sr-users June 2014

sr-users@lists.kamailio.org

86 participants
145 discussions

MSILO add on deliver event route still no joy
by Roberto Fichera 13 Jun '14

13 Jun '14

Hi All, I'm trying to add an event route to be execute for each successfully delivered message from msilo module, but I'm still not getting running the event route I have defined for it in the script. If someone can have a look at the patch and give me some hint would be really great. Thanks in advance. Roberto Fichera.

2 15

Transport protocol related question - how to bind a dialog to a negotiated transport protocol
by Klaus Feichtinger 12 Jun '14

12 Jun '14

Hello, I wonder if it is allowed using transport protocol UDP for SIP NOTIFY requests (which are generated by Kamailio/presence module), when the SUBSCRIBE dialog was established using TCP as transport protocol. In other words: this is a principal question if it is allowed changing the transport protocol for in-dialog transactions e.g. from TCP to UDP. Initially I rather thought that in-dialog transactions shall use the same transport protocol as the dialog itself (e.g. SIP INFO requests within a standard media session), except the message would be too big for UDP, where a change to TCP is recommended. Can anybody give me a hint, if the current behaviour of Kamailio is correct or not? Or - how can I bind Kamailio to a specific transport protocol (for messages that are created by modules themselves)? Kamailio is sending NOTIFY requests with UDP, even when the subscription was done with TCP (see excerpt below). 09:58:10.360749 IP (tos 0x0, ttl 64, id 35302, offset 0, flags [DF], proto TCP (6), length 444) 10.1.1.14.37883 > 10.1.1.44.5060: P, cksum 0x1cb3 (correct), 1:393(392) ack 1 win 457 <nop,nop,timestamp 624699305 795715664> SUBSCRIBE sip:116006@10.1.1.44;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 10.1.1.14:5070;rport;branch=z9hG4bK1540213071 From: <sip:1@10.1.1.14:5070>;tag=620071678 To: <sip:116006@10.1.1.44;transport=TCP> Call-ID: 449986375 CSeq: 20 SUBSCRIBE Contact: <sip:1@10.1.1.14:37883> Max-Forwards: 70 Expires: 1200 Event: presence Content-Length: 0 09:58:10.361324 IP (tos 0x10, ttl 64, id 65438, offset 0, flags [DF], proto TCP (6), length 431) 10.1.1.44.5060 > 10.1.1.14.37883: P, cksum 0x29fb (correct), 1:380(379) ack 393 win 215 <nop,nop,timestamp 795715792 624699305> SIP/2.0 202 OK Via: SIP/2.0/TCP 10.1.1.14:5070;rport=37883;branch=z9hG4bK1540213071 From: <sip:1@10.1.1.14:5070>;tag=620071678 To: <sip:116006@10.1.1.44;transport=TCP>;tag=4f7a7e54f75c89f5b968c90011d693b5-9eed Call-ID: 449986375 CSeq: 20 SUBSCRIBE Expires: 1200 Contact: <sip:10.1.1.44:5060;transport=tcp> Content-Length: 0 09:58:10.361507 IP (tos 0x10, ttl 64, id 32295, offset 0, flags [none], proto UDP (17), length 484) 10.1.1.44.5060 > 10.1.1.14.37883: SIP, length: 456 NOTIFY sip:1@10.1.1.14:37883 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.44;branch=z9hG4bKc408.509e6347000000000000000000000000.0 To: sip:1@10.1.1.14;tag=620071678 From: sip:116006@10.1.1.44;tag=4f7a7e54f75c89f5b968c90011d693b5-9eed CSeq: 2 NOTIFY Call-ID: 449986375 Content-Length: 0 User-Agent: kamailio (4.0.4 (i386/linux)) Max-Forwards: 70 Event: presence Contact: <sip:10.1.1.44:5060;transport=tcp> Subscription-State: active;expires=1200 Thx Klaus

