Tag : sip
Home » Posts tagged sip" (paged 9)
I am new to Asterisk forum :).I have a requirement of terminating3G Mobile originated calls (coming through 3G-MSC)to EPBX Phones via Asterisk PBX.Setup:Mobile—-> Mobile Switching Center ( 3G)—–SIP interface—>Asterisk PBX—>SIP Phone.I wan..
HiIm using asterisk 1.8 but Im sure this applies to other versions.If someone puts a call divert on a handset such as a Snom phone I get this type of SIP message on receipt of an inbound call:Got SIP response 302 Moved Temporarily back from xxx.xxx.xxx.xxx:xxxxxWh..
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It might be the case that you are are trying to use SIP client over 3G and It registers and call can be initiated from the client but it cant receive call; cause asterisk sever marks it as unreachable immediately after registration. Even more, the ab..
Im preparing a setup before installing it within the next few days.In this setup, Im using a SPA112 as an ATA for an analog phone. The target phone is a Gigaset A400 DECT handset.In my lab, Ive got another A400 handset and an old Matracom 46 handset.W..
I am using Asterisk 12 svn, from today, and after a few thousand calls, I run out of ports. This happens eith PJSIOP and regular old SIP. I think it is RTP related. Any idea how can I troblshoot this. It happened teh same with Asterisk 11. On the ot..
allI am using Asterisk 12.4.0 on debian 7.6 x64I experience some troubles with some specific calls, so I want to dig into this as deep as possibleI run these CLI commands: > sip set history on (answer: SIP History Recording Enabled) > sip set debug p..
Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDIuse alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to al..
Hello.I have properly configured my asterisk server with some SIP accounts that work very well. Now when I want to send a chat message from my SIP client(Zoiper), I receive the following error message:*WARNING [2578]: chan_sip.c: 14511 receive_messa..
Thank you for your reply.After setting pjsip set logger on, the following message is displayed.It seems that the 9002(SIP client) refuse INVITE message. Are SIP methods too many?Thanks, MMEEGGAA——–..