I am still working on integrating Nortel with FreeSWITCH I have the calls coming in through FreeSWITCH and parsing/matching XML Here is the latest issue though the call runs through XML, looks like it is working then the calls drops and no audio was ever heard on the Nortel extension calling in I get SIP errors 415 from the Nortel side and a 501 from the FreeSWITCH The call hangs up with an reason 79 https://wiki.freeswitch.org/wiki/Hangup_Causes I checked and the audio is PCMU:8000 What am I missing here? Something to mention, I have been using an Asterisk as a gateway to pass calls from the Nortel to FS There is no issue when I do it this way, but Asterisk was a temporary patch and I need to make this work on FreeSWITCH Thank you in advance for any assistance Joel
A sip trace plus log would be helpful, 'sofia global siptrace on’, place the call and paste bin the output to pastebin.freeswitch.org -- Brian West brian@freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On May 15, 2014, at 12:31 PM, Joel White <joelewhite@gmail.com> wrote:
I am still working on integrating Nortel with FreeSWITCH
I have the calls coming in through FreeSWITCH and parsing/matching XML
Here is the latest issue though
the call runs through XML, looks like it is working
then the calls drops and no audio was ever heard on the Nortel extension calling in
I get SIP errors 415 from the Nortel side and a 501 from the FreeSWITCH
The call hangs up with an reason 79
https://wiki.freeswitch.org/wiki/Hangup_Causes
I checked and the audio is PCMU:8000
What am I missing here?
Something to mention, I have been using an Asterisk as a gateway to pass calls from the Nortel to FS
There is no issue when I do it this way, but Asterisk was a temporary patch and I need to make this work on FreeSWITCH
Thank you in advance for any assistance
Joel _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
http://pastebin.freeswitch.org/pastebin.php?dl=22567 This siptrace is the CS1000 calling directly to FreeSWITCH 52321 is an extension on the Nortel On Thu, May 15, 2014 at 1:54 PM, Brian West <brian@freeswitch.org> wrote:
A sip trace plus log would be helpful, 'sofia global siptrace on’, place the call and paste bin the output to pastebin.freeswitch.org -- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On May 15, 2014, at 12:31 PM, Joel White <joelewhite@gmail.com> wrote:
I am still working on integrating Nortel with FreeSWITCH
I have the calls coming in through FreeSWITCH and parsing/matching XML
Here is the latest issue though
the call runs through XML, looks like it is working
then the calls drops and no audio was ever heard on the Nortel extension calling in
I get SIP errors 415 from the Nortel side and a 501 from the FreeSWITCH
The call hangs up with an reason 79
https://wiki.freeswitch.org/wiki/Hangup_Causes
I checked and the audio is PCMU:8000
What am I missing here?
Something to mention, I have been using an Asterisk as a gateway to pass calls from the Nortel to FS
There is no issue when I do it this way, but Asterisk was a temporary patch and I need to make this work on FreeSWITCH
Thank you in advance for any assistance
Joel _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
http://pastebin.freeswitch.org/22568 This siptrace is from the CS1000 to Asterisk to FreeSWITCH This go through successfully and media was exchanged. This of course being a temporary solution until I can make FreeSWITCH work. On Thu, May 15, 2014 at 4:18 PM, Joel White <joelewhite@gmail.com> wrote:
http://pastebin.freeswitch.org/pastebin.php?dl=22567
This siptrace is the CS1000 calling directly to FreeSWITCH
52321 is an extension on the Nortel
On Thu, May 15, 2014 at 1:54 PM, Brian West <brian@freeswitch.org> wrote:
A sip trace plus log would be helpful, 'sofia global siptrace on’, place the call and paste bin the output to pastebin.freeswitch.org -- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On May 15, 2014, at 12:31 PM, Joel White <joelewhite@gmail.com> wrote:
I am still working on integrating Nortel with FreeSWITCH
I have the calls coming in through FreeSWITCH and parsing/matching XML
Here is the latest issue though
the call runs through XML, looks like it is working
then the calls drops and no audio was ever heard on the Nortel extension calling in
I get SIP errors 415 from the Nortel side and a 501 from the FreeSWITCH
The call hangs up with an reason 79
https://wiki.freeswitch.org/wiki/Hangup_Causes
I checked and the audio is PCMU:8000
What am I missing here?
