June 2014 - sr-users - kamailio.org

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 #!

1 0

Kamailio and Asterisk
by Carlos Rangel 11 Jun '14

11 Jun '14

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 #!

1 0

kamailio and app_python
by Victor Seva 11 Jun '14

11 Jun '14

Hi, is there any way to return values from a call to python_exec function orther than the default returned integer? Following code generates this error: Oct 1 09:08:23 zskio /usr/sbin/kamailio[21789]: ERROR: app_python [python_support.c:38]: python_exec2: Unhandled exception in the Python code: Oct 1 09:08:23 zskio /usr/sbin/kamailio[21789]: ERROR: app_python [python_support.c:78]: #011RuntimeError: no such function if (!python_exec("get_provider", $(var(clean_uri){s.prefixes}))) { xlog("L_ERR","Can not choose a provider for $rU\n"); sl_send_reply("503","Can not choose a provider for $rU"); exit; } def get_provider(self, msg, param): # snipped code... result = self.util.get_provider(prefixes, position, mode) if result is not None: LM_ERR("mode: %s position: %d result: %s" % (mode, position, result)) # this raises error msg.call_function('rewritehost', result['provider']['ip']) return 1 return -1 I tried to use headers as transport but this simple code does not work: Oct 1 09:45:31 zskio /usr/sbin/kamailio[22303]: ERROR: <script>: No header if (!is_present_hf("X-Test-header")) { xlog("L_ERROR", "No header\n"); } else { xlog("L_INFO", "Header added\n"); } def test_set_header(self, msg): msg.call_function('append_hf', 'X-Test-header: test\r\n') return 1 Thanks in advance.

4 6

many-to-one network topology hide with TOPOH
by Maciej Marczyński 11 Jun '14

11 Jun '14

Hi Everyone, recently I'm playing around with the topoh module but I couldn't find the functionality I'm looking for. I'd like to hide my network topology before the end-user but not only the IPs of all VIA and RR but also the amount of hops - so to map 4 VIAs to 1 different VIA etc. The idea is to reduce the size of packets as some of my subscribers suffers from packets fragmentation. Do you think it's possible now or maybe easy to implement? Best Regards, Maciej

2 1

Routing logic and record-routing
by DURECU, OLIVIER (OLIVIER) 11 Jun '14

11 Jun '14

Hello all, There is something I can't understand using Kamailio as proxy to route SUBSCRIBE/NOTIFY dialogs between a client and a server I do record_route() on the first SUBSCRIBE received by Kamailio The next SUBSCRIBE is well handled by Kamailio in the [WITHIN_DIALOG] bloc but the 200/OK no more contains record-route header So the 3rd SUBSCRIBE no more sets a relevant route and Kamailio rejets it with a 404 ? How to proceed so that 200/Ok within dialog still contains Record-route header ? Here are the piece of routing logic involved if (is_method("PUBLISH|SUBSCRIBE")) { record_route(); $ru = "sip:192.168.0.250:6070"; route(RELAY); exit; } And the as-is route[WITHINDLG] function route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(RELAY); } Thanks for your help Olivier

2 1

kamailio + rtpengine: damaged media (Red Hat Linux)
by Alexey Rybalko 11 Jun '14

11 Jun '14

Hello to all! I encountered strange issue with rtpengine: voice during a call is heard like random binary data. (Video freezes during a video call). *It goes fine during 2-3 seconds* before the media flow becomes glitched. It's related to SRTP calls only (RTP-SRTP, SRTP-RTP, SRTP-SRTP). RTP calls through mediaproxy are ok. Used platform: Red Hat Linux 6.4, rtpengine (3.3.0 ?, latest form GitHub), kamailio (4.2.0-dev6) + websocket module. I have successfully installed (manually) mediaproxy daemon, Kamailio module, iptables library and kernel module. Kernel module was loaded (lsmod lists it, there are file in /proc/mediaproxy/0) and iptables were configured as expected. Mediaproxy log looks fine except of a few suspicious messages "No support for kernel packet forwarding available". I'm not sure they are related to the mentioned problem but I would like to have working kernel support. Attached example is for SIP-to-websocket call. rtpengine confguration is: *KERNEL=yes TABLE=0 FALLBACK=no RTP_IP=10.xx.xx.xxLISTEN_NG=127.0.0.1:2223 <http://127.0.0.1:2223> * That would be great if you make any suggestions/comments on this issue. Need to mention that I haven't saw such problems on Debian Linux. Couple of other questions: - What does message "Call-ID to delete not found" means? - "x_tables" is mentioned as required in the documentation for rtpengine kernel module (xt_MEDIAPROXY). Seems to be for Debian only. Is it required for other distros? with best regards, Alexey

2 4

Location issue
by Igor Potjevlesch 11 Jun '14

11 Jun '14

Hello, I'm running Kamailio 4.1.3 and having a strange issue. After a restart of Kamailio, some username are still in memory and updated accordingly with usual re-REGISTER flow. But the entries in location db is deleted. Kamailio tries to do a SQL UPDATE which fails because of that. So, for each username which are not in location DB, I must do a "kamctl ul rm" and wait for the next REGISTER. The new REGISTER will challenge the UAC for a new Authorization which should result in an INSERT INTO instead of an UPDATE. Can someone help me to solve this? Regards, Igor.

