El Miércoles, 29 de Abril de 2009, Iñaki Baz Castillo escribió:
El Miércoles, 29 de Abril de 2009, troxlinux escribió:
Hi Bogdan , I don't have any alias en mi opensips.cfg , the only thing
that I have is that when they make calls to the pstn they leave to
that ip port
route[4] {
Hi Bogdan , something stranger happens when I put the debug in 6 I
don't see that it shows me the opensips log
tail -f /var/log/openser.log
twoxserver /sbin/opensips[3744]: INFO:core:sig_usr: signal 15 received
twoxserver /sbin/opensips[3733]: INFO:core:sig_usr: signal 15 received
twoxserver
Hi,
Get a ngrep capture of the SIP traffic between * and OSIPS . Typically a
retransmission is triggered by a lack of response from the other party,
but to see what response is lacking, you need to see the ngrep capture
of the SIP traffic.
Regards,
Bogdan
troxlinux wrote:
Hi list , I have
excuseme , I didn't remember that there was a list
2009/4/27 Alex Balashov abalas...@evaristesys.com:
You may wish to consider posting this to the SER-Asterisk-Interwork list.
regardss
--
rickygm
http://gnuforever.homelinux.com
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Users mailing
You may wish to consider posting this to the SER-Asterisk-Interwork list.
troxlinux wrote:
Hi list , I have some days fighting with asterisk and opensips to
solve this problem, when I use asterisk to listen my voicemail and to
call to the pstn, asterisk shows me this error message:
