search for: 3aasterisk

...sip:7531 at 192.168.100.197:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: "asterisk" <sip:asterisk at asterisk>;tag=as4ea953db To: <sip:sip:7531 at 192.168.100.197:5060>;tag=2580238520 Contact: <sip:asterisk at 192.168.100.254 <sip%3Aasterisk at 192.168.100.254>> Call-ID: 3026028457 at 192_168_100_197 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: yes Message-Account: sip:asterisk at...

...help appreciated? I attempted a SIP debug and this is a sample out out: <-- SIP read from 192.168.209.1:48099: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.209.253:5060;branch=z9hG4bK661b7c81;rport=5060;received= 192.168.209.253 From: "asterisk" <sip:asterisk at 192.168.209.253<sip%3Aasterisk at 192.168.209.253> >;tag=as7b41af2a To: "Ananth" <sip:Ananth2 at 192.168.102.10 <sip%3AAnanth2 at 192.168.102.10> >;tag=2bb81ff3969 Call-ID: 7758606e479584bf2c20a80d792841f5 at 192.168.209.253 CSeq: 102 NOTIFY Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) -----...

...0.113:15956;rinstance=542e2865b2c6abe1 SIP/2.0 (71) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.20.249:5060;branch=z9hG4bK29f64fe1;rport (65) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 2: From: "asterisk" < sip:asterisk at 192.168.20.249 <sip%3Aasterisk at 192.168.20.249>>;tag=as4bdc3589 (61) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 3: To: <sip:241 at 192.168.20.113:15956;rinstance=542e2865b2c6abe1> (61) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 4: Contact: < sip:asterisk at 192.168.20.249 <sip%3Aasterisk at 192....

Hi, Using FOP, I've met a situation which makes me ask this simple question : Are both A and B commands bellow equivalent ? A. CLI: originate SIP/9122 application dial Local/9123 at local B. AMI/FOP: 192.168.64.5 -> Action: Originate 192.168.64.5 -> Channel: SIP/9122 192.168.64.5 -> Async: True 192.168.64.5 -> Callerid: 9122 Guest2 <9122> 192.168.64.5 -> Exten: 9123

...ip%3A5101234567 at 209.204.152.219> SIP/2.0 (45) [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.253.35:5060;branch=z9hG4bK23a50508;rport (65) [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 2: From: "asterisk" < sip:asterisk at 192.168.253.35 <sip%3Aasterisk at 192.168.253.35>>;tag=as1826f152 (61) [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 3: To: < sip:5101234567 at 209.204.152.219 <sip%3A5101234567 at 209.204.152.219>> (36) [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 4: Contact: < sip:asterisk at 192.168.253.35 <si...

...54.178>> Call-ID: 266322e108872eab12fb307772a4af79 at 74.54.54.178 CSeq: 102 NOTIFY User-Agent: VoIPMS SERAST Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:asterisk at 74.54.54.178 <sip%3Aasterisk at 74.54.54.178> Voice-Message: 0/0 (0/0) SIP/2.0 489 Bad event Via: SIP/2.0/UDP 74.54.54.178:5060 ;branch=z9hG4bK008e70db;received=74.54.54.178;rport=5060 From: "Unknown" <sip:Unknown at 74.54.54.178 <sip%3AUnknown at 74.54.54.178> >;tag=as5c60da37 To: <sip:s at my.ip....

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can