2 6

Kamailio and Freepbx Integration
by Carlos Rangel 10 Jun '14

10 Jun '14

Hello I am working on a project which looks for integration between Kamailio with Freepbx. We want to have remote sip users connected to the Freepbx server which is behind a NAT Firewall. The remote users are dynamic so IP addresses are always changing so we want Kamailio to authenticate this users and then fREEPBX to handle the call flow. I just started and have done a few trials without success based on the Kamailio-Asterisk integration tutorials on the documentation page. From other post I have read I know the following: - Kamailio and Asterisk can be on the same or different servers. I would look for a integrating with different servers. - Freepbx does not use realtime, instead uses the .conf files however the user information is contained on the database tables, I just need to select the correct table and configure Kamailio to get the user information from there. - Since the system will work behind a Nat firewall I assume that need to include the -- #!define WITH_NAT--- functionality on the kamailio.cfg. - Freepbx or Asterisk should not ask for authentication from Kamailio--- although I am not sure how exactly to accomplish this. I am writing this post just to see if anyone has had a previous experience to share or someone can give me an insight about the tricks on this. I will be working on this on the next days and share my experiences. Thanks a lot for your help. Carlos Rangel

1 0

Crash Kamailio 4.1 (qm_debug_frag(): BUG: qm_*: prev. fragm. tail overwritten)
by Andrés Souto 10 Jun '14