2 2

Re: [SR-Users] Crash Kamailio 4.1.3
by Igor Potjevlesch 12 Jun '14

12 Jun '14

Hello, Do you any ideas about the reason of this issue? Is it caused by the last call? Do you have enough information with the PAI? Regards, Igor. 2014-05-28 13:47 GMT+02:00 Igor Potjevlesch <igor.potjevlesch(a)gmail.com>: > Hello, > > p *msg->pai : > > $4 = {type = HDR_PAI_T, name = { > s = 0x7fad2cf8af74 "P-Asserted-Identity: > \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: > application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 > 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 > a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 > PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 > telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", > len = 19}, body = { > s = 0x7fad2cf8af89 "\"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: > application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 > 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 > a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 > PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 > telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", > len = 43}, len = 66, > parsed = 0x7fad389a7220, next = 0x7fad2cf8baf0} > > > p *((p_id_body_t*)msg->pai->parsed) : > > $5 = {id = 0x0, num_ids = 0, next = 0x0} > > Regards, > > > > Igor. > > > 2014-05-28 13:29 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > > Hello, >> >> I need now: >> >> p *msg->pai >> p *((p_id_body_t*)msg->pai->parsed) >> >> Cheers, >> Daniel >> >> >> On 28/05/14 12:46, Igor Potjevlesch wrote: >> >> Hello Daniel, >> >> Here is the result of p *msg : >> >> $3 = {id = 128120, pid = 2414, tval = {tv_sec = 1401265832, tv_usec = >> 5951}, fwd_send_flags = {f = 0 '\000', blst_imask = 0 '\000'}, >> rpl_send_flags = {f = 0 '\000', >> blst_imask = 0 '\000'}, first_line = {type = 1, len = 48, u = >> {request = {method = { >> s = 0x7fad2cf8ac30 "INVITE sip:0123456789@domainSIP/2.0\r\nVia: SIP/2.0/UDP >> a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: 69\r\nFrom: >> \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 6}, uri = { >> s = 0x7fad2cf8ac37 "sip:0123456789@domain SIP/2.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 31}, version = { >> s = 0x7fad2cf8ac57 "SIP/2.0\r\nVia: SIP/2.0/UDP >> a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: 69\r\nFrom: >> \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 7}, >> method_value = 1}, reply = {version = { >> s = 0x7fad2cf8ac30 "INVITE sip:0123456789@domainSIP/2.0\r\nVia: SIP/2.0/UDP >> a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: 69\r\nFrom: >> \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 6}, status = { >> s = 0x7fad2cf8ac37 "sip:0123456789@domain SIP/2.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 31}, reason = { >> s = 0x7fad2cf8ac57 "SIP/2.0\r\nVia: SIP/2.0/UDP >> a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: 69\r\nFrom: >> \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 7}, >> statuscode = 1}}}, via1 = 0x7fad2cf8b158, via2 = 0x0, headers = >> 0x7fad2cf8b118, last_header = 0x7fad2cf8bb30, parsed_flag = >> 18446744073709551615, >> h_via1 = 0x7fad2cf8b118, h_via2 = 0x0, callid = 0x7fad2cf8b7a0, to = >> 0x7fad2cf8b548, cseq = 0x7fad2cf8b7e0, from = 0x7fad2cf8b300, contact = >> 0x7fad2cf8b760, >> maxforwards = 0x7fad2cf8b2c0, route = 0x0, record_route = 0x0, >> content_type = 0x7fad2cf8baf0, content_length = 0x7fad2cf8bb30, >> authorization = 0x0, expires = 0x0, >> proxy_auth = 0x7fad2cf8b8d0, supported = 0x7fad2cf8ba70, require = 0x0, >> proxy_require = 0x0, unsupported = 0x0, allow = 0x7fad2cf8ba30, event = >> 0x0, accept = 0x0, >> accept_language = 0x0, organization = 0x0, priority = 0x0, subject = >> 0x0, user_agent = 0x7fad2cf8b850, server = 0x0, content_disposition = 0x0, >> diversion = 0x0, >> rpid = 0x7fad2cf8b890, refer_to = 0x0, session_expires = 0x0, min_se = >> 0x0, sipifmatch = 0x0, subscription_state = 0x0, date = 0x7fad384baa90, >> identity = 0x0, >> identity_info = 0x0, pai = 0x7fad2cf8bab0, ppi = 0x0, path = 0x0, >> privacy = 0x0, body = 0x0, >> eoh = 0x7fad2cf8afea "\r\nv=0\r\no=root 1751857229 1751857230 IN IP4 >> a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 a.b.c.d\r\nt=0 >> 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:3 >> GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> unparsed = 0x7fad2cf8afea "\r\nv=0\r\no=root 1751857229 1751857230 IN >> IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 a.b.c.d\r\nt=0 >> 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:3 >> GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> rcv = {src_ip = {af = 2, len = 4, u = {addrl = {1950469485, >> 140381949486832}, addr32 = {1950469485, 0, 943417072, 32685}, addr16 = >> {52589, >> 29761, 0, 0, 26352, 14395, 32685, 0}, addr = >> "m\315At\000\000\000\000\360f;8\255\177\000"}}, dst_ip = {af = 2, len = 4, >> u = {addrl = {67638457, 0}, >> addr32 = {67638457, 0, 0, 0}, addr16 = {5305, 1032, 0, 0, 0, 0, >> 0, 0}, addr = "\271\024\b\004", '\000' <repeats 11 times>}}, src_port = >> 5060, dst_port = 5060, >> proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = >> 2, sa_data = "\023\304m\315At\000\000\000\000\000\000\000"}, sin = >> {sin_family = 2, >> sin_port = 50195, sin_addr = {s_addr = 1950469485}, sin_zero = >> "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, >> sin6_flowinfo = 1950469485, sin6_addr = {__in6_u = {__u6_addr8 = >> '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, >> __u6_addr32 = {0, 0, 0, >> 0}}}, sin6_scope_id = 0}}, bind_address = 0x7fad383b6588, >> proto = 1 '\001'}, >> buf = 0x7fad2cf8ac30 "INVITE sip:0123456789@domain SIP/2.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 1254, new_uri = { >> s = 0x7fad2ceabdef "sip:0123456789@gw SIP/2.0\r\nRecord-Route: >> <sip:a.b.c.d;lr=on>\r\nVia: SIP/2.0/UDP >> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5060\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 INVITE\ >> sendrecv\r\n", len = 1254, new_uri = { >> s = 0x7fad2ceabdef "sip:0123456789@gw SIP/2.0\r\nRecord-Route: >> <sip:a.b.c.d;lr=on>\r\nVia: SIP/2.0/UDP >> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5060\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 INVITE\---Type >> <return> to continue, or q <return> to quit--- >> r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nDate: Wed, 28 >> May 2014 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY, INFO\r\nSupported: replaces, >> timer\r\nP-Asserted-Identity: \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 19}, dst_uri = {s = 0x0, len = 0}, parsed_uri_ok = 0, parsed_uri = >> {user = { >> s = 0x7fad2ceabdf3 "0123456789@gw SIP/2.0\r\nRecord-Route: >> <sip:a.b.c.d;lr=on>\r\nVia: SIP/2.0/UDP >> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5060\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nDate: Wed, 28 >> May 2014 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY, INFO\r\nSupported: replaces, >> timer\r\nP-Asserted-Identity: \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 10}, passwd = {s = 0x0, len = 0}, host = { >> s = 0x7fad2ceabdfe "gw SIP/2.0\r\nRecord-Route: <sip:a.b.c.d;lr=on>\r\nVia: >> SIP/2.0/UDP >> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5060\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nDate: Wed, 28 >> May 2014 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY, INFO\r\nSupported: replaces, >> timer\r\nP-Asserted-Identity: \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 4}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, >> sip_params = {s = 0x0, len = 0}, headers = { >> s = 0x0, len = 0}, port_no = 0, proto = 0, type = SIP_URI_T, flags >> = 0, transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = >> {s = 0x0, >> len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, >> lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, >> transport_val = { >> s = 0x0, len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = >> {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, >> len = 0}, lr_val = { >> s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, >> len = 0}}, parsed_orig_ruri_ok = 0, parsed_orig_ruri = {user = {s = 0x0, >> len = 0}, passwd = { >> s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len >> = 0}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers >> = {s = 0x0, >> len = 0}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = 0, >> transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = {s = >> 0x0, len = 0}, >> maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = >> 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, >> transport_val = {s = 