callid></callid> <localtag></localtag> <remotetag></remotetag> </joindialogid> <eventdata> <invocationtype>explicit</invocationtype> </eventdata> <userdata></userdata> <softkeyid>0</softkeyid> <applicationid>0</applicationid> </softkeyeventmsg>
</x-cisco-remotecc-request>
<------------->
--- (16 headers 3 lines) ---
Sending to xxx.xxx.xxx.xxx:50604 (no NAT)
Call OutOfDialog--001e-67a906f5-5333c2b8@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" <sip:4005@xxx.xxx.xxx.xxx>;tag=203a07fceb4b00eff1377deb-da93e2ee
To: <sip:4004@xxx.xxx.xxx.xxx>;tag=as141fffdd
Call-ID: OutOfDialog--001e-67a906f5-5333c2b8@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.
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_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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sgriepentrog at digium... Guest
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Posted: Tue Jan 20, 2015 12:27 pm Post subject: [asterisk-users] Problem with Cisco Phones |
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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.cook@gyron.net (jordan.cook@gyron.net)> wrote:
Quote: | > Next step is packet capture to see if there is a clue as to the cause of the
Quote: | 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" <sip:4005@xxx.xxx.xxx.xxx>;tag=203a07fceb4b00eff1377deb-da93e2ee
To: <sip:4004@xxx.xxx.xxx.xxx>
Call-ID: OutOfDialog--001e-67a906f5-5333c2b8@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: <sip:4005@xxx.xxx.xxx.xxx:50604;transport=tcp>
Referred-By: "4005" <sip:4005@xxx.xxx.xxx.xxx>
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: <9a2a9191@xxx.xxx.xxx.xxx>
<?xml version="1.0" encoding="UTF-8"?>
<x-cisco-remotecc-request> <softkeyeventmsg> <softkeyevent>Conference</softkeyevent> <dialogid> <callid>203a07fc-eb4b001c-1bf7ad61-614d38c1@xxx.xxx.xxx.xxx</callid> <localtag>203a07fceb4b00ed3e4e2321-d9cb1581</localtag> <remotetag>as4a087ee2</remotetag> </dialogid> <linenumber>0</linenumber> <participantnum>0</participantnum> <consultdialogid> <callid>203a07fc-eb4b001d-14750420-d3d10a57@xxx.xxx.xxx.xxx</callid> <localtag>203a07fceb4b00ee46f74fd6-4ed3acbd</localtag> <remotetag>as18747c6d</remotetag> </consultdialogid> <state>false</state> <joindialogid> <callid></callid> <localtag></localtag> <remotetag></remotetag> </joindialogid> <eventdata> <invocationtype>explicit</invocationtype> </eventdata> <userdata></userdata> <softkeyid>0</softkeyid> <applicationid>0</applicationid> </softkeyeventmsg>
</x-cisco-remotecc-request>
<------------->
--- (16 headers 3 lines) ---
Sending to xxx.xxx.xxx.xxx:50604 (no NAT)
Call OutOfDialog--001e-67a906f5-5333c2b8@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" <sip:4005@xxx.xxx.xxx.xxx>;tag=203a07fceb4b00eff1377deb-da93e2ee
To: <sip:4004@xxx.xxx.xxx.xxx>;tag=as141fffdd
Call-ID: OutOfDialog--001e-67a906f5-5333c2b8@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
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--
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 |
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jordan.cook at gyron.net Guest
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Posted: Thu Jan 22, 2015 5:31 am Post subject: [asterisk-users] Problem with Cisco Phones |
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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:
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sgriepentrog at digium... Guest
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Posted: Thu Jan 22, 2015 10:48 am Post subject: [asterisk-users] Problem with Cisco Phones |
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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.cook@gyron.net (jordan.cook@gyron.net)> wrote:
Quote: | > Apparently this is a known problem past v8 firmware:
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
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--
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 |
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