10 Jun '14

Hello, Kamailio crashes with this message: May 29 12:03:45 kamfree-manager kamailio[8705]: : <core> [mem/q_malloc.c:159]: qm_debug_frag(): BUG: qm_*: prev. fragm. tail overwritten(c0c0c000, abcdefed)[0x7ff5a5031a98:0x7ff5a5031ac8]! May 29 12:03:45 kamfree-manager kamailio[8697]: ALERT: <core> [main.c:775]: handle_sigs(): child process 8705 exited by a signal 6 May 29 12:03:45 kamfree-manager kamailio[8697]: ALERT: <core> [main.c:778]: handle_sigs(): core was generated The problem is produced in this line of the script: $rU = $dbr(ra=>[0,0]); I'm using last commit in 4.1 branch (ad5235229f59ba5d1f4216e144291e110fee0211) on Debian wheezy. The backtrace: #0 0x00007ff5a538b475 in raise () from /lib/x86_64-linux-gnu/libc.so.6 #1 0x00007ff5a538e6f0 in abort () from /lib/x86_64-linux-gnu/libc.so.6 #2 0x0000000000546a00 in qm_debug_frag (qm=0x7ff5a4f58010, f=0x7ff5a5031a98) at mem/q_malloc.c:161 #3 0x000000000054796d in qm_malloc (qm=0x7ff5a4f58010, size=1024, file=0x5c9ec1 "<core>: action.c", func=0x5cb1d8 "do_action", line=832) at mem/q_malloc.c:386 #4 0x000000000041affa in do_action (h=0x7fff810cd2b0, a=0x7fff810cd390, msg=0x7ff5a502ed28) at action.c:832 #5 0x00007ff5a2295d88 in pv_set_ruri_user (msg=0x7ff5a502ed28, param=0x7ff5a4fb3750, op=254, val=0x7fff810cd4e0) at pv_core.c:2073 #6 0x0000000000467bbf in lval_pvar_assign (h=0x7fff810d0200, msg=0x7ff5a502ed28, lv=0x7ff5a501ea00, rv=0x7ff5a501f110) at lvalue.c:362 #7 0x00000000004680a5 in lval_assign (h=0x7fff810d0200, msg=0x7ff5a502ed28, lv=0x7ff5a501ea00, rve=0x7ff5a501f108) at lvalue.c:410 #8 0x0000000000423fec in do_action (h=0x7fff810d0200, a=0x7ff5a501eb68, msg=0x7ff5a502ed28) at action.c:1478 #9 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a5013ee0, msg=0x7ff5a502ed28) at action.c:1599 #10 0x000000000041c18d in do_action (h=0x7fff810d0200, a=0x7ff5a5020a10, msg=0x7ff5a502ed28) at action.c:1090 #11 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a5020a10, msg=0x7ff5a502ed28) at action.c:1599 #12 0x000000000041c18d in do_action (h=0x7fff810d0200, a=0x7ff5a5020e50, msg=0x7ff5a502ed28) at action.c:1090 #13 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a50119b0, msg=0x7ff5a502ed28) at action.c:1599 #14 0x000000000041a2b7 in do_action (h=0x7fff810d0200, a=0x7ff5a4fd3fc0, msg=0x7ff5a502ed28) at action.c:715 #15 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a4fd2800, msg=0x7ff5a502ed28) at action.c:1599 #16 0x0000000000422c70 in do_action (h=0x7fff810d0200, a=0x7ff5a4fd7af0, msg=0x7ff5a502ed28) at action.c:1235 #17 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a4fd1ed8, msg=0x7ff5a502ed28) at action.c:1599 #18 0x000000000041c18d in do_action (h=0x7fff810d0200, a=0x7ff5a4fd7d10, msg=0x7ff5a502ed28) at action.c:1090 #19 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a4fcf1d8, msg=0x7ff5a502ed28) at action.c:1599 #20 0x000000000041a2b7 in do_action (h=0x7fff810d0200, a=0x7ff5a4fb7d88, msg=0x7ff5a502ed28) at action.c:715 #21 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a4fb6b68, msg=0x7ff5a502ed28) at action.c:1599 #22 0x000000000041c18d in do_action (h=0x7fff810d0200, a=0x7ff5a4fb9838, msg=0x7ff5a502ed28) at action.c:1090 #23 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a4faf5a0, msg=0x7ff5a502ed28) at action.c:1599 #24 0x0000000000425524 in run_top_route (a=0x7ff5a4faf5a0, msg=0x7ff5a502ed28, c=0x0) at action.c:1685 #25 0x00000000004a6fdb in receive_msg ( buf=0x921920 "INVITE sip:2000@vpbx2.ur SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.0.101;rport;branch=z9hG4bK14cFrXZ0N53jS\r\nMax-Forwards: 67\r\nFrom: \"1000\" <sip:1000@vpbx1.ur>;tag=FjmFN79ZrXH0e\r\nTo: <sip:2000@vpbx2.ur>\r\nCall-"..., len=1061, rcv_info=0x7fff810d04b0) at receive.c:212 #26 0x000000000053c544 in udp_rcv_loop () at udp_server.c:536 ---Type <return> to continue, or q <return> to quit--- #27 0x000000000046ee54 in main_loop () at main.c:1617 #28 0x0000000000471df4 in main (argc=3, argv=0x7fff810d07e8) at main.c:2533 Complete log: http://pastebin.com/YZirUkfe Thanks. Regards, -- *Andrés Souto* VoIP Engineer @ Quobis <http://www.quobis.com/> | e: andres.souto(a)quobis.com | t: +34902999465

2 8

re-use Expires from 200 OK
by Daniel Ciprus 10 Jun '14

10 Jun '14

Is there any way to extract Expires from 200 OK (response to REGISTER) ? I will need to re-use this value in 3rd party registration being sent to application server. Thanks for any hints .. -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 1

Iptel.org cannot make call
by Tommy Yan 09 Jun '14

09 Jun '14

Dear Sir/Madam, We setup the Linksys SPA3000 to make a call via IPTEL.ORG in between my family members around the world. But recently, we found out that we cannot dial out. So, would you please help to check this ? Regards, Tommy.

2 1

kamailio dispatcher failover setup
by bajeesh＠itplus.in 09 Jun '14

09 Jun '14

Hi, I am first in kamailio,I have installed the kamailio from source code and added a user using kamctl,Now I can connect with that user from soft phone. My task is making a Load balancer and fail over for asterisk servers, I have some other asterisk servers, I need make the kamailio as a Load balancer, How can I do that? 1. I need to forward all calls to asterisks servers. 2. I have created a Dispatcher list in db, how can I configure that? Please help me on this. Hopefully, Bajeesh T.B

2 1

Kamailio on PPC
by ashwin 09 Jun '14

09 Jun '14

Hi, I tried to compile Kamailio on Freescale P2020RDB-PCA.And I'm getting the following error bison -d -b cfg cfg.y make: *** [cfg.tab.h] Broken Pipe When i run that command without running 'make'. Just 'bison -d -b cfg cfg.y'.It was working.So found from Internet that it could be the problem with memory when the bison process is running.So I allocated a swap file for the build process. dd if=/dev/zero of=swapfile.1 bs=1048576 count=100 ; mkswap ./swapfile.1 ; swapon ./swapfile.1 But after that attempt also I'm still getting the make: *** [cfg.tab.h] Broken Pipe when Im doing the compilation. Yours recommendations will be very helpful. Thanks Ashwin -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-on-PPC-tp128097.html Sent from the Users mailing list archive at Nabble.com.

