If u want some help put your 53xx and sip config files.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jafar mohammed Sent: Sunday, April 03, 2005 9:41 PM To: [email protected] Subject: [Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP hi's i have been trying to configure my AS5300 to work with my asterisk box. i have tried SIP, calls come, answered and AS5300 sends BYE message after not more than 5 secs. I have also tried MGCP, but i believe i am not configuring that right. here is the output of the sip debug. please help me out or lead me to the direction of sorting this problem out. thank you INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 66.178.100.66:5060 From: <sip:66.178.100.66>;tag=8CB7504-1904 To: <sip:[EMAIL PROTECTED]> Date: Mon, 04 Apr 2005 00:16:50 GMT Call-ID: [EMAIL PROTECTED] Supported: timer Min-SE: 600 Cisco-Guid: 2899651584-2748649945-2861211020-3122285050 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: <sip:66.178.100.66>;party=calling;screen=no;privacy=off Timestamp: 1112573810 Contact: <sip:66.178.100.66:5060> Expires: 180 Allow-Events: telephone-event MIME-Version: 1.0 Content-Type: multipart/mixed;boundary=uniqueBoundary Content-Length: 431 --uniqueBoundary Content-Type: application/sdp v=0 o=CiscoSystemsSIP-GW-UserAgent 5042 571 IN IP4 66.178.100.66 s=SIP Call c=IN IP4 66.178.100.66 t=0 0 m=audio 18992 RTP/AVP 3 19 c=IN IP4 66.178.100.66 a=rtpmap:3 GSM/8000 a=rtpmap:19 CN/8000 a=ptime:10 --uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, GCI,acd52c00a3d511d9aa8a9d8cba1a49fa --uniqueBoundary-- -*- - 21 headers, 21 lines * Using latest SIP request as basis request * Sending to 66.178.100.66 : 5060 (NAT) Apr 4 00:16:40 NOTICE[25296]: chan_sip2.c:5872 check_user_full: User name from URI: 66.178.100.66, Digest auth user: (null) == Authentication turned off, no secret for user 66.178.100.66 * No RDNIS header in SIP packet -- - SIPFromURI: <sip:66.178.100.66>;tag=8CB7504-1904 --> Transmitting (no NAT) response to 66.178.100.66:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.178.100.66:5060 From: <sip:66.178.100.66>;tag=8CB7504-1904 To: <sip:[EMAIL PROTECTED]>;tag=as08ade073 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 -*- -- Executing Answer("SIP/66.178.100.66-bf34", "") in new stack * SDP preparation: We're at 62.56.250.198 port 17962 * Answering with preferred capability 0x2 (gsm) * Answering with preferred capability 0x4 (ulaw) * Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1(g723) --> Reliably Transmitting (no NAT) response to 66.178.100.66:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 66.178.100.66:5060 From: <sip:66.178.100.66>;tag=8CB7504-1904 To: <sip:[EMAIL PROTECTED]>;tag=as08ade073 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 265 v=0 o=root 25296 25296 IN IP4 62.56.250.198 s=session c=IN IP4 62.56.250.198 t=0 0 m=audio 17962 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - -*- -- Executing Wait("SIP/66.178.100.66-bf34", "2") in new stack --- Sip read from 66.178.100.66:50341 ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 66.178.100.66:5060 From: <sip:66.178.100.66>;tag=8CB7504-1904 To: <sip:[EMAIL PROTECTED]>;tag=as08ade073 Date: Mon, 04 Apr 2005 00:16:50 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK -*- - 9 headers, 0 lines --- Sip read from 66.178.100.66:53065 BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 66.178.100.66:5060 From: <sip:66.178.100.66>;tag=8CB7504-1904 To: <sip:[EMAIL PROTECTED]>;tag=as08ade073 Date: Mon, 04 Apr 2005 00:16:50 GMT Call-ID: [EMAIL PROTECTED] User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 6 Timestamp: 1112573810 CSeq: 102 BYE Content-Length: 0 -*- - 11 headers, 0 lines * Sending to 66.178.100.66 : 5060 (non-NAT) --> Transmitting (no NAT) response to 66.178.100.66:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 66.178.100.66:5060 From: <sip:66.178.100.66>;tag=8CB7504-1904 To: <sip:[EMAIL PROTECTED]>;tag=as08ade073 Call-ID: [EMAIL PROTECTED] CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 -*- == Spawn extension (AS5300, 9001, 2) exited non-zero on 'SIP/66.178.100.66-bf34' -- Executing Hangup("SIP/66.178.100.66-bf34", "") in new stack == Spawn extension (AS5300, h, 1) exited non-zero on 'SIP/66.178.100.66-bf34' Destroying SIP dialogue '[EMAIL PROTECTED]' __________________________________________________ Do You Yahoo!? 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