Something to mention, I have been using an Asterisk as a gateway to pass calls from the Nortel to FS
There is no issue when I do it this way, but Asterisk was a temporary patch and I need to make this work on FreeSWITCH
Thank you in advance for any assistance
Joel _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I don't miss seeing this ridiculous Nortel SIP one bit... So: - The Nortel includes a phone-context attribute in the username portion of the URI. - The Nortel uses a multipart SIP body (of which the SDP is one part). - The Nortel uses a Contact address with a domain in the URI and a "maddr" attribute to signify the actual signaling IP. When forwarding the INVITE to the Polycom (via Kamailio, apparently) more than likely it's blowing up on the multipart SDP. Check this out: http://wiki.freeswitch.org/wiki/Variable_sip_copy_multipart There is no shortage of hoops to jump through with the CS1000. On Thu, May 15, 2014 at 4:21 PM, Joel White <joelewhite@gmail.com> wrote:
http://pastebin.freeswitch.org/22568
This siptrace is from the CS1000 to Asterisk to FreeSWITCH
This go through successfully and media was exchanged. This of course being a temporary solution until I can make FreeSWITCH work.
On Thu, May 15, 2014 at 4:18 PM, Joel White <joelewhite@gmail.com> wrote:
http://pastebin.freeswitch.org/pastebin.php?dl=22567
This siptrace is the CS1000 calling directly to FreeSWITCH
52321 is an extension on the Nortel
On Thu, May 15, 2014 at 1:54 PM, Brian West <brian@freeswitch.org> wrote:
A sip trace plus log would be helpful, 'sofia global siptrace on’, place the call and paste bin the output to pastebin.freeswitch.org -- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On May 15, 2014, at 12:31 PM, Joel White <joelewhite@gmail.com> wrote:
I am still working on integrating Nortel with FreeSWITCH
I have the calls coming in through FreeSWITCH and parsing/matching XML
Here is the latest issue though
the call runs through XML, looks like it is working
then the calls drops and no audio was ever heard on the Nortel extension calling in
I get SIP errors 415 from the Nortel side and a 501 from the FreeSWITCH
The call hangs up with an reason 79
https://wiki.freeswitch.org/wiki/Hangup_Causes
I checked and the audio is PCMU:8000
What am I missing here?
Something to mention, I have been using an Asterisk as a gateway to pass calls from the Nortel to FS
There is no issue when I do it this way, but Asterisk was a temporary patch and I need to make this work on FreeSWITCH
Thank you in advance for any assistance
Joel _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
Thank you Question for you Kristian This Multipart SIP variable, is it set in the dialplan or globally? On Thu, May 15, 2014 at 5:58 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
I don't miss seeing this ridiculous Nortel SIP one bit...
So:
- The Nortel includes a phone-context attribute in the username portion of the URI. - The Nortel uses a multipart SIP body (of which the SDP is one part). - The Nortel uses a Contact address with a domain in the URI and a "maddr" attribute to signify the actual signaling IP.
When forwarding the INVITE to the Polycom (via Kamailio, apparently) more than likely it's blowing up on the multipart SDP. Check this out:
http://wiki.freeswitch.org/wiki/Variable_sip_copy_multipart
There is no shortage of hoops to jump through with the CS1000.
On Thu, May 15, 2014 at 4:21 PM, Joel White <joelewhite@gmail.com> wrote:
http://pastebin.freeswitch.org/22568
This siptrace is from the CS1000 to Asterisk to FreeSWITCH
This go through successfully and media was exchanged. This of course being a temporary solution until I can make FreeSWITCH work.
On Thu, May 15, 2014 at 4:18 PM, Joel White <joelewhite@gmail.com> wrote:
http://pastebin.freeswitch.org/pastebin.php?dl=22567
This siptrace is the CS1000 calling directly to FreeSWITCH
52321 is an extension on the Nortel
On Thu, May 15, 2014 at 1:54 PM, Brian West <brian@freeswitch.org>wrote:
A sip trace plus log would be helpful, 'sofia global siptrace on’, place the call and paste bin the output to pastebin.freeswitch.org -- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On May 15, 2014, at 12:31 PM, Joel White <joelewhite@gmail.com> wrote:
I am still working on integrating Nortel with FreeSWITCH
I have the calls coming in through FreeSWITCH and parsing/matching XML
Here is the latest issue though
the call runs through XML, looks like it is working
then the calls drops and no audio was ever heard on the Nortel extension calling in
I get SIP errors 415 from the Nortel side and a 501 from the FreeSWITCH
The call hangs up with an reason 79
https://wiki.freeswitch.org/wiki/Hangup_Causes
I checked and the audio is PCMU:8000
What am I missing here?