2 1

How connect kamailio to remote redis server ?
by yann christophe 09 Jun '14

09 Jun '14

Hello, I have two machines , kamailio 1.1.1.1 and redis 1.1.1.2 this is my configuration in the kamailio.cfg for redis: modparam("ndb_redis", "server", "name=local;addr=1.1.1.2;port= 6379") but it's correct ? because i don't see my kamailio in my client list redis thanks for your help Yann

2 1

how to call the user of other sip proxy server
by 启程 09 Jun '14

09 Jun '14

Hi,I installed kamailio . The user of other sip proxy server can call me ,but I can not call them.I　want to call them ,how to config it ?

2 1

Branch feedback behaviour
by Alex Balashov 08 Jun '14

08 Jun '14

Hello, I've got a situation like this: UAC ---> (Kamailio) ----------> 1. redirect server on branch 1 <---------- 2. 302 redirect ----------> 3. ACK ----------------------------> UAC B Problem is, if CANCEL comes in at the exact moment that the INVITE to the redirect server (#1) is "in flight", i.e. has not returned a 302 yet, then the transaction gets cancelled and 302 is fed back to the calling UAC. Having the 302 fed back to the calling UAC in any circumstances is quite undesirable. I just need to drop it. Problem is, in all other situations, I need to use it--I consume it in a failure_route and create new branches based on it. What's the best way to accomplish this? I'm concerned that if I use a global onreply_route to filter all 302s going back to callers, that will prevent it from percolating down to the failure_route. Suggestions appreciated, and thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 3

SDP and NAT
by Marc Soda 06 Jun '14

06 Jun '14

Hey all, I have a Kamailio box functioning as a proxy. When some UAs send an INVITE the SDP has their private IP (c=IN IP4 10.x.x.x). Kamailio is passing this on without changing it to the proper public IP. What's the best way to rewrite that? I'm using a config based on the default one.

2 1

Webrtc: Don't catch 488 between JSSIP and SIP UA's
by LAA 06 Jun '14

06 Jun '14

Apologize. Previous message was too long. L. El 02/06/2014 20:25, "LAA" <ornitorrinco7424(a)gmail.com> escribió: > Hi all, > > Another guy strugling his mind trying to get a configuration to enable > calls between WebRTC UA (JSSIP) to standard SIP UA (Twinkle or SjPhone) > I've been working with the examples that were shared by Carlos Ruiz Diaz > and Peter Dunkley (thanks to both). > > http://www.slideshare.net/crocodilertc/webrtc-websockets > http://caruizdiaz.com/2014/02/26/webrtc-kamailio/ > > Kamailio is not running behind NAT. I'm using rtpproxy-ng module with > Kamailio 4.1.3, and Rtpengine. > > I share a link with my current configuration, wich is based in Peters > example, with websocket support from websocket.cfg example. > > - Calls between SIP standard UA's are working OK. I have some endpoint > behind nat. > - Calls between JSIP UA's are working OK. So, websocket support is > running. > - Calls from JSIP and Twinkle are NOT WORKING OK. sip UA send's back a > 488 response, and Kamailio send it back to JSSIP (Incompatible SDP). > - Calls from Twinkle to Jsip are NOT WORKING OK: Kamailio sends an INVITE > to JSIP, and it returns an error. And Kamailio sends 488 to Twinkle. > > > It seems as if Kamailio is not catching 488. I share a snippet of my > config, and links to tcpdump captures: > > https://www.dropbox.com/s/i7c9ty57oauujc4/fromws0.pcap > https://www.dropbox.com/s/q3q30pgzvdoswts/kamailio.cfg > https://www.dropbox.com/s/rqtjwcbgg1foaoq/tows0.pcap > > What am I missing? > > > Best regards. > > Luis. > > > > > > > >

1 1

HOW CONNECT KAMAILIO TO REMOTE REDIS SERVER
by yann christophe 06 Jun '14

06 Jun '14

Hello, I have two machines , kamailio 1.1.1.1 and redis 1.1.1.2 this is my configuration in the kamailio.cfg for redis: modparam("ndb_redis", "server", "name=local;addr=1.1.1.2;port=6379") but it's correct ? because i don't see my kamailio in my client list redis thanks for your help Yann

1 0

CANCEL requests
by Andrea Meroni 06 Jun '14

06 Jun '14

Hi all, just a simple question: which is the timer that triggers if no reply is received for the current transaction and makes Kamailio send a CANCEL request? Andrea

