Thanks for your answer, here is some more debug information, if is a codec interrupt issue, how can i fix it?My Sipura uses UID 1234. The huawei softswitch IP address is 10.220.0.2. The Asterisk IP address is 10.223.6.98.The Sipura is registered to the Asterisk box and the Asterisk box is registered to the Huawei softswitch. Thanks a lot for your help,Carlos Andres Medina--- INCOMING -- -- Executing Macro("SIP/10.220.0.2-08191e48", "incoming|SIP/1234") in new stack -- Executing Dial("SIP/10.220.0.2-08191e48", "SIP/1234|30") in new stackWe're at 10.223.6.98 port 19404Adding codec 0x4 (ulaw) to SDPAdding codec 0x8 (alaw) to SDPAdding non-codec 0x1 (telephone-event) to SDP13 headers, 11 linesReliably Transmitting (no NAT) to 10.223.6.99:5150:INVITE
sip:[EMAIL PROTECTED]:5150 SIP/2.0Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872;rportFrom: "Anonymous" sip:[EMAIL PROTECTED];tag=as448023d0To: sip:[EMAIL PROTECTED]:5150Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Thu, 19 Oct 2006 01:56:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Type: application/sdpContent-Length: 236v=0o=root 1760 1760 IN IP4 10.223.6.98s=sessionc=IN IP4 10.223.6.98t=0 0m=audio 19404 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -- Called 1234-- SIP read from 10.223.6.99:5150:SIP/2.0 100 TryingTo: sip:[EMAIL PROTECTED]:5150From: "Anonymous"
sip:[EMAIL PROTECTED];tag=as448023d0Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEVia: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872Server: Sipura/SPA2000-2.0.10(e)Content-Length: 0--- (8 headers 0 lines)- SIP read from 10.223.6.99:5150:SIP/2.0 180 RingingTo: sip:[EMAIL PROTECTED]:5150;tag=e2a724add55f408bi0From: "Anonymous" sip:[EMAIL PROTECTED];tag=as448023d0Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEVia: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872Server: Sipura/SPA2000-2.0.10(e)Content-Length: 0--- (8 headers 0 lines)--- -- SIP/1234-08197388 is ringingTransmitting (no NAT) to 10.220.0.2:5061:SIP/2.0 180 RingingVia: SIP/2.0/UDP
10.220.0.2:5061;branch=z9hG4bK94161ad88;received=10.220.0.2From: Anonymoussip:[EMAIL PROTECTED];tag=961d1a68To: sip:[EMAIL PROTECTED];user=phone;tag=as40afbad8Call-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: sip:[EMAIL PROTECTED]Content-Length: 0-- SIP read from 10.223.6.99:5150:SIP/2.0 200 OKTo: sip:[EMAIL PROTECTED]:5150;tag=e2a724add55f408bi0From: "Anonymous" sip:[EMAIL PROTECTED];tag=as448023d0Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEVia: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872Contact: sip:[EMAIL PROTECTED]:5150Server: Sipura/SPA2000-2.0.10(e)Content-Length: 229Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFERSupported:
x-sipuraContent-Type: application/sdpv=0o=- 78549 78549 IN IP4 10.223.6.99s=-c=IN IP4 10.223.6.99t=0 0m=audio 21101 RTP/AVP 8 100 101a=rtpmap:8 PCMA/8000a=rtpmap:100 NSE/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=ptime:30a=sendrecv--- (12 headers 12 lines)---Found RTP audio format 8Found RTP audio format 100Found RTP audio format 101Peer audio RTP is at port 10.223.6.99:21101Found description format PCMAFound description format NSEFound description format telephone-eventCapabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)list_route: hop: sip:[EMAIL PROTECTED]:5150set_destination: Parsing sip:[EMAIL PROTECTED]:5150 for address/port to send toset_destination: set destination to
10.223.6.99, port 5150Transmitting (no NAT) to 10.223.6.99:5150:ACK sip:[EMAIL PROTECTED]:5150 SIP/2.0Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK403f58ec;rportFrom: "Anonymous" sip:[EMAIL PROTECTED];tag=as448023d0To: sip:[EMAIL PROTECTED]:5150;tag=e2a724add55f408bi0Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0--- -- SIP/1234-08197388 answered SIP/10.220.0.2-08191e48We're at 10.223.6.98 port 15322Adding codec 0x4 (ulaw) to SDPAdding codec 0x8 (alaw) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 10.220.0.2:5061:SIP/2.0 200 OKVia: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bK94161ad88;received=10.220.0.2From:
Anonymoussip:[EMAIL PROTECTED];tag=961d1a68To: sip:[EMAIL PROTECTED];user=phone;tag=as40afbad8Call-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: sip:[EMAIL PROTECTED]Content-Type: application/sdpContent-Length: 233v=0o=root 1760 1760 IN IP4 10.223.6.98s=sessionc=IN IP4 10.223.6.98t=0 0m=audio 15322 RTP/AVP 0 8 97a=rtpmap:0