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VoiceStick + Asterisk: Incoming call: Got SIP response 500Wish to discuss a provider that doesn't have a specific forum? Post your questions, concerns, comments here. |
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| Hello all, I finally got VoiceStick working under Asterisk (name your trunk i2telecom.com and nothing else!!!) but it isn't working exactly right. It registers and I see the call hitting the CLI (turned on debug also). It is routing properly. However, it cannot connect to my softphone extension. SJPhone complains about cannot negotiate codec. I am using chandave's instructions. I even tried allowing all but then I get a strange conversation that sounds like crossed lines. Not sure what to post to diagnose; however, I noticed this after the call disconnects: asterisk1*CLI> <-- SIP read from 206.165.50.116:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.0.122:5060;branch=z9hG4bK7b3a9747;rport=50 60 From: <sip:+15045551234@206.165.50.108>;tag=as638b51fb To: <sip:+15045551212@209.247.16.75>;tag=VPSF506035226 29707 Call-ID: HOUMGC0720060710030522042025@209.244.63.57 CSeq: 102 BYE Content-Length: 0 --- (7 headers 0 lines)--- -- Incoming call: Got SIP response 500 "Internal Server Error" back from 206.165.50.116 Any ideas? Thanks in advance!! |
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| Thread | Thread Starter | Forum | Replies | Last Post |
| SIP settings for VOICESTICK | pujitha | Grandstream Support Forum | 0 | May 25th, 2006 08:36 AM |
| SIP response 500 error - is it a NAT problem | supa | BroadVoice Support Forum | 1 | February 7th, 2006 07:55 AM |
| sip 500 internal server error...is this a nat issue | supa | Asterisk Support Forum | 0 | February 6th, 2006 08:03 PM |
| No response by HT-488 to incoming PSTN call to FXO port | parwaaz | Grandstream Support Forum | 0 | January 29th, 2006 07:01 PM |
| Incoming DID problem and my provider's response. | Veazer | Linksys (Sipura) VoIP Support Forum | 0 | January 29th, 2006 06:52 AM |