2 2

many-to-one network topology hide with TOPOH
by Maciej Marczyński 06 Jun '14

06 Jun '14

Hi Everyone, recently I'm playing around with the topoh module but I couldn't find the functionality I'm looking for. I'd like to hide my network topology before the end-user but not only the IPs of all VIA and RR but also the amount of hops - so to map 4 VIAs to 1 different VIA etc. The idea is to reduce the size of packets as some of my subscribers suffers from packets fragmentation. Do you think it's possible now or maybe easy to implement? Best Regards, Maciej

2 1

FS#439 - CRASH: segmentation fault after INVITE
by Nuno Reis 06 Jun '14

06 Jun '14

Hi. I've just opened a bug report FS#439 on a segmentation fault after an invite to a uknown DNS domain. It is reproducible and is happening in kamailio 4.1.3 (642428c) Please let me know if anything else is needed. Thanks. -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

1 0

Listen on eth alias interface
by Olle E. Johansson 05 Jun '14

05 Jun '14

Hi! I have an interface which is an alias in Linux that I want Kamailio to listen to. The interface name is eth0:p01 I've tried with listen=udp:eth0:p01:5080 listen=eth0:p01:5080 listen=eth0:p01 Kamailio doesn't accept this syntax. Claims "syntax error" and " ip address, interface name or hostname expected". Since this way of naming interfaces is common in Linux, I think we should support it. Or? Is there a way - are quotation marks allowed here? Cheers, /O

3 8

MySQL query result pagination
by Allen Zhang 05 Jun '14

05 Jun '14

Hi, When an user does a REGISTER and has a large number of msilo messages stored, say 7k, kamailio server easily runs out of private memory when trying to allocate memory to the result set. I know I can set a max on the messages a user can store, but I'd like to support a crazy number like 7k messages. I'm trying to introduce paging to the query. So that I can process N number of results at one time. I can't find an easy way of doing this with the current MySQL driver. The db_mysql_query() function doesn't have built-in paging and with this function it's not easy to build a complex query like this: Select * from silo where ...(the matching algorithm) AND id > lower_limit AND id < higher_limit, ordered by id. Is there a way to do this? Or maybe I'm on the wrong track and there is already a solution to this problem? Regards, Allen Zhang

2 4

Re: [SR-Users] MySQL query result pagination
by Allen Zhang 05 Jun '14

05 Jun '14

I found this: SELECT column FROM table LIMIT N OFFSET offset Increment offset by index*N in the loop to fetch N records at a time. But the remaining question is how do I do this using kamailio's MySQL driver? Raw query? Allen From: Allen Zhang Sent: Thursday, 5 June 2014 12:19 p.m. To: 'sr-users(a)lists.sip-router.org' Subject: MySQL query result pagination Hi, When an user does a REGISTER and has a large number of msilo messages stored, say 7k, kamailio server easily runs out of private memory when trying to allocate memory to the result set. I know I can set a max on the messages a user can store, but I'd like to support a crazy number like 7k messages. I'm trying to introduce paging to the query. So that I can process N number of results at one time. I can't find an easy way of doing this with the current MySQL driver. The db_mysql_query() function doesn't have built-in paging and with this function it's not easy to build a complex query like this: Select * from silo where ...(the matching algorithm) AND id > lower_limit AND id < higher_limit, ordered by id. Is there a way to do this? Or maybe I'm on the wrong track and there is already a solution to this problem? Regards, Allen Zhang

1 0

CNXCC and multiple servers
by Kelvin Chua 04 Jun '14

04 Jun '14

I was just playing around with the dmq module when i thought of an idea, cnxcc being able to track users across servers. cnxcc would have to be aware of dialogs created on another server. is this something doable? Kelvin Chua

2 1

mediaproxy
by Slava Bendersky 03 Jun '14

03 Jun '14

Hello Everyone, I am trying setup mediappoxy on the gateway and log get fill up with this message. Jun 3 12:28:11 /usr/sbin/kamailio[3595]: ERROR: rtpproxy-ng [rtpproxy.c:1425]: rtpp_test(): proxy responded with invalid response Jun 3 12:28:11 /usr/sbin/kamailio[3597]: ERROR: rtpproxy-ng [rtpproxy.c:1425]: rtpp_test(): proxy responded with invalid response Jun 3 12:17:59 mediaproxy-ng[3294]: Failed to properly parse UDP command line '3394_0 d7:command4:pinge' from 127.0.0.1:59326, using fallback RE Jun 3 12:17:59 mediaproxy-ng[3294]: Failed to properly parse UDP command line '3402_1 d7:command4:pinge' from 127.0.0.1:51867, using fallback RE [root@sipgw01 kamailio]# rpm -qa | grep kam kamailio-4.1.3-2.1.x86_64 Any help thank you Slava.

