- sr-users - kamailio.org

Kamailio and Freepbx Integration
by Carlos Rangel 16 Jun '14

16 Jun '14

Hello List I am trying to intgrate Kamailio latest release 4.1.4 and Freepbx 5.2.11 Asterisk 11.0.9 in the same box on an internal network following Daniel´s tutorial (http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb ) . I am very new with Kamailio and I tried like 15 times, seriously. The difference is that I am not using the sipusers table nor sipregs (I may later use sipregs) nor the voicemail tables (Freepbx can manage voicemail). I believe I was able to modify the kamailio.cfg file (DBURL and DBASTURL) in order to fetch the user information from the Freepbx tables. There is a table called devices that contents all the information needed. I also managed to not fill the secret parameter (password) on the extension configuration so Asterisk does not ask for authentication twice as recommended in the tutorial. I put the password in other table and then tell Kamailio to fetch it from the corresponding table. I can control if the users are registered in the system as long as the user is created in Freepbx with the right password, again password is not on the secret column. The weird thing is that my users can register in Asterisk even if they are not created in Kamailio (kamctl add). I was expecting them not to register if they are not created on the database. I have configured Freepbx to bind to a specific internal IP and port 5080 and configures kamailio as in the post #!ifdef WITH_ASTERISK asterisk.bindip = "xxx.xxx.xxx.xxx" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "xxx.xxx.xxx.xxx" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif Also I have added to kamailio.cfg #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_ANTIFLOOD I dont think this is the normal behavior, does anybody have an insight of what could be happening?? MY goal is to add the #!define WITH_NAT functionality (I havent done it yet)and then have remote trusted users connected to the PBX. Of course the concerns is to keep hackers away from my system. Thanks in advance for your help and for this great piece of software!!!! Regards and greetings from Mexico Carlos

1 0

stampeeding server.
by Måns Nilsson 16 Jun '14

16 Jun '14

Hi, I've got a 3.3.2 Kamailio that from time to time gets very upset and starts rushing these log entries to syslog: Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore The rate is about 10000 lines per second. This fills the log partition, which locks other things up and the users are unhappy. The server recovers nicely if restarted. I have scarce logs because I have to remove logs to get service back. Any suggestions? More data on request. -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 HELLO, everybody, I'm a HUMAN!!

2 3

Kamailio help
by Nouri El Sebai (Student) 16 Jun '14

16 Jun '14

Dear Sir/Madam, I am trying to connect (in Lebanon) as a subscribed user on Kamailio that has been installed on a remote server in the US. I am being faced with the error: Forbidden (403) Reason: Not relaying Kindly advise what I should do. Best regards, Nouri El Sebai

2 1

rtpproxy_manage and ip addresses
by tonchek＠mac.com 13 Jun '14

13 Jun '14

HI to ALL! As I understand the logic of rpproxy_manage work, it uses packet source IP and port number from m= field of SDP as socket for rtpproxy process. So, we can see it in SIP and rtpproxy debug messages: IP 77.94.133.231.5060 > 77.94.133.232.5060: UDP, length 1170 [...headers stripped...] v=0 o=root 1523540081 1523540081 IN IP4 188.40.66.75 s=Asterisk PBX 1.6.1.20 c=IN IP4 188.40.66.75 t=0 0 m=audio 15260 RTP/AVP 8 0 101 [...sdp stripped...] Jun 13 21:38:09 puma rtpproxy[4728]: INFO:handle_command: new session on a port 57680 created, tag FppNrF9K7S8gF;1 Jun 13 21:38:09 puma rtpproxy[4728]: INFO:handle_command: pre-filling caller's address with 77.94.133.231:15260 where 172.22.11.16 is SIP packet source IP address and 17646 is the port number from SDP. Is it possible to change this behavior and use IP address from incoming SDP as well as port number? Thank you Anton

1 1

inbound kamailio dispatcher faulty routing of ack after 200 OK
by Gijs Kwakkel 13 Jun '14

13 Jun '14

client1 <> provider <> kamailio <> asterisk <> client2 when client1 calls client2, everything seems to work. sounds works. after a few seconds the call gets terminated. after tcpdump syslog and config checks i figured out that asterisk sends 200 OK through kamailio to the provider, after this the provider sends a ACK to kamailio, kamailio however sends this ACK to itself and not to asterisk. I added the config, syslog and wireshark output (converted)

2 6

Re: [SR-Users] Is t_on_failure with uac_req_send possible?
by Ian Lin 13 Jun '14

13 Jun '14

Hi Klaus, I've got the same problem on kamailio-4.1.0 and I couldn't see the sip messages reach to onreply route while using uac_req_send. Could you please give me a hint about this? Thanks! Cheers, Ian -- ------------------------ Ian Lin 林于巽 ------------------------

2 2

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

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