0x0, >> len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, >> len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, >> lr_val = {s = 0x0, >> len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = >> 0}}, add_rm = 0x7fad2ce523c8, body_lumps = 0x0, reply_lump = 0x0, >> add_to_branch_s = '\000' <repeats 57 times>, add_to_branch_len = 0, >> hash_index = 26854, msg_flags = 145, flags = 2, set_global_address = {s = >> 0x0, len = 0}, >> set_global_port = {s = 0x0, len = 0}, force_send_socket = 0x0, path_vec >> = {s = 0x0, len = 0}, instance = {s = 0x0, len = 0}, reg_id = 0, ruid = {s >> = 0x0, len = 0}, >> location_ua = {s = 0x0, len = 0}, ldv = {flow = {decoded = 0, rcv = >> {src_ip = {af = 0, len = 0, u = {addrl = {0, 0}, addr32 = {0, 0, 0, 0}, >> addr16 = {0, 0, 0, 0, 0, >> 0, 0, 0}, addr = '\000' <repeats 15 times>}}, dst_ip = {af >> = 0, len = 0, u = {addrl = {0, 0}, addr32 = {0, 0, 0, 0}, addr16 = {0, 0, >> 0, 0, 0, 0, 0, 0}, >> addr = '\000' <repeats 15 times>}}, src_port = 0, dst_port = >> 0, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 0, >> sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, >> sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = >> "\000\000\000\000\000\000\000"}, sin6 = { >> sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr >> = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, >> 0, 0, 0, 0, 0, 0}, >> __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, >> bind_address = 0x0, proto = 0 '\000'}}}} >> >> Regards, >> >> >> >> Igor. >> >> >> >> 2014-05-28 12:01 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >> >>> Hello, >>> >>> can you give the output in gdb for: >>> >>> set print elements 2000 >>> p *msg >>> >>> Cheers, >>> Daniel >>> >>> >>> On 28/05/14 11:48, Igor Potjevlesch wrote: >>> >>> hello, >>> >>> Kamailio 4.1.3 crashed today, with an segmentation fault error : >>> kernel: kamailio[2406]: segfault at 18 ip 0000000000495792 sp >>> 00007fff3d9d2d80 error 4 in kamailio[400000+275000] >>> >>> Here is the result of BT : >>> #0 0x0000000000495792 in pv_get_strval (msg=0x7fad2cf8a530, >>> param=0x7fad3858c288, res=0x7fff3d9d2eb0, sval=0x18) at pvapi.c:521 >>> #1 0x00007fad365b0da8 in pv_get_pai (msg=0x7fad2cf8a530, >>> param=0x7fad3858c288, res=0x7fff3d9d2eb0) at pv_core.c:1021 >>> #2 0x0000000000499518 in pv_get_spec_value (msg=0x7fad2cf8a530, >>> sp=0x7fad3858c270, value=0x7fff3d9d2eb0) at pvapi.c:1266 >>> #3 0x00007fad34c6d7d9 in extra2strar (extra=0x7fad3858c260, >>> rq=0x7fad2cf8a530, val_arr=0x7fad34e7ae30, >>> int_arr=0x7fad34e7b37c,type_arr=0x7fad34e7b4e7 "\002\002\002\002") at >>> acc_extra.c:261 >>> #4 0x00007fad34c65393 in acc_db_request (rq=0x7fad2cf8a530) at acc.c:474 >>> #5 0x00007fad34c6fb43 in acc_onreply (t=0x7fad2ce0b138, >>> req=0x7fad2cf8a530, reply=0x7fad3859faa8, code=200) at acc_logic.c:471 >>> #6 0x00007fad34c7008d in tmcb_func (t=0x7fad2ce0b138, type=512, >>> ps=0x7fff3d9d3120) at acc_logic.c:559 >>> #7 0x00007fad36e6a478 in run_trans_callbacks_internal >>> (cb_lst=0x7fad2ce0b1a8, type=512, trans=0x7fad2ce0b138, >>> params=0x7fff3d9d3120) at t_hooks.c:290 >>> #8 0x00007fad36e6a68a in run_trans_callbacks_with_buf (type=512, >>> rbuf=0x7fad2ce0b1f8, req=0x7fad2cf8a530, repl=0x7fad3859faa8, flags=183) at >>> t_hooks.c:336 >>> #9 0x00007fad36e9cc06 in relay_reply (t=0x7fad2ce0b138, >>> p_msg=0x7fad3859faa8, branch=0, msg_status=183, cancel_data=0x7fff3d9d3480, >>> do_put_on_wait=1) at t_reply.c:2001 >>> #10 0x00007fad36e9f0b7 in reply_received (p_msg=0x7fad3859faa8) at >>> t_reply.c:2499 >>> #11 0x000000000045d837 in do_forward_reply (msg=0x7fad3859faa8, mode=0) >>> at forward.c:777 >>> #12 0x000000000045e0f8 in forward_reply (msg=0x7fad3859faa8) at >>> forward.c:860 >>> #13 0x00000000004a5887 in receive_msg ( >>> buf=0x924600 "SIP/2.0 183 Session Progress\r\nVia: SIP/2.0/UDP >>> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0;received=a.b.c.d\r\nVia: >>> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5"..., len=872, >>> rcv_info=0x7fff3d9d3800) at receive.c:273 >>> #14 0x000000000053c838 in udp_rcv_loop () at udp_server.c:536 >>> #14 0x000000000053c838 in udp_rcv_loop () at udp_server.c:536 >>> #15 0x000000000046d42b in main_loop () at main.c:1617 >>> #16 0x00000000004704d3 in main (argc=7, argv=0x7fff3d9d3b38) at >>> main.c:2533 >>> >>> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> >> >