3 3

cxdx_forced_peer - more info
by Daniel Ciprus 03 Jun '14

03 Jun '14

Hi, Could any of you please point me to the doc which explains more in details parameter cxdx_forced_peer ? The only place I found is http://kamailio.org/docs/modules/stable/modules/ims_icscf.html#idp190000. thanks -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 2

DB ERROR using presence status persistence
by Nuno Reis 03 Jun '14

03 Jun '14

Hi guys. I've setup presence management in kamailio and i'm currently persisting the presence info in a postgres DB. I'm having a recurring issue on DB key violations, here's a log error output on that: Jun 3 15:09:46 144227 ERROR: pua_dialoginfo [pua_dialoginfo.c:264]: __dialog_sendpublish(): bad reply or missing CONTACT hdr Jun 3 15:10:01 723993 ERROR: presence [presentity.c:807]: update_presentity(): No E_Tag match a.1401768964.671509.1990.6 Jun 3 15:10:06 242334 WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) Jun 3 15:10:06 242370 WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [ERROR: duplicate key value violates unique constraint "pua_pua_idx"#012DETAIL: Key (etag, tuple_id, call_id, from_tag)=(a.1401768964.671497.1902.6, , , ) already exists.#012] Jun 3 15:10:06 243415 WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) Jun 3 15:10:06 243449 WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [ERROR: duplicate key value violates unique constraint "pua_pua_idx"#012DETAIL: Key (etag, tuple_id, call_id, from_tag)=(a.1401768964.671497.1902.6, , , ) already exists.#012] Jun 3 15:10:06 244458 WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) Jun 3 15:10:06 244487 WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [ERROR: duplicate key value violates unique constraint "pua_pua_idx"#012DETAIL: Key (etag, tuple_id, call_id, from_tag)=(a.1401768964.671497.1902.6, , , ) already exists.#012] Jun 3 15:10:06 244504 ERROR: db_postgres [km_dbase.c:247]: db_postgres_submit_query(): 0x7f21476b1910 PQsendQuery Error: ERROR: duplicate key value violates unique constraint "pua_pua_idx"#012DETAIL: Key (etag, tuple_id, call_id, from_tag)=(a.1401768964.671497.1902.6, , , ) already exists.#012 Query: update kamailio.pua set expires=1401807715,cseq=0,etag='a.1401768964.671497.1902.6',desired_expires=1401811600,version=2 where pres_uri='sip:208@freguesiadearroios.govoice.pt' AND pres_id='DIALOG_PUBLISH.440163220-35219-52(a)BJC.BGI.B.BAI' AND flag=1024 AND event=32 Jun 3 15:10:06 244521 ERROR: <core> [db_query.c:337]: db_do_update(): error while submitting query Jun 3 15:10:06 244548 ERROR: db_postgres [km_dbase.c:537]: db_postgres_store_result(): invalid query, execution aborted Jun 3 15:10:06 244560 ERROR: db_postgres [km_dbase.c:538]: db_postgres_store_result(): driver error: PGRES_FATAL_ERROR, ERROR: duplicate key value violates unique constraint "pua_pua_idx"#012DETAIL: Key (etag, tuple_id, call_id, from_tag)=(a.1401768964.671497.1902.6, , , ) already exists.#012 Jun 3 15:10:06 244584 WARNING: db_postgres [km_dbase.c:647]: db_postgres_update(): unexpected result returned Jun 3 15:10:06 244591 ERROR: pua [pua.c:1076]: db_update(): while updating in database I've read the module documentation, but there isn't much documentation on it and i'm basically guessing and trying here. Can someone tell me what am i doing wrong or maybe what should a try to be able to persist the presence status in a postgres database. I know that i could use in memory persistence or other type of DB, but i really want to know if i can use a normal postgres DB for presence persistence and what could be used to avoid key violations. Looking forward to hear from you. Best Regards, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

