Thanks for your answer, here is some more debug information, if is a codec interrupt issue, how can i fix it?<br><br>My Sipura uses UID 1234. The huawei softswitch IP address is 10.220.0.2. The Asterisk IP address is 10.223.6.98.<br><br>The Sipura is registered to the Asterisk box and the Asterisk box is registered to the Huawei softswitch. <br><br>Thanks a lot for your help,<br><br>Carlos Andres Medina<br><br><span style="font-weight: bold;">------------------- INCOMING ------------------------------------------</span><br><br> -- Executing Macro("SIP/10.220.0.2-08191e48", "incoming|SIP/1234") in new stack<br> -- Executing Dial("SIP/10.220.0.2-08191e48", "SIP/1234|30") in new stack<br>We're at 10.223.6.98 port 19404<br>Adding codec 0x4 (ulaw) to SDP<br>Adding codec 0x8 (alaw) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br>13 headers, 11 lines<br>Reliably Transmitting (no NAT) to 10.223.6.99:5150:<br>INVITE
sip:1234@10.223.6.99:5150 SIP/2.0<br>Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872;rport<br>From: "Anonymous" <sip:Anonymous@10.223.6.98>;tag=as448023d0<br>To: <sip:1234@10.223.6.99:5150><br>Contact: <sip:Anonymous@10.223.6.98><br>Call-ID: 7addcc5162ce1ee57bc58cba40828bbd@10.223.6.98<br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Date: Thu, 19 Oct 2006 01:56:50 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Content-Type: application/sdp<br>Content-Length: 236<br><br>v=0<br>o=root 1760 1760 IN IP4 10.223.6.98<br>s=session<br>c=IN IP4 10.223.6.98<br>t=0 0<br>m=audio 19404 RTP/AVP 0 8 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br><br>---<br>-- Called 1234<br><br><-- SIP read from 10.223.6.99:5150:<br>SIP/2.0 100 Trying<br>To: <sip:1234@10.223.6.99:5150><br>From: "Anonymous"
<sip:Anonymous@10.223.6.98>;tag=as448023d0<br>Call-ID: 7addcc5162ce1ee57bc58cba40828bbd@10.223.6.98<br>CSeq: 102 INVITE<br>Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872<br>Server: Sipura/SPA2000-2.0.10(e)<br>Content-Length: 0<br><br><br>--- (8 headers 0 lines)---<br><br><-- SIP read from 10.223.6.99:5150:<br><span style="font-weight: bold;">SIP/2.0 180 Ringing</span><br>To: <sip:1234@10.223.6.99:5150>;tag=e2a724add55f408bi0<br>From: "Anonymous" <sip:Anonymous@10.223.6.98>;tag=as448023d0<br>Call-ID: 7addcc5162ce1ee57bc58cba40828bbd@10.223.6.98<br>CSeq: 102 INVITE<br>Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872<br>Server: Sipura/SPA2000-2.0.10(e)<br>Content-Length: 0<br><br><br>--- (8 headers 0 lines)---<br><span style="font-weight: bold;"> -- SIP/1234-08197388 is ringing</span><br>Transmitting (no NAT) to 10.220.0.2:5061:<br><span style="font-weight: bold;">SIP/2.0 180 Ringing</span><br>Via: SIP/2.0/UDP
10.220.0.2:5061;branch=z9hG4bK94161ad88;received=10.220.0.2<br>From: Anonymous<sip:Anonymous@10.220.0.2>;tag=961d1a68<br>To: <sip:4875129@10.223.6.98;user=phone>;tag=as40afbad8<br>Call-ID: 436fedbce988d7eea66f167d06a0558b@10.220.0.2<br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: <sip:4875129@10.223.6.98><br>Content-Length: 0<br><br><-- SIP read from 10.223.6.99:5150:<br><span style="font-weight: bold;">SIP/2.0 200 OK</span><br>To: <sip:1234@10.223.6.99:5150>;tag=e2a724add55f408bi0<br>From: "Anonymous" <sip:Anonymous@10.223.6.98>;tag=as448023d0<br>Call-ID: 7addcc5162ce1ee57bc58cba40828bbd@10.223.6.98<br>CSeq: 102 INVITE<br>Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872<br>Contact: <sip:1234@10.223.6.99:5150><br>Server: Sipura/SPA2000-2.0.10(e)<br>Content-Length: 229<br>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER<br>Supported:
x-sipura<br>Content-Type: application/sdp<br><br>v=0<br>o=- 78549 78549 IN IP4 10.223.6.99<br>s=-<br>c=IN IP4 10.223.6.99<br>t=0 0<br>m=audio 21101 RTP/AVP 8 100 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:100 NSE/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=ptime:30<br>a=sendrecv<br><br>--- (12 headers 12 lines)---<br>Found RTP audio format 8<br>Found RTP audio format 100<br>Found RTP audio format 101<br>Peer audio RTP is at port 10.223.6.99:21101<br>Found description format PCMA<br>Found description format NSE<br>Found description format telephone-event<br>Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)<br>Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>list_route: hop: <sip:1234@10.223.6.99:5150><br>set_destination: Parsing <sip:1234@10.223.6.99:5150> for address/port to send to<br>set_destination: set destination to