Something to mention, I have been using an Asterisk as a gateway to pass calls from the Nortel to FS
There is no issue when I do it this way, but Asterisk was a temporary patch and I need to make this work on FreeSWITCH
Thank you in advance for any assistance
Joel
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
inserting <action application="set" data="sip_copy_multipart=false"/> in the dialplan allows calls to ring the Polycom phone on Kamailio However no media is exchanged or at least no audio is heard on either end On Fri, May 16, 2014 at 8:53 AM, Joel White <joelewhite@gmail.com> wrote:
Thank you
Question for you Kristian
This Multipart SIP variable, is it set in the dialplan or globally?
On Thu, May 15, 2014 at 5:58 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
I don't miss seeing this ridiculous Nortel SIP one bit...
So:
- The Nortel includes a phone-context attribute in the username portion of the URI. - The Nortel uses a multipart SIP body (of which the SDP is one part). - The Nortel uses a Contact address with a domain in the URI and a "maddr" attribute to signify the actual signaling IP.
When forwarding the INVITE to the Polycom (via Kamailio, apparently) more than likely it's blowing up on the multipart SDP. Check this out:
http://wiki.freeswitch.org/wiki/Variable_sip_copy_multipart
There is no shortage of hoops to jump through with the CS1000.
On Thu, May 15, 2014 at 4:21 PM, Joel White <joelewhite@gmail.com> wrote:
http://pastebin.freeswitch.org/22568
This siptrace is from the CS1000 to Asterisk to FreeSWITCH
This go through successfully and media was exchanged. This of course being a temporary solution until I can make FreeSWITCH work.
On Thu, May 15, 2014 at 4:18 PM, Joel White <joelewhite@gmail.com>wrote:
http://pastebin.freeswitch.org/pastebin.php?dl=22567
This siptrace is the CS1000 calling directly to FreeSWITCH
52321 is an extension on the Nortel
On Thu, May 15, 2014 at 1:54 PM, Brian West <brian@freeswitch.org>wrote:
A sip trace plus log would be helpful, 'sofia global siptrace on’, place the call and paste bin the output to pastebin.freeswitch.org -- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On May 15, 2014, at 12:31 PM, Joel White <joelewhite@gmail.com> wrote:
I am still working on integrating Nortel with FreeSWITCH
I have the calls coming in through FreeSWITCH and parsing/matching XML
Here is the latest issue though
the call runs through XML, looks like it is working
then the calls drops and no audio was ever heard on the Nortel extension calling in
I get SIP errors 415 from the Nortel side and a 501 from the FreeSWITCH
The call hangs up with an reason 79
https://wiki.freeswitch.org/wiki/Hangup_Causes
I checked and the audio is PCMU:8000
What am I missing here?
Something to mention, I have been using an Asterisk as a gateway to pass calls from the Nortel to FS
There is no issue when I do it this way, but Asterisk was a temporary patch and I need to make this work on FreeSWITCH
Thank you in advance for any assistance
Joel
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
http://pastebin.freeswitch.org/22575 Here is the Siptrace On Fri, May 16, 2014 at 9:16 AM, Joel White <joelewhite@gmail.com> wrote:
inserting <action application="set" data="sip_copy_multipart=false"/> in the dialplan allows calls to ring the Polycom phone on Kamailio
However no media is exchanged or at least no audio is heard on either end
On Fri, May 16, 2014 at 8:53 AM, Joel White <joelewhite@gmail.com> wrote:
Thank you
Question for you Kristian
This Multipart SIP variable, is it set in the dialplan or globally?
On Thu, May 15, 2014 at 5:58 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
I don't miss seeing this ridiculous Nortel SIP one bit...
So:
- The Nortel includes a phone-context attribute in the username portion of the URI. - The Nortel uses a multipart SIP body (of which the SDP is one part). - The Nortel uses a Contact address with a domain in the URI and a "maddr" attribute to signify the actual signaling IP.
When forwarding the INVITE to the Polycom (via Kamailio, apparently) more than likely it's blowing up on the multipart SDP. Check this out:
http://wiki.freeswitch.org/wiki/Variable_sip_copy_multipart
There is no shortage of hoops to jump through with the CS1000.
On Thu, May 15, 2014 at 4:21 PM, Joel White <joelewhite@gmail.com>wrote:
http://pastebin.freeswitch.org/22568
This siptrace is from the CS1000 to Asterisk to FreeSWITCH
This go through successfully and media was exchanged. This of course being a temporary solution until I can make FreeSWITCH work.