1 1

ODP: cnxcc
by Andrzej Ciupek 03 Jun '14

03 Jun '14

Yes. I want prepaid subscribers be able to create only one channel to prevent frauds for simultaneous calls. Greetings Andrzej Ciupek ________________________________ Od: Carlos Ruiz Díaz<mailto:carlos.ruizdiaz@gmail.com> Wysłano: ‎2014-‎04-‎26 16:42 Do: Andrzej Ciupek<mailto:andrzej.ciupek@hotmail.com> DW: Kamailio (SER) - Users Mailing List<mailto:sr-users@lists.sip-router.org> Temat: Re: [SR-Users] cnxcc Hi Andrzej, are you trying to setup cnxcc to check both credit and maximum simultaneous calls? Regards, On Sat, Apr 26, 2014 at 9:26 AM, Andrzej Ciupek <andrzej.ciupek(a)hotmail.com>wrote: > Hello > > I’am using default config with options from an example. It looks like > event route isn’t execuded after hangup. > > Apr 26 16:26:14 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: > 1000 has 1 call(s) > Apr 26 16:26:15 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: > 1000 has 1 call(s) > Apr 26 16:26:16 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: > 1000 has 1 call(s) > Apr 26 16:26:17 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: > 1000 has 1 call(s) > Apr 26 16:26:18 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: > 1000 has 1 call(s) > Apr 26 16:26:19 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: > 1000 has 1 call(s) > Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: > 1000 has 1 call(s) > Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4604]: INFO: <script>: > Setting up money based credit control > Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4604]: INFO: <script>: > Setting up channel based credit control > Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4604]: INFO: <script>: > CNXCC ROUTE: 1000 has 2 call(s) > > kamailio.cfg > … > … > # ----- prepaid call controll ----- > > #!define DLG_FLAG 28 > #!define CC_FLAG 29 > > loadmodule "dialog.so" > > modparam("dialog", "hash_size", 2048) > modparam("dialog", "default_timeout", 3600) > modparam("dialog", "db_mode", 0) > modparam("dialog", "dlg_flag", DLG_FLAG) > > modparam("rtimer", "timer", "name=ta;interval=1;mode=1;") > modparam("rtimer", "exec", "timer=ta;route=SHOW_CHANNEL_COUNT") > > loadmodule "cnxcc.so" > > modparam("cnxcc", "dlg_flag", CC_FLAG) > modparam("cnxcc", "credit_check_period", 1) #check every 1 second > > … > … > … > route { > … > … > if (!allow_source_address("2") and !allow_source_address("1")){ > route(CNXCC); > route(PSTN); > } > … > } > > route[CNXCC] > { > > $var(client) = "1000"; > > xlog("L_INFO", "Setting up money based credit control"); > > $var(credit) = "10"; # 10$ of credit > $var(cost_per_sec) = "1"; # 1$ per sec > $var(i_pulse) = "1"; # 1$ to establish the call > $var(f_pulse) = "1"; # 1$ per second > > # if only one call is established, that call should last 9 seconds. > > if (!cnxcc_set_max_credit("$var(client)", > "$var(credit)", > "$var(cost_per_sec)", > "$var(i_pulse)", > "$var(f_pulse)")) { > xlog("Error setting up credit control"); > return; > } > $var(count) = -1; > > if (!cnxcc_get_channel_count("$var(client)", "$var(count)")) { > xlog("Error getting customer's channel count"); > } > > xlog("L_INFO", "CNXCC ROUTE: $var(client) has $var(count) > call(s)"); > > if ($var(retcode) < -1) { > xlog("Too many channels for customer"); > sl_send_reply(403, "Forbidden"); > > if (!cnxcc_terminate_all("$var(client)")) { > xlog("Error terminating customer's calls"); > } else { > xlog("Terminating customer's call"); > } > > exit; > } > } > > route[SHOW_CHANNEL_COUNT] > { > $var(count) = @cnxcc.channels["1000"].count; > xlog("L_INFO", "1000 has $var(count) call(s)"); > } > > event_route[dialog:failed] > { > xlog("dialog failed"); > } > > event_route[cnxcc:call-shutdown] > { > xlog("L_INFO", "Shutting down channel by callcontroll"); > xlog("L_INFO", "[$ci]: call killed"); > $var(count) = $var(count)-1; > if (!cnxcc_get_channel_count("$var(client)", "$var(count)")) { > xlog("Error getting customer's channel count"); > } > xlog("L_INFO", "CNXCC ROUTE: $var(client) has $var(count) > call(s)"); > # perform some kind of notification, database update, email > sending, etc > } > > Greetings > Andrzej Ciupek > > *Od:* Carlos Ruiz Díaz <carlos.ruizdiaz(a)gmail.com> > *Wysłano:* ‎sobota‎, ‎26‎ ‎kwietnia‎ ‎2014 ‎15‎:‎19 > *Do:* Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org>, > Andrzej Ciupek <andrzej.ciupek(a)hotmail.com> > > Hi, > > sorry guys, I forgot about this. I will make my own tests and post the > results. > > Andrzej, can you post the configuration file you are using? > > Regards, > Carlos > > > On Sat, Apr 26, 2014 at 4:15 AM, Kelvin Chua <kelchy(a)gmail.com> wrote: > >> No, I have not yet found the solution for this issue on my setup. >> >> Maybe Carlos can help us out >> On Apr 26, 2014 5:10 PM, "Andrzej Ciupek" <andrzej.ciupek(a)hotmail.com> >> wrote: >> >>> Hello >>> >>> Have You solved this issue ? >>> I’am using Kamailio 4.1.3 and I’am getting the same results. >>> >>> Greetings >>> Andrzej Ciupek >>> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > > -- > Carlos > http://caruizdiaz.com > http://ngvoice.com > +52 55 3048 3303 > -- Carlos http://caruizdiaz.com http://ngvoice.com +52 55 3048 3303