10.223.6.99, port 5150<br>Transmitting (no NAT) to 10.223.6.99:5150:<br>ACK sip:1234@10.223.6.99:5150 SIP/2.0<br>Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK403f58ec;rport<br>From: "Anonymous" <sip:Anonymous@10.223.6.98>;tag=as448023d0<br>To: <sip:1234@10.223.6.99:5150>;tag=e2a724add55f408bi0<br>Contact: <sip:Anonymous@10.223.6.98><br>Call-ID: 7addcc5162ce1ee57bc58cba40828bbd@10.223.6.98<br>CSeq: 102 ACK<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Content-Length: 0<br><br><br>---<br><span style="font-weight: bold;"> -- SIP/1234-08197388 answered SIP/10.220.0.2-08191e48</span><br>We're at 10.223.6.98 port 15322<br>Adding codec 0x4 (ulaw) to SDP<br>Adding codec 0x8 (alaw) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br>Reliably Transmitting (no NAT) to 10.220.0.2:5061:<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bK94161ad88;received=10.220.0.2<br>From:
Anonymous<sip:Anonymous@10.220.0.2>;tag=961d1a68<br>To: <sip:4875129@10.223.6.98;user=phone>;tag=as40afbad8<br>Call-ID: 436fedbce988d7eea66f167d06a0558b@10.220.0.2<br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: <sip:4875129@10.223.6.98><br>Content-Type: application/sdp<br>Content-Length: 233<br><br>v=0<br>o=root 1760 1760 IN IP4 10.223.6.98<br>s=session<br>c=IN IP4 10.223.6.98<br>t=0 0<br>m=audio 15322 RTP/AVP 0 8 97<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:97 telephone-event/8000<br>a=fmtp:97 0-16<br>a=silenceSupp:off - - - -<br> -- Attempting native bridge of SIP/10.220.0.2-08191e48 and SIP/1234-08197388<br><br><-- SIP read from 10.220.0.2:5061:<br>ACK sip:4875129@10.223.6.98 SIP/2.0<br>Via: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bKf4540cd33<br>Call-ID: 436fedbce988d7eea66f167d06a0558b@10.220.0.2<br>From:
Anonymous<sip:Anonymous@10.220.0.2>;tag=961d1a68<br>To: <sip:4875129@10.223.6.98;user=phone>;tag=as40afbad8<br>CSeq: 1 ACK<br>Max-Forwards: 70<br>Content-Length: 0<br><br><br>--- (8 headers 0 lines)---<br><br><-- SIP read from 10.220.0.2:5061:<br>BYE sip:4875129@10.223.6.98 SIP/2.0<br>Via: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bK412198644<br>Call-ID: 436fedbce988d7eea66f167d06a0558b@10.220.0.2<br>From: Anonymous<sip:Anonymous@10.220.0.2>;tag=961d1a68<br>To: <sip:4875129@10.223.6.98;user=phone>;tag=as40afbad8<br>CSeq: 2 BYE<br><span style="font-weight: bold;">Reason: Q.850;cause=100;text="Invalid information element contents"</span><br>Max-Forwards: 70<br>Content-Length: 0<br><br>--- (9 headers 0 lines)---<br><br><span style="font-weight: bold;">--------------------- OUTGOING ------------------------------------------------------</span><br><br><-- SIP read from 10.220.0.2:5060:<br><span style="font-weight: bold;">SIP/2.0 503 Service
Unavailable</span><br>Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK32b32640;rport=5060<br>Call-ID: 6365afa34dc3ae2318bac62b13945272@10.223.6.98<br>From: "4875129"<sip:4875129@10.223.6.98>;tag=as1a151f0f<br>To: <sip:6024042@10.220.0.2>;tag=b1d10bb9<br>CSeq: 102 INVITE<br><span style="font-weight: bold;">Reason: Q.850;cause=98;text="Message not compatible with call state or message type non-existent or not implemented"</span><br style="font-weight: bold;">Content-Length: 0<br><br><br>--- (8 headers 0 lines)---<br>-- Got SIP response 503 "Service Unavailable" back from 10.220.0.2<br>Transmitting (no NAT) to 10.220.0.2:5060:<br>ACK sip:6024042@10.220.0.2 SIP/2.0<br>Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK32b32640;rport<br>From: "4875129" <sip:4875129@10.223.6.98>;tag=as1a151f0f<br>To: <sip:6024042@10.220.0.2>;tag=b1d10bb9<br>Contact: <sip:4875129@10.223.6.98><br>Call-ID: 6365afa34dc3ae2318bac62b13945272@10.223.6.98<br>CSeq: 102
ACK<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Content-Length: 0<br><br><br>---<br>-- SIP/epmbogota-08194768 is circuit-busy<br> == Everyone is busy/congested at this time (1:0/1/0)<br> == Auto fallthrough, channel 'SIP/1234-0818f228' status is 'CONGESTION'<br>Transmitting (no NAT) to 10.223.6.99:5150:<br>SIP/2.0 503 Service Unavailable<br>Via: SIP/2.0/UDP 10.223.6.99:5150;branch=z9hG4bK-8808b7d3;received=10.223.6.99<br>From: <sip:1234@10.223.6.98>;tag=1fdd8f37d10c2e33o0<br>To: <sip:6024042@10.223.6.98>;tag=as0d9917db<br>Call-ID: b4719687-fc4f4f23@10.223.6.99<br>CSeq: 101 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: <sip:6024042@10.223.6.98><br>Content-Length: 0<br>X-Asterisk-HangupCause: Circuit/channel congestion<br><br><br><p> 
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