I’m headbanging on this from a couple of days, begging here for some help
I’m configuring tls on asterisk for the first time to experiment with an open (public) service idea about having asterisk accepting any sip user (with the sip.conf option ‘autocreatepeer=yes’)
and call each other on the same server and perhaps to other asterisk servers with the same configuration. Something like ‘skype for poors’ for the ‘average joe’.
I’m using asterisk 10.7.0 on a debian squeeze dedicated server (with public ip).
I’ve followed this tutorial: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial and got no errors but when dialing a test context:
exten => _X.,1,Answer exten => _X.,n,playback(tt-weasels)
exten => _X.,n,echo exten => _X.,n,Hangup()
i get no audio.
On the client side, I’ve tried with many softphones (bink, jitsi, microsip, phonerlite) on both windows and linux, on two different computers but same result.
I’ve also enabled srtp, checked the sip debug trace, recompiled libsrtp from sources, tried different combination of parameters in sip.conf, enabled and disabled some port forwardings on the client’s router but same result: all looks ok, but i get no audio.
If not using tls (but the usual udp and rtp), audio works full-duplex
Anyone had a similar problem ?
Any hints ?
Let me know if i can provide more info.
Thanks for supporting, regards and have a nice day, Mike
* You are viewing the archive for August 8th, 2012
Set alwaysauthreject=yes in [general] section of sip.conf. Restarted asterisk
However when I attempt to register I still get:
[2012-08-08 21:11:34] NOTICE chan_sip.c: Registration from
[2012-08-08 21:12:42] NOTICE chan_sip.c: Registration from
Based on the Asterisk security advisory
(http://downloads.asterisk.org/pub/security/AST-2011-011.html) I would have expected 1.4.42 to respond the same in both cases (since the issue was fixed in 18.104.22.168). Am I missing something obvious?
I have a scenario where half a dozen phones at a site appear to be dropping offline for a few seconds every few hours, but the connection between them and the asterisk server remains up.
It’s been suggested to me that the problem might be to do with qualify –
which is enabled in this case. However, I don’t really want to disable it if at all possible – it’s a very good early warning indicator of network problems, and has often proved useful in diagnosing network faults – especially with end users’ *DSL connections not provided by us.
AIUI from the documentation, qualifysmoothing effectively “averages” the last two qualify results. Is there any way to increase this, so a device won’t be considered unavailable until, for example, 3 consecutive qualify packets have been missed?
Thanks in advance.
I’m benchmarking the performance of a Patton Smartnode 411X gateway. My setup is :
GSM phone <--PSTN--> SN411X <--SIP--> Asterisk <--SIP--> SIP phone
My reference setup is:
GSM phone <--PSTN--> analog phone
In the first case, it takes roughly 10s from the moment GSM user hits Send button to the moment Asterisk gets the incoming call (or the SIP phone rings). In the 2nd case with my reference setup, this delay drops to 4s.
Can I (or should I try to) reduce this delay without loosing Caller ID ?
In my country, Caller ID format is ETSI.