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: coy...@bks.tv
JID: coy...@bryansktel.ru
I use FREE operation system: 3.9.4-calculate GNU/Linux
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