On Thu, May 15, 2014 at 4:18 PM, Joel White <joelewhite@gmail.com>wrote:
http://pastebin.freeswitch.org/pastebin.php?dl=22567
This siptrace is the CS1000 calling directly to FreeSWITCH
52321 is an extension on the Nortel
On Thu, May 15, 2014 at 1:54 PM, Brian West <brian@freeswitch.org>wrote:
A sip trace plus log would be helpful, 'sofia global siptrace on’, place the call and paste bin the output to pastebin.freeswitch.org -- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On May 15, 2014, at 12:31 PM, Joel White <joelewhite@gmail.com> wrote:
I am still working on integrating Nortel with FreeSWITCH
I have the calls coming in through FreeSWITCH and parsing/matching XML
Here is the latest issue though
the call runs through XML, looks like it is working
then the calls drops and no audio was ever heard on the Nortel extension calling in
I get SIP errors 415 from the Nortel side and a 501 from the FreeSWITCH
The call hangs up with an reason 79
https://wiki.freeswitch.org/wiki/Hangup_Causes
I checked and the audio is PCMU:8000
What am I missing here?
Something to mention, I have been using an Asterisk as a gateway to pass calls from the Nortel to FS
There is no issue when I do it this way, but Asterisk was a temporary patch and I need to make this work on FreeSWITCH
Thank you in advance for any assistance
Joel
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Ok, I do have audio between the Nortel and FreeSWITCH I just kept playing with the config until it worked Also I had the Nortel sending to the wrong SIP profile On Fri, May 16, 2014 at 9:26 AM, Joel White <joelewhite@gmail.com> wrote:
http://pastebin.freeswitch.org/22575
Here is the Siptrace
On Fri, May 16, 2014 at 9:16 AM, Joel White <joelewhite@gmail.com> wrote:
inserting <action application="set" data="sip_copy_multipart=false"/> in the dialplan allows calls to ring the Polycom phone on Kamailio
However no media is exchanged or at least no audio is heard on either end
On Fri, May 16, 2014 at 8:53 AM, Joel White <joelewhite@gmail.com> wrote:
Thank you
Question for you Kristian
This Multipart SIP variable, is it set in the dialplan or globally?
On Thu, May 15, 2014 at 5:58 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
I don't miss seeing this ridiculous Nortel SIP one bit...
So:
- The Nortel includes a phone-context attribute in the username portion of the URI. - The Nortel uses a multipart SIP body (of which the SDP is one part). - The Nortel uses a Contact address with a domain in the URI and a "maddr" attribute to signify the actual signaling IP.
When forwarding the INVITE to the Polycom (via Kamailio, apparently) more than likely it's blowing up on the multipart SDP. Check this out:
http://wiki.freeswitch.org/wiki/Variable_sip_copy_multipart
There is no shortage of hoops to jump through with the CS1000.
On Thu, May 15, 2014 at 4:21 PM, Joel White <joelewhite@gmail.com>wrote:
http://pastebin.freeswitch.org/22568
This siptrace is from the CS1000 to Asterisk to FreeSWITCH
This go through successfully and media was exchanged. This of course being a temporary solution until I can make FreeSWITCH work.
On Thu, May 15, 2014 at 4:18 PM, Joel White <joelewhite@gmail.com>wrote:
http://pastebin.freeswitch.org/pastebin.php?dl=22567
This siptrace is the CS1000 calling directly to FreeSWITCH
52321 is an extension on the Nortel
On Thu, May 15, 2014 at 1:54 PM, Brian West <brian@freeswitch.org>wrote:
> A sip trace plus log would be helpful, 'sofia global siptrace on’, > place the call and paste bin the output to pastebin.freeswitch.org > -- > Brian West > brian@freeswitch.org > > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > > > On May 15, 2014, at 12:31 PM, Joel White <joelewhite@gmail.com> > wrote: > > I am still working on integrating Nortel with FreeSWITCH > > I have the calls coming in through FreeSWITCH and parsing/matching > XML > > Here is the latest issue though > > the call runs through XML, looks like it is working > > then the calls drops and no audio was ever heard on the Nortel > extension calling in > > I get SIP errors 415 from the Nortel side and a 501 from the > FreeSWITCH > > The call hangs up with an reason 79 > > https://wiki.freeswitch.org/wiki/Hangup_Causes > > I checked and the audio is PCMU:8000 > > What am I missing here? > > > Something to mention, I have been using an Asterisk as a gateway to > pass calls from the Nortel to FS > > There is no issue when I do it this way, but Asterisk was a > temporary patch and I need to make this work on FreeSWITCH > > Thank you in advance for any assistance > > Joel > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Glad you got it working! On Fri, May 16, 2014 at 9:55 AM, Joel White <joelewhite@gmail.com> wrote:
Ok, I do have audio between the Nortel and FreeSWITCH
I just kept playing with the config until it worked
Also I had the Nortel sending to the wrong SIP profile
On Fri, May 16, 2014 at 9:26 AM, Joel White <joelewhite@gmail.com> wrote:
http://pastebin.freeswitch.org/22575
Here is the Siptrace
On Fri, May 16, 2014 at 9:16 AM, Joel White <joelewhite@gmail.com> wrote:
inserting <action application="set" data="sip_copy_multipart=false"/> in the dialplan allows calls to ring the Polycom phone on Kamailio
However no media is exchanged or at least no audio is heard on either end
On Fri, May 16, 2014 at 8:53 AM, Joel White <joelewhite@gmail.com>wrote:
Thank you
Question for you Kristian
This Multipart SIP variable, is it set in the dialplan or globally?
