- sr-users - kamailio.org

SIREMIS: Configure SIP Trunk/DID/Extension
by Stretchem 03 Dec '13

03 Dec '13

REPOST: Have successfully installed the latest stable Kamailio and Siremis. Have perused countless online Wiki's and other documentation, but can't seem to find any definitive tutorials or answers to the following: 1. Where would I set up a SIP registration to my carrier? (Server Services -> UACReg List?) 2. Where or how would I see the result of that registration? 3. Where would I set up DID's from the above carrier? 4. How would I inbound route those DID's to specific SIP extensions? (Have successfully registered extensions under Subscriber Services -> Subscriber List) 5. How would I outbound route calls from extensions to the carrier in #1? As I understand it, Kamailio settings are stored in a database; Siremis is a web gui to that database. If that is true, then I don't mind knowing the accompanying 'kamctl' commands to accomplish the above, since I'll see the respective settings in the Siremis web gui anyways. Also don't mind paying for the above tutorial, for which I'll turn into a Googlable blog entry. I need a routine or process for which non-CLI associates can make entries/edits. SIP Trunk with DIDs <--> Siremis on Kamailio <--> SIP Extension Thanks in advance.

1 0

deb specs for 4.1
by Daniel-Constantin Mierla 03 Dec '13

03 Dec '13

Initially sent to sr-dev, but maybe some people from users community can help as well... Hello, wondering if any debian/ubuntu fan has some time today/tomorrow morning for updating the specs to include the new modules coming with 4.1. The list is here: - http://www.kamailio.org/wiki/features/new-in-4.1.x Fortunately Peter updated the specs he maintains for rpm-based distros - the one for centos 6 might be useful to figure out dependencies - see in source tree pkg/kamailio/centos/6/. Reply quickly here if you start working on them -- there can be many people, I just want to know how I plan my effort to prepare the release. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Hello everyone.
by Yurisbel Jimenez Bermudez 03 Dec '13

03 Dec '13

I'm trying to setup Kamailio everything install correctly but I can't connect to my server with the users created. Not even in my local network it seems like the machine doesn't respond to the packets or maybe the packets are not delivery to my server but when I ping the machine everything is good I'm redirecting the port 5060 to my server same as range ports from 10000 to 20000 but can't connect Any one knows which is the problem here? I'm using Kamailio 4.0 on a virtual machine with virtual box. A copy of the same machine its running a web server of course with a different mac and IP and everything works fine.

2 1

Help required regarding error in Installation
by Owais ul Haq 03 Dec '13

03 Dec '13

Hello all, Have a nice day. I am given a task in office to setup Kamailio-3.1.1 on Fedora 14 machine. Being new to this, I went online for guidance. I got a proper method on the link given below and followed the instructions given. http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git But problem occured when I reached on step 4 i.e "COMPILE KAMAILIO". When I run command "MAKE ALL". Error occurs that asks for file "pcre.h" as shown in photo attached. Can some one please guide me what to do now ? Any help will be appreciated. Thanks. Regards, Kamailio newbie, Owais.

2 2

401 Unauthorized
by Joli Martinez 02 Dec '13

02 Dec '13

Hello all, I am trying to register a sip client to Kamailio. I have added the username password to kamctl and verified it in the opeser db that the user and password are correct. I have other users registered but I am unable to add new users. I keep getting the message below from tcpdump. Also kamctl online is not accurate. I have two other users connected and they dont show up there. [bad udp cksum a8b7!] UDP, length 433 E.......@.......2..........hSIP/2.0 401 Unauthorized Please help Thanks,

2 2

New developer: Luis Martin Gil
by Daniel-Constantin Mierla 02 Dec '13

02 Dec '13

Hello, I want to announce that a new person got developer GIT write access to repository: Luis Martin Gil. He has developed a new module named ndb_cassandra, which offers a connector to cassandra no-sql system directly from configuration style (e.g., for getting/setting items). The module is already part of the master branch since few days ago. His git commit id is: luismartingil My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Next Kamailio World - April 2-4, 2014, in Berlin, Germany
by Daniel-Constantin Mierla 02 Dec '13

