search for: ext100

...p:442087926805@213.137.73.179>;tag=as34968f1d Call-ID: DF214E6E-FE9511D8-9BD8BD4E-2B0684DF@213.137.81.27 CSeq: 101 ACK Content-Length: 0 8 headers, 0 lines localhost*CLI> Sip read: REGISTER sip:192.168.1.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bKda87dee87d2b42ca From: <sip:ext100@192.168.1.250>;tag=d96a1d9a3a8eb4af To: <sip:ext100@192.168.1.250> Contact: <sip:ext100@192.168.1.60> Call-ID: d1b18f3c3621e97d@192.168.1.60 CSeq: 402 REGISTER Expires: 120 User-Agent: Grandstream BT100 1.0.4.67 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,...

...some copy of extensions and some other file ..pls. call to asterisk server -> user 100 is busy -> recorded msg say "all lines are busy , pls wait" ,"you are second caller in 'queue', pls wait" -> caller is on hold till user 100 busy then ring user 100 (ext100) another call to asterisk server -> user 100 is busy (same as up) -> recorded msg say "all lines are busy , pls wait" , "you are third caller in 'queue', pls wait" -> caller is on hold till this call is on second level than asterisk say "you are second ca...

hello all i,m newbie in asterisk and now want to sip and h323 connection. this is my scenario: phone(ext100)--->freepbx---sip--->system1---H323--->system2--->freepbx--->phone(ext200) when i call 100 from 200, every thing is ok and phone is ringing but when i call 200 from 100, it says "service unavailable". i debug asterisk in my system 2 and see below message: "Dropping...

Hello, I'm using digium E1 cards and serving SIP users at Asterisk. After the following call (see below) CDR shows two records. First looks as outbound call, but the second - as inbound call. Is it a bug or intended behavior? Call flow: SIP (ext: 100) -> ZAP (national number) SIP (ext: 100) transfers to SIP (ext: 200) SIP (ext: 200) -> ZAP (national number). In CDR it looks like