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john at feith.com
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 Posted: Fri Jun 12, 2009 7:29 pm Post subject: [Freeswitch-users] No VoIP audio after upgrading to latest s |
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Upgraded from Apr 3 svn to svn 13769.
Calling from openzap to 9999 (music on hold) works.
Calling from openzap to 9995 (5 sec echo test) works.
Calling from openzap to vmail works.
Calling from Grandstream to 9999 (music on hold) works.
Calling from Grandstream to 9995 (5 sec echo test) doesn't work
... call goes through however silence is heard.
Calling from Grandstream to vmail doesn't work ... call goes
through however vmail disconnects apparently due to receiving silence.
Calling from Grandstream to openzap doesn't work ... call goes
through and the Grandstream can hear what is said on the openzap
side, however openzap hears silence from the Grandstream.
Calling from Grandstream to Grandstream doesn't work ... call goes
through however both sides hear silence.
Suggestions on how to proceed?
-- John
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| Feith Systems | Voice: 1-215-646-8000 | Email: john@feith.com |
| John Wehle | Fax: 1-215-540-5495 | |
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john at feith.com
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| Posted: Fri Jun 12, 2009 7:51 pm Post subject: [Freeswitch-users] No VoIP audio after upgrading to latest s |
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BTW: in all cases show channels says PCMU 8000 is being used
for the read and well as write codec.
-- John
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| Feith Systems | Voice: 1-215-646-8000 | Email: john@feith.com |
| John Wehle | Fax: 1-215-540-5495 | |
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msc at freeswitch.org
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| Posted: Fri Jun 12, 2009 7:57 pm Post subject: [Freeswitch-users] No VoIP audio after upgrading to latest s |
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On Fri, Jun 12, 2009 at 5:28 PM, John Wehle <john@feith.com (john@feith.com)> wrote:
| Quote: | Upgraded from Apr 3 svn to svn 13769.
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That's a pretty decent jump. I think possibly that the configs changed, especially the auto-nat stuff. For kicks, try launching freeswitch with the "-nonat" flag and see if your symptoms persist. It may be that you need to get newer versions of the sip profile config files.
If you didn't make any modifications to internal.xml and external.xml then delete those from conf/sip_profiles and then go to your fs source and do a "make samples" to get fresh copies of those two files.
If you have modified those two files then I recommend looking at the new default config versions of those two files and integrating your changes into the new ones.
Let us know how it goes...
-MC
| Quote: |
Calling from openzap to 9999 (music on hold) works.
Calling from openzap to 9995 (5 sec echo test) works.
Calling from openzap to vmail works.
Calling from Grandstream to 9999 (music on hold) works.
Calling from Grandstream to 9995 (5 sec echo test) doesn't work
... call goes through however silence is heard.
Calling from Grandstream to vmail doesn't work ... call goes
through however vmail disconnects apparently due to receiving silence.
Calling from Grandstream to openzap doesn't work ... call goes
through and the Grandstream can hear what is said on the openzap
side, however openzap hears silence from the Grandstream.
Calling from Grandstream to Grandstream doesn't work ... call goes
through however both sides hear silence.
Suggestions on how to proceed?
-- John
-------------------------------------------------------------------------
| Feith Systems | Voice: 1-215-646-8000 | Email: john@feith.com (john@feith.com) |
| John Wehle | Fax: 1-215-540-5495 | |
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john at feith.com
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| Posted: Fri Jun 12, 2009 8:20 pm Post subject: [Freeswitch-users] No VoIP audio after upgrading to latest s |
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Yet more information ... a packet trace of a openzap to Grandstream
call shows:
Source Destination Packet
FreeSWITCH Grandstream SIP Request: INVITE ...
Grandstream FreeSWITCH SIP Status: 100 Trying
Grandstream FreeSWITCH SIP Status: 180 Ringing
Grandstream FreeSWITCH SIP Status: 200, with session description
FreeSWITCH Grandstream SIP Request: ACK ...
FreeSWITCH Grandstream RTP Payload type=ITU-T G.711 PCMU ...
FreeSWITCH Grandstream RTP Payload type=ITU-T G.711 PCMU ...
...
FreeSWITCH Grandstream RTP Payload type=ITU-T G.711 PCMU ...
FreeSWITCH Grandstream SIP Request: BYE ...
Grandstream FreeSWITCH SIP Status: 200 OK
The interesting thing is I don't see the Grandstream attempt to send
audio. Is there something that FreeSWITCH needs to say to the Grandstream
in order for the phone to send audio?
-- John
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| Feith Systems | Voice: 1-215-646-8000 | Email: john@feith.com |
| John Wehle | Fax: 1-215-540-5495 | |
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john at feith.com
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| Posted: Fri Jun 12, 2009 9:27 pm Post subject: [Freeswitch-users] No VoIP audio after upgrading to latest s |
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| Quote: | I think possibly that the configs changed, specially the auto-nat stuff
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Yep ... a closer look at the packet trace showed FreeSWITCH settings
the Contact as 10.10.10.1 instead of the actual IP address of the machine.
| Quote: | If you have modified those two files then I recommend looking at the new
default config versions of those two files and integrating your changes into
the new ones.
|
Yep ... that's my SOP.
Looking at the internal.xml supplied with the new FS I see:
<param name="ext-rtp-ip" value="auto-nat"/>
<param name="ext-sip-ip" value="auto-nat"/>
Once I commented out those entries everything worked fine.
I'm kind of surprised that this default changed ... the older FS came
with these commented out and worked fine in the simple configuration
where the server and phones are on the same network segment.
In any case my config has been adjusted, things are working, it's
Friday, and I get to go home so life is good. 
-- John
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| Feith Systems | Voice: 1-215-646-8000 | Email: john@feith.com |
| John Wehle | Fax: 1-215-540-5495 | |
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brian at freeswitch.org
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| Posted: Sat Jun 13, 2009 11:18 am Post subject: [Freeswitch-users] No VoIP audio after upgrading to latest s |
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The bigger question is why was it finding that IP if it was wrong.
/b
On Jun 12, 2009, at 10:24 PM, John Wehle wrote:
| Quote: | Yep ... that's my SOP.
Looking at the internal.xml supplied with the new FS I see:
<param name="ext-rtp-ip" value="auto-nat"/>
<param name="ext-sip-ip" value="auto-nat"/>
Once I commented out those entries everything worked fine.
I'm kind of surprised that this default changed ... the older FS came
with these commented out and worked fine in the simple configuration
where the server and phones are on the same network segment.
In any case my config has been adjusted, things are working, it's
Friday, and I get to go home so life is good.
-- John |
Brian West
brian@freeswitch.org (brian@freeswitch.org)
-- Meet us at ClueCon! http://www.cluecon.com |
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