02 Dec '13

Hello, I would like to quickly announce the dates and place for next Kamailio World Conference: April 2-4, 2014, in Berlin, Germany. More information is available on the web site: - http://conference.kamailio.com/k01/2013/12/kamailio-world-2014-april-2-4-in… More details will be published in the near future. Looking forward to meeting many of you at the event in Berlin next year, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

forwarding doesn't work
by 1717 02 Dec '13

02 Dec '13

> > Greetings > > 1. Calling sip: mail(a)idd1717.com doesn't ring the forwarded sip uri, any idea? > 2. Can an sip alias name be enabled to be used for sip registration, just like the original user name? > > Thanks > James Wong

2 1

SIREMIS: Configure SIP Trunk/DID/Extension
by Stretchem 01 Dec '13

01 Dec '13

Have successfully installed the latest stable Kamailio and Siremis. Have perused countless online Wiki's and other documentation, but can't seem to find any definitive tutorials or answers to the following: 1. Where would I set up a SIP registration to my carrier? (Server Services -> UACReg List?) 2. Where or how would I see the result of that registration? 3. Where would I set up DID's from the above carrier? 4. How would I inbound route those DID's to specific SIP extensions? (Have successfully registered extensions under Subscriber Services -> Subscriber List) 5. How would I outbound route calls from extensions to the carrier in #1? As I understand it, Kamailio settings are stored in a database; Siremis is a web gui to that database. If that is true, then I don't mind knowing the accompanying 'kamctl' commands to accomplish the above, since I'll see the respective settings in the Siremis web gui anyways. Also don't mind paying for the above tutorial, for which I'll turn into a Googlable blog entry. I need a routine or process for which non-CLI associates can make entries/edits. SIP Trunk with DIDs <--> Siremis on Kamailio <--> SIP Extension Thanks in advance.

1 0

Kamailio for route traffic only
by driver 30 Nov '13

30 Nov '13

Hello, I have such infrastructure: Asterisk1 (provider 1) - IP1 Asterisk2 (provider 2) - IP2 Softphones on client computers with configured two IPs to two Asterisk servers. User has to manually switch between each server. I need to change this that user will have only one account configured (eg. Kamailio) and Kamailio will be used only to forward to Ast1 or Ast2 by previously defined rules. How is it possible? comp1 --------> Kamailio -------> Ast1 comp2 --------> Kamailio -------> Ast1 comp3 --------> Kamailio -------> Ast2 comp4 --------> Kamailio -------> Ast2 Thank you, Marcin

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Some Help with Kamailio v4.0
by Tom Allen 30 Nov '13

30 Nov '13

I am spinning up my first instance of Kamailio and I can't seem to get it to start. I think my issue lies in the kamailio.cfg file, but I don't want to smash that entire file here, it is basically the default. I try to start and get this... root@kh1:/usr/local/etc/kamailio# /etc/init.d/kamailio debug start Starting Kamailio: loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: xxx:5060 udp: 192.168.xxxx:5060 tcp: 127.0.0.1:5060 tcp: xxxx:5060 tcp: 192.168.xxx:5060 Aliases: tcp: kh1:5060 tcp: kh1.xxxx.com:5060 tcp: localhost:5060 udp: kh1:5060 udp: kh1.xxxx.com:5060 udp: localhost:5060 kamailio error, failed to start. I am running a fresh load of Debian .. Linux kh1 3.2.0-4-amd64 #1 SMP Debian 3.2.51-1 x86_64 GNU/Linux I am using a mysql backend. Nothing is in production. What else can I provide you to help me figure this out? Thank you. Respectfully, Tom Allen Virtual Technologies Group

2 3

kamailio 4.0.3 - contact header in ACK
by Adnan 29 Nov '13

29 Nov '13

Hi, We are migrating from our openser 1.4 to kamailio 4.0.3 and have run into a couple of issues. One of them is described below and we need to know if it is a known bug or not. Kamailio receives an INVITE from a gateway and finally responds with 200 OK. Then an ACK is generated by the gateway towards the Kamailio that some how does not get relayed to the next hop. The only special aspect here is that the ACK does not contain a Contact header. Is that something that could confuse kamailio and make it unable to relay? According to SIP RFC, a contact header in ACK is not a must have. Our case is the one where ACK is sent by the UAC after receiving a 200 OK final response. Everything else about the ACK looks OK e.g. ACK has a branch different from the the branch value in the UAC INVITE. Is it a a known bug in 4.0.3? In the configuration, we handle ACK as below: if (is_method("ACK")) { if (t_check_trans()) { t_relay(); exit; } else { exit; } Br Adnan