2 19

Kamailio v4.1.4 Released
by Daniel-Constantin Mierla 12 Jun '14

12 Jun '14

Hello, Kamailio SIP Server v4.1.4 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.3. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.4. For more details about version 4.1.4 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/06/kamailio-v4-1-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Problem with MSRP AUTH
by Muhammad Shahzad 12 Jun '14

12 Jun '14

Hi, I am trying to authentication MSRP connection using the example code of msrp event route in module documentation here, http://kamailio.org/docs/modules/4.1.x/modules/msrp.html#idp119248 -- ... } else if ($msrp(method)=="AUTH") { ... if (!pv_www_authenticate("WEBRTC_SIP_REALM", "$var(passwd)", "0", "$msrp(method)")) { if (auth_get_www_authenticate("WEBRTC_SIP_REALM", "0", "$var(wauth)")) { xlog("L_INFO", "Generated www authenticate header for MSRP is [$var(wauth)] \n"); msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Internal Server Error"); }; exit; }; ... -- However i see in logs following error message and authentication fails, -- ERROR: auth [auth_mod.c:690]: pv_www_authenticate2(): failed to get method value -- Which is indicates that value of $msrp(method) is null. However, as you can see in example code in URL provided above we have an IF condition which explicitly checks $msrp(method) == "AUTH". For the sake of testing i even replaced the variable with actual string value, and still get the same error. So i am guessing it is a bug. What do you guys suggest? Kamailio: v4.1.3 (i386/linux) 236326 MSRP Lib: Crocodile MSRP - v1.0.0 Thank you.