On Thu, May 15, 2014 at 5:58 PM, Kristian Kielhofner <kris@kriskinc.com
wrote:
I don't miss seeing this ridiculous Nortel SIP one bit...
So:
- The Nortel includes a phone-context attribute in the username portion of the URI. - The Nortel uses a multipart SIP body (of which the SDP is one part). - The Nortel uses a Contact address with a domain in the URI and a "maddr" attribute to signify the actual signaling IP.
When forwarding the INVITE to the Polycom (via Kamailio, apparently) more than likely it's blowing up on the multipart SDP. Check this out:
http://wiki.freeswitch.org/wiki/Variable_sip_copy_multipart
There is no shortage of hoops to jump through with the CS1000.
On Thu, May 15, 2014 at 4:21 PM, Joel White <joelewhite@gmail.com>wrote:
http://pastebin.freeswitch.org/22568
This siptrace is from the CS1000 to Asterisk to FreeSWITCH
This go through successfully and media was exchanged. This of course being a temporary solution until I can make FreeSWITCH work.
On Thu, May 15, 2014 at 4:18 PM, Joel White <joelewhite@gmail.com>wrote:
> http://pastebin.freeswitch.org/pastebin.php?dl=22567 > > This siptrace is the CS1000 calling directly to FreeSWITCH > > 52321 is an extension on the Nortel > > > On Thu, May 15, 2014 at 1:54 PM, Brian West <brian@freeswitch.org>wrote: > >> A sip trace plus log would be helpful, 'sofia global siptrace on’, >> place the call and paste bin the output to pastebin.freeswitch.org >> -- >> Brian West >> brian@freeswitch.org >> >> >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On May 15, 2014, at 12:31 PM, Joel White <joelewhite@gmail.com> >> wrote: >> >> I am still working on integrating Nortel with FreeSWITCH >> >> I have the calls coming in through FreeSWITCH and parsing/matching >> XML >> >> Here is the latest issue though >> >> the call runs through XML, looks like it is working >> >> then the calls drops and no audio was ever heard on the Nortel >> extension calling in >> >> I get SIP errors 415 from the Nortel side and a 501 from the >> FreeSWITCH >> >> The call hangs up with an reason 79 >> >> https://wiki.freeswitch.org/wiki/Hangup_Causes >> >> I checked and the audio is PCMU:8000 >> >> What am I missing here? >> >> >> Something to mention, I have been using an Asterisk as a gateway to >> pass calls from the Nortel to FS >> >> There is no issue when I do it this way, but Asterisk was a >> temporary patch and I need to make this work on FreeSWITCH >> >> Thank you in advance for any assistance >> >> Joel >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
Dear All, I have tried compiling freeswitch on Windows 7 64bit, Visual C++ 2010 and Visual C++ 2010 express. Both are giving me errors. Eg could not open freeswitchcore.lib. grateful if anyone can help regards raj moloo On Thu, May 15, 2014 at 9:31 PM, Joel White <joelewhite@gmail.com> wrote:
I am still working on integrating Nortel with FreeSWITCH
I have the calls coming in through FreeSWITCH and parsing/matching XML
Here is the latest issue though
the call runs through XML, looks like it is working
then the calls drops and no audio was ever heard on the Nortel extension calling in
I get SIP errors 415 from the Nortel side and a 501 from the FreeSWITCH
The call hangs up with an reason 79
https://wiki.freeswitch.org/wiki/Hangup_Causes
I checked and the audio is PCMU:8000
What am I missing here?
Something to mention, I have been using an Asterisk as a gateway to pass calls from the Nortel to FS
There is no issue when I do it this way, but Asterisk was a temporary patch and I need to make this work on FreeSWITCH
Thank you in advance for any assistance
Joel
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Email Disclaimer: This email and all its contents are subject to the disclaimer at http://www.uom.ac.mu/emaildisclaimer -- <http://www.uom.ac.mu>
participants (4)
-
Brian West
-
Joel White
-
Kristian Kielhofner
-
Raj Moloo