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:email@example.com:5060 SIP/2.0**Record-Route: **Via: SIP/2.0/UDP 18.104.22.168;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..
Im noticing strange behavior in one of our Asterisk nodes where the ACK is always sent to the proxy, but RR is not enabled for calls. The proxy drops the ACK.Im using the AMI interface to originate a call:Action: login Username: myusername Secret: mypassw..
Please suggest the way to change the time zone in below sip debug logs. INVITE sip:firstname.lastname@example.org:5060 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rportMax-Forwards: 70From: xxxxxxxxxx ;tag=as23a29r59To: Contact: Call-..
I have been sending OPTIONS requests 1) programatically (my own code), 2)manually via SIP VERIFY PEER x and 3)automatcially by setting verify=yes in sip.conf. The trouble is I do not see anything except an ACK 200 come back from endpoints and it d..