2 17

Download SER SIP server
by Horo, Priyam 12 Jun '14

12 Jun '14

Hi, I cannot find the link to download the SER SIP server. I went to http://www.iptel.org/download, but do not see how to download 0.9.6 version. Could you please point me to the link. Thanks, Priyam.

3 2

check for the number of open branches in Reply
by Sebastian Damm 12 Jun '14

12 Jun '14

Hi, I have a scenario where I want to send a custom SIP package whenever a call is successfully established or definitely missed. This works pretty well, except when a user has multiple devices online. I need to send exactly one packet. When initiating the call I set a flag, in the reply route I check for that flag, and on the first 4xx or 2xx response the packet gets sent out. After that I reset the flag, so no other packet is generated for another response coming in. Now when a user has two devices and rejects the call on the first one but accepts it on the second, I see this call as missed, not answered. So I need to use the first 2xx response or the last 4xx response. For that I need to find out how many branches are open at the moment. But I'm missing something. I registered two devices on the same AOR. I tried accessing $branch(count) in the onreply route, but it is always 0. $br and $bR are <null>. Does anyone have an idea, how to get that information in the reply_route? Best Regards, Sebastian

3 5

planning the release of v4.1.4
by Daniel-Constantin Mierla 12 Jun '14

12 Jun '14

Hello, I am considering to release v4.1.4 next week, in one of the days within the interval June 10-12, depending on work that has to be done. If anyone is aware of issue not reported on tracker or patches that have not been backported, add to the tracker or write a message to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 4

Kamailio accepts and stores XCAP XML document from jitsi but rejects it on retrieval
by Alex Villacís Lasso 11 Jun '14

