Hi all!

I've got a problem with Kamailio<->Cisco<->PSTN.

Called from PSTN:

16:20:14.328786 IP (tos 0x80, ttl 255, id 0, offset 0, flags [none], proto UDP (17), length 1166) 172.16.16.3.58446 > 172.16.17.8.sip: SIP, length: 1138 INVITE sip:599674@172.16.17.8:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 Remote-Party-ID: sip:595311@172.16.16.3;party=calling;screen=no;privacy=off From: sip:595311@172.16.16.3;tag=144D20C-8A7 To: sip:599674@172.16.17.8 Date: Thu, 06 Jun 2013 04:25:44 GMT Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4@172.16.16.3 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 4251252826-3449229794-2149122083-881571867 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1370492744 Contact: sip:595311@172.16.16.3:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 279 v=0 o=CiscoSystemsSIP-GW-UserAgent 6723 8551 IN IP4 172.16.16.3 s=SIP Call c=IN IP4 172.16.16.3 t=0 0 m=audio 18550 RTP/AVP 8 18 101 c=IN IP4 172.16.16.3 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 16:20:14.329130 IP (tos 0x10, ttl 64, id 17361, offset 0, flags [none], proto UDP (17), length 327, bad cksum 0 (->bc99)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 299 SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 From: sip:595311@172.16.16.3;tag=144D20C-8A7 To: sip:599674@172.16.17.8 Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4@172.16.16.3 CSeq: 101 INVITE Server: kamailio (4.1.0-dev6 (x86_64/freebsd)) Content-Length: 0 16:20:14.335619 IP (tos 0x10, ttl 64, id 17363, offset 0, flags [none], proto UDP (17), length 359, bad cksum 0 (->bc77)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 331 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 From: sip:595311@172.16.16.3;tag=144D20C-8A7 To: sip:599674@172.16.17.8 Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4@172.16.16.3 CSeq: 101 INVITE Server: kamailio (4.1.0-dev6 (x86_64/freebsd)) Content-Length: 0 16:20:14.362576 IP (tos 0x10, ttl 64, id 17365, offset 0, flags [none], proto UDP (17), length 623, bad cksum 0 (->bb6d)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 595 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 Record-Route: sip:172.16.17.8;lr=on;did=558.b04 From: sip:0074832595311@172.16.16.3;tag=144D20C-8A7 To: sip:0074832599674@172.16.17.8;tag=1054623052 Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4@172.16.16.3 CSeq: 101 INVITE Contact: sip:0074832599674@10.120.0.18:32225;user=phone Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3140 1.0.7.76 Allow-Events: talk, hold Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 16:20:23.621104 IP (tos 0x10, ttl 64, id 17380, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba50)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 Record-Route: sip:172.16.17.8;lr=on;did=558.b04 From: sip:0074832595311@172.16.16.3;tag=144D20C-8A7 To: sip:0074832599674@172.16.17.8;tag=1054623052 Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4@172.16.16.3 CSeq: 101 INVITE Contact: sip:0074832599674@10.120.0.18:32225;user=phone Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3140 1.0.7.76 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 266 v=0 o=0074832599674 8002 8000 IN IP4 10.120.0.18 s=SIP Call c=IN IP4 10.120.0.18 t=0 0 m=audio 39206 RTP/AVP 8 18 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 16:20:24.126589 IP (tos 0x10, ttl 64, id 17383, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba4d)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865 SIP/2.0 200 OK 16:20:25.135006 IP (tos 0x10, ttl 64, id 17389, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba47)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865 SIP/2.0 200 OK 16:20:27.144412 IP (tos 0x10, ttl 64, id 17395, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba41)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865 SIP/2.0 200 OK

And no ACK from Cisco.

Is it Cisco config problem?

P.S. no sip-ua configuration,

dial-peer voice 10002 voip description ** xxx ** preference 1 destination-pattern 5T voice-class codec 1 session protocol sipv2 session target ipv4:xxx session transport udp

I had no idea about no ACK. Maybe OK from Kamailio incorrect?

-- WBR, Victor JID: coyote@bks.tv JID: coyote@bryansktel.ru I use FREE operation system: 3.9.4-calculate GNU/Linux