That was it. My sip provider applied the patch to his Asterisk server
that was referenced in the link you was so kind to provide, and again
everything worked as it should.
Thank you very much!
Ivan
Den 1. nov. 2009 kl. 21:20 skrev Anthony Minessale:
Session-Expires: -1;refresher=uas
On Mon, Nov 2, 2009 at 12:58 PM, Ivan C Myrvold i...@myrvold.org wrote:
That was it. My sip provider applied the patch to his Asterisk server that
was referenced in the link you was so kind to provide, and again everything
worked as it should.
Thank you very much!
This is why Tony's
No one have any idea why this is not working? I have combed through
the log, but couldn't find any clue there.
Incoming calls from my sip provider is working perfect, but for
outgoing calls it looks like Freeswitch is not letting the incoming
rtp to the local sip phone.
Ivan
On 30. okt.
Session-Expires: -1;refresher=uas
nta: 200 OK has fatal syntax errors
This is a know-bug in asterisk.
see: https://issues.asterisk.org/view.php?id=15621
On Sun, Nov 1, 2009 at 4:40 AM, Ivan C Myrvold i...@myrvold.org wrote:
No one have any idea why this is not working? I have combed
I have already set debug to 9, on both profiles.
Ivan
Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:
See that 200 OK that keeps coming in over and over and over and over
again? That's because they never received your ACK. If you can turn on
sofia loglevel to 9 and then watch where you send
fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external siptrace on
sofia loglevel all 9
^
Then run your call, then do this:
sofia loglevel all 0
sofia profile external siptrace off
sofia profile internal siptrace off
fsctl
Yes, now I got a more detailed trace. Thank you for helping me with
this.
A new pastebin at http://pastebin.freeswitch.org/10905
Ivan
Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable:
fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external
Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578
Ivan
Den 27. okt. 2009 kl. 18:34 skrev Eliot Gable:
No, the IP address the media originates from does not need to be tied
to the SIP IP address. Can
On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold i...@myrvold.org wrote:
Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578
Ivan
Uh... you wanna try that PB number again?
-MC
Oh, what happened to it?
Anyway, here is a new pb:
http://pastebin.freeswitch.org/10867
Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold i...@myrvold.org
wrote:
Here is a debug log from a call from an internal phone out to an
See that 200 OK that keeps coming in over and over and over and over
again? That's because they never received your ACK. If you can turn on
sofia loglevel to 9 and then watch where you send the ACK, you will
probably have your answer to why the other system did not receive it.
If you're still not
I have used a SIP provider for more than a year. A few days ago, he
said he was moving to a new server, and asked me to reconfigure. I
did, and everything seemed to work fine, until I did an outgoing call
to an external telephone. I found out I had no audio, in neither
direction. Incoming
Make sure you let their media IPs through your firewall. Also, if you
are behind a NAT, check you have things passing to the correct
internal address.
On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold i...@myrvold.org wrote:
I have used a SIP provider for more than a year. A few days ago, he
said
The server is on a public IP, so there is no nat issue here.
I can also see the rtp messages on wireshark starting just after the
183 Session Progress message on the server, but just in one direction,
coming in to the server.
So it looks like Freeswitch is stopping the rtp.
Is this because
No, the IP address the media originates from does not need to be tied
to the SIP IP address. Can you send a Wireshark capture taken on the
FreeSWITCH server of both call legs? Or, if you can, pastebin a debug
log from FreeSWITCH console with sofia loglevel set to 9 and siptrace
on for any Sofia
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