11 Jun '14

I am trying to set up XCAP support for SIP SIMPLE, and using jitsi as the sample client. The kamailio database was previously configured using ODBC and kamailio-4.1.3 and was apparently working correctly. I was following the guide at http://nil.uniza.sk/instant-messaging/simple/configuring-xcap-support-im-an… as well as the standard documentation at http://kamailio.org/docs/modules/4.1.x/modules/xcap_server.html . However, the rls support is currently disabled. When I use jitsi to connect, and add one contact to the empty contact list, I see the database being populated (mysql backend): mysql> select * from xcap; +----+-------------+-----------------+--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+----------+-----------------------+--------+-----------------------------------------------------------------------+------+ | id | username | domain | doc | doc_type | etag | source | doc_uri | port | +----+-------------+-----------------+--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+----------+-----------------------+--------+-----------------------------------------------------------------------+------+ | 3 | avillacisIM | pbx.elastix.com | <?xml version="1.0" encoding="UTF-8" standalone="no"?><resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"><list name="RootGroup"><entry uri="sip:bmacias@pbx.elastix.com"><display-name>bmacias</display-name></entry></list></resource-lists> | 4 | sr-1401316328-31413-1 | 0 | /xcap-root/resource-lists/users/sip:avillacisIM@pbx.elastix.com/index | 0 | | 4 | avillacisIM | pbx.elastix.com | <?xml version="1.0" encoding="UTF-8" standalone="no"?><ruleset xmlns="urn:ietf:params:xml:ns:common-policy"><rule id="presence_allow"><conditions><identity><one id="sip:bmacias@pbx.elastix.com"/></identity></conditions><actions><sub-handling xmlns="urn:ietf:params:xml:ns:pres-rules">allow</sub-handling></actions><transformations><provide-devices xmlns="urn:ietf:params:xml:ns:pres-rules"><all-devices/></provide-devices><provide-persons xmlns="urn:ietf:params:xml:ns:pres-rules"><all-persons/></provide-persons><provide-services xmlns="urn:ietf:params:xml:ns:pres-rules"><all-services/></provide-services></transformations></rule></ruleset> | 2 | sr-1401316328-31416-1 | 0 | /xcap-root/pres-rules/users/sip:avillacisIM@pbx.elastix.com/presrules | 0 | +----+-------------+-----------------+--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+----------+-----------------------+--------+-----------------------------------------------------------------------+------+ 2 rows in set (0.00 sec) As far as I can see, both XML documents are well formed. However, if I shutdown and then restart jitsi, I see (with wireshark) that the server returned a 500 Server Error when trying to fetch resource-lists . At the same time, the following appears on the log: May 28 17:39:12 elx /usr/sbin/kamailio[31421]: ALERT: <script>: ===== xhttp: request [HTTP/1.1] GET => /xcap-root/xcap-caps/global/index May 28 17:39:12 elx /usr/sbin/kamailio[31421]: ALERT: <script>: ===== xhttp: request [HTTP/1.1] GET => /xcap-root/xcap-caps/global/index May 28 17:39:12 elx /usr/sbin/kamailio[31421]: ALERT: <script>: ===== xhttp: xcap-caps : <null> May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ALERT: <script>: ===== xhttp: request [HTTP/1.1] GET => /xcap-root/resource-lists/users/sip:avillacisIM@pbx.elastix.com/index May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ALERT: <script>: ===== xhttp: request [HTTP/1.1] GET => /xcap-root/resource-lists/users/sip:avillacisIM@pbx.elastix.com/index May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ALERT: <script>: ===== xhttp: resource-lists : sip:avillacisIM@pbx.elastix.com May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ERROR: xcap_server [xcap_server.c:758]: xcaps_get_db_doc(): invalid xml doc retrieved from database May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ERROR: xcap_server [xcap_server.c:1171]: w_xcaps_get(): could not fetch xcap document What can I do to solve this issue. This is preventing contacts from being retrieved or added once the first contact is stored. My event-route looks like this (WITH_XCAPSRV is defined elsewhere in the configuration): #!ifdef WITH_XHTTP event_route[xhttp:request] { #!ifdef WITH_XHTTP_RPC $var(xhttp_rpc_root) = $(hu{s.substr,0,9}); if ($var(xhttp_rpc_root) == "/http_rpc") { dispatch_xhttp_rpc(); } #!endif #!ifdef WITH_XHTTP_PI $var(xhttp_rpc_root) = $(hu{s.substr,0,8}); if ($var(xhttp_rpc_root) == "/http_pi") { dispatch_xhttp_pi(); } #!endif #!ifdef WITH_XCAPSRV if ($hu =~ "^/xcap-root/") { route(XCAPSRV); } #!endif set_reply_close(); set_reply_no_connect(); # if ($Rp != 80 ##!ifdef WITH_TLS # && $Rp != 443 ##!endif # ) { # # xlog("L_WARN", "HTTP request received on $Rp\n"); # xhttp_reply("403", "Forbidden", "text/html", "Forbidden"); # exit; # } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } #!endif #!ifdef WITH_XCAPSRV route[XCAPSRV] { xlog("L_ALERT", "===== xhttp: request [$rv] $rm => $hu\n"); # Remove port specification from Host header and use as authentication domain $var(http_domain) = $(hdr(Host){s.select,0,:}); if (!www_authorize("$var(http_domain)", "subscriber")) { www_challenge("$var(http_domain)", "0"); exit; } set_reply_close(); set_reply_no_connect(); if ($hu=~"^/xcap-root/oma_status-icon/users/.