Tag : ack
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need your advice i dont understand why reply ACK goes to diferrent ip (192.168.88.32)SCREEN belowhttp://tinypic.com/view.php?pic=s6m7me&s=9#.VzsVhvl96IkT..
, 2 weks ago I asked questions about PJSIP and T.38 but got no replies. I upgraded Asterisk to git as of yesterday (309dd2a), and Im still having the same issues. In the trace below, Im sending a fax from Hylafax server through iaxmodem on Asterisk..
I am trying to debug a SIP issue, between an Asterisk 1.2.32 system that is behind a network device to which I dont have ready access, which is performing NAT with possibly some kind of SIP ALG, and an Asterisk 11system on a public IP.My question..
*friends help me **cant get incoming calls in asterisk**(when i connect **80081 in softphone —every thing is ok**)****INVITE sip:80081@10.47.10.10:5060 SIP/2.0**Record-Route: **Via: SIP/2.0/UDP 200.152.125.221;branch=z9hG4bKd4fd.b3489837.0**Via: SIP/2.0/..
Is it possible to use the instant messaging feature of Polycom phones in Asterisk? At the moment Im seeing this in the SIP messaging when I try to send one from a Polycom 450.INVITE sip:0100@:5060;user=phone SIP/2.0Via: SIP/2.0/UDP ;branch=z9hG4bK484dcd1fDD872ECEFr..
I am trying to setup a Grandstream GXP2160 IP-phone with secure calling (SRTP).Secure signaling SSIP for registration is working great !I follow this guide : https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+TutorialBut when I try to make a c..
I have installed the latest version 12 that has been released (12.1.0.rc3).I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesnt support it (no telephony event in the rtpmap) the asterisk responds OK in the signall..