3 10

what is the difference between outbound module and stun module
by vast 03 Jun '14

03 Jun '14

Hi, what is the difference between （make FLAVOUR=kamailio cfg STUN=1 outbound module ）and （stun module）？ and the debug log has this： DEBUG: <core> [sr_module.c:707]: find_mod_export_record(): find_export_record: <bind_ob> not found INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob INFO: rr [rr_mod.c:159]: mod_init(): outbound module not available can you tell me about it , Thank you

1 0

sqlops
by Kelvin Chua 03 Jun '14

03 Jun '14

what is wrong with this sql query? it works outside kamailio. select s.* from (select @p1:=4088112345 p) parm , lrn.lrnx s; ERROR: sqlops [sql_api.c:435]: sql_exec_xquery(): cannot do the query Kelvin Chua

2 2

app_python not able to import from Router
by Örn Arnarson 02 Jun '14

02 Jun '14

Hello, I've tried installing kamailio both from deb and from source (4.0.4 in both cases) on Ubuntu server 14.04. In both cases, I'm having problems with importing from the Router module in python, which I presume is made available when invoked from kamailio with the python module loaded. Here's what I see when trying to start kamailio with a python script with "from Router import LM_ERR": May 28 16:34:13 vcr01 kamailio: NOTICE: <core> [sr_module.c:627]: load_module(): /usr/lib/x86_64-linux-gnu/kamailio/modules/app_python.so: exports dlflags interface is deprecated and it will not be supported in newer versions; consider using mod_register() instead May 28 16:34:13 vcr01 /usr/sbin/kamailio[19652]: INFO: rr [../outbound/api.h:49]: ob_load_api(): Failed to import bind_ob May 28 16:34:13 vcr01 /usr/sbin/kamailio[19652]: INFO: rr [rr_mod.c:159]: mod_init(): outbound module not available May 28 16:34:13 vcr01 /usr/sbin/kamailio[19652]: ERROR: app_python [python_support.c:132]: python_handle_exception(): mod_init: Unhandled exception in the Python code:#012Traceback (most recent call last):#012 File "/etc/kamailio/callrouting.py", line 2, in <module>#012 from Router import LM_ERR#012ImportError: cannot import name LM_ERR May 28 16:34:13 vcr01 /usr/sbin/kamailio[19652]: ERROR: <core> [sr_module.c:939]: init_mod(): init_mod(): Error while initializing module app_python (/usr/lib/x86_64-linux-gnu/kamailio/modules/app_python.so) I load the module so: loadmodule "app_python.so" and set the following parameters: modparam("app_python", "script_name", "/etc/kamailio/handler.py") modparam("app_python", "mod_init_function", "mod_init") What am I doing wrong? Am I missing some additional modules from kamailio? handler.py is just the sample python script provided with the package. Any pointers appreciated. Regards, Örn

3 3

JSSIP with Kamalio with SIP end points
by Steve Ng 02 Jun '14

02 Jun '14

Hi, My users are from a web application using JSSIP. They each have a sip trunk account, which could be from any provider. JSSIP would connect to Kamailio's websocket server passing the SIP trunk credential information and Kamailio will relay it to that specific sip trunk provider? Is there anyway that I could use Kamailio as the middleman? Is it a combination of dispatcher module and websocket module? Thanks! Regards, Steve

2 5

drouting failure loop
by Pete Ashdown 02 Jun '14

02 Jun '14

I'm trying to get my head around how to do a failure loop in droute through all possible destinations, and if all destinations are exhausted, go to failureroute. Does anyone have an example of this? Without the ability to do a while on (!t_relay()) to determine whether to go to use_next_gw(), I'm not sure how I can. Thanks in advance.

2 1

new ngcp-rtpengine release?
by Juha Heinanen 01 Jun '14

01 Jun '14

i noticed that there is new entry in ngcp-rtpengine changelog: ngcp-rtpengine (3.3.0.0+0~mr3.4.0.0) unstable; urgency=low does that mean that there is new stable release 3.3? what does ~mr3.4.0.0 mean? -- juha

2 1

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sr-users June 2014