*/.*$") $var(doc_uri) = $(hu{re.subst,/(^\/xcap-root\/)oma_status-icon(\/users\/.*\/).*$/\1org.openmobilealliance.pres-content\2oma_status-icon\/preseImage/}); else $var(doc_uri) = $hu; # xcap ops $xcapuri(u=>data) = $var(doc_uri); if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else if($xcapuri(u=>xuid)=~".+@.+") $var(uri) = "sip:" + $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; xlog("L_ALERT", "===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); if ($xcapuri(u=>auid) == "xcap-caps") { if ($rm == "GET") { $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> <auid>org.openmobilealliance.user-profile</auid> <auid>org.openmobilealliance.pres-content</auid> <auid>org.openmobilealliance.search</auid> <auid>org.openmobilealliance.xcap-directory</auid> </auids> <extensions></extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> <namespace>urn:oma:xml:xdm:user-profile</namespace> <namespace>urn:oma:xml:prs:pres-content</namespace> <namespace>urn:oma:xml:xdm:search</namespace> <namespace>urn:oma:xml:xdm:xcap-directory</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "OK", "application/xcap-caps+xml", "$var(xbody)"); } else { append_to_reply("Allow: GET\r\n"); xhttp_reply("405", "Method Not Allowed", "", ""); } exit; } # be sure auth user access only its documents if ($au != $(var(uri){uri.user})) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>$si:$sp</body></html>"); exit; } switch($rm) { case "PUT": xcaps_put("$var(uri)", "$var(doc_uri)", "$rb"); if ($xcapuri(u=>auid) =~ "pres-rules") { pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); # } else if ($xcapuri(u=>auid) =~ "rls-services" || $xcapuri(u=>auid) =~ "resource-lists") { # rls_update_subs("$var(uri)", "presence"); } else if ($xcapuri(u=>auid) =~ "pidf-manipulation") { pres_refresh_watchers("$var(uri)", "presence", 2, "$xcapuri(u=>uri_adoc)", "$xcapuri(u=>file)"); } exit; break; case "GET": xcaps_get("$var(uri)", "$var(doc_uri)"); exit; break; case "DELETE": xcaps_del("$var(uri)", "$var(doc_uri)"); if ($xcapuri(u=>auid) =~ "pres-rules") { pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); # } else if ($xcapuri(u=>auid) =~ "rls-services" || $xcapuri(u=>auid) =~ "resource-lists") { # rls_update_subs("$var(uri)", "presence"); } else if ($xcapuri(u=>auid) =~ "pidf-manipulation") { pres_refresh_watchers("$var(uri)", "presence", 2, "$xcapuri(u=>uri_adoc)", "$xcapuri(u=>file)"); } exit; break; case "POST": if ($xcapuri(u=>auid) =~ "search") { xhttp_reply("501", "Not Implemented", "", ""); } else { if ($xcapuri(u=>auid) =~ "xcap-directory") { append_to_reply("Allow: GET\r\n"); } else { append_to_reply("Allow: DELETE, GET, PUT\r\n"); } xhttp_reply("405", "Method Not Allowed", "", ""); } exit; break; } # other http requests xhttp_reply("404", "Not Found", "", ""); exit; } #!endif

2 11

Re: [SR-Users] Kamailio and Asterisk
by Carlos Rangel 11 Jun '14

11 Jun '14

Found the problem per a previous post. I had uncommented in kamctlrc by mistake. #STANDARD_MODULES="standard acc lcr domain group permissions registrar usrloc msilo # alias_db uri_db speeddial avpops auth_db pdt dialog dispatcher # dialplan" Thanks CR De: Carlos Rangel [mailto:crangel@globaltelesourcing.com] Enviado el: miércoles, 11 de junio de 2014 02:19 p.m. Para: sr-users(a)lists.sip-router.org Asunto: Kamailio and Asterisk Hello Trying to do my first Kamailio and Asterisk integration following the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I am using Centos though. Everything goes well except when trying to create the kamailio DB. The core tables cannot be created. Any ideas how to solve this problem??. Thanks CR [root@gt-ev-i08 kamailio-4.1.3]# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset INFO: creating database kamailio ... INFO: granting privileges to database kamailio ... INFO: creating standard tables into kamailio ... /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql: line 264: /usr/local/share/kamailio//mysql/#-create.sql: No such file or directory ERROR: Creating core tables failed at #!

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sr-users June 2014