3 4

Dialog module columns not found in .c files
by Grant Bagdasarian 29 Nov '13

29 Nov '13

Hello, Using 4.0.4: I'm getting the following error (removed the values part of the query): 3(2931) ERROR: db_unixodbc [dbase.c:133]: db_unixodbc_submit_query(): rv=-1. Query= insert into Dialogs (HashEntry,HashID,CallID,FromURI,FromTag,ToURI,ToTag,CallerSock,CalleeSock,StartTime,State,Timeout,CallerCSeq,CalleeCSeq,CallerRouteSet,CalleeRouteSet,CallerContact,CalleeContact,SFlags,ToRouteName,req_uri,xdata,iflags ) 3(2931) ERROR: db_unixodbc [con.c:220]: db_unixodbc_extract_error(): unixodbc:SQLExecDirect=42S22:1:207:[FreeTDS][SQL Server]Invalid column name 'req_uri'. The dialog.c file infact doesn't contain the req_uri_column, x_data_column and the iflags_column in static param_export_t mod_params[],static int mod_init(void). The dlg_db_handler.c files does contain them in the beginning of the file and in static int select_entire_dialog_table(db1_res_t ** res, int fetch_num_rows). I haven't checked the other remaining files, but I think the only problem is the renaming of these columns in the kamailio script. Just wondering if anyone else has noticed this?

1 0

what could be the problem here? 500 Invalid routeset?
by Kelvin Chua 29 Nov '13

29 Nov '13

U 2013/11/28 09:06:09.411796 6X.2X1.4.1X5:5060 -> 2X8.1X1.3X.2X:5060 INVITE sip:+18X88X441X1@2X8.1X1.3X.2X SIP/2.0. Record-Route: <sip:6X.2X1.4.1X5;lr=on;ftag=gK0662d671>. Record-Route: <sip:6X.2X1.4.9X;lr=on;ftag=gK0662d671>. Accept: application/sdp. Allow: INVITE,ACK,CANCEL,BYE. Via: SIP/2.0/UDP 6X.2X1.4.1X5;branch=z9hG4bKcff3.da8920b2.0. Via: SIP/2.0/UDP 6X.2X1.4.9X;branch=z9hG4bKcff3.66c425c4.0. Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK06Bbcfc5fb404048790. From: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. To: <sip:+18X88X441X1@6X.2X1.4.9X>. Call-ID: 1443253238_103264169(a)192.168.47.68. CSeq: 525 INVITE. Max-Forwards: 47. Contact: <sip:+18X16X620X3@192.168.47.68:5060>. Content-Length: 327. Content-Disposition: session; handling=required. Content-Type: application/sdp. Remote-Party-ID: <sip:+18X16X620X3@192.168.47.68:5060 >;privacy=off;screen=no. . v=0. o=Sonus_UAC 30934 15833 IN IP4 192.168.47.68. s=SIP Media Capabilities. c=IN IP4 6X.2X1.4.9X. t=0 0. m=audio 5770 RTP/AVP 0 18 96 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:96 iLBC/8000. a=fmtp:96 mode=30. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=maxptime:30. U 2013/11/28 09:06:09.413388 2X8.1X1.3X.2X:5060 -> 6X.2X1.4.1X5:5060 SIP/2.0 183 Session Progress. Record-Route: <sip:6X.2X1.4.1X5;lr=on;ftag=gK0662d671>. Record-Route: <sip:6X.2X1.4.9X;lr=on;ftag=gK0662d671>. Via: SIP/2.0/UDP 6X.2X1.4.1X5;branch=z9hG4bKcff3.da8920b2.0;rport=5060. Via: SIP/2.0/UDP 6X.2X1.4.9X;branch=z9hG4bKcff3.66c425c4.0. Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK06Bbcfc5fb404048790. From: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. To: <sip:+18X88X441X1@6X.2X1.4.9X >;tag=bbd932f8f9dbf9743f9b7cadcbf622ac.10eb. Call-ID: 1443253238_103264169(a)192.168.47.68. CSeq: 525 INVITE. Server: kamailio (4.0.4 (x86_64/linux)). Content-Length: 0. . U 2013/11/28 09:06:09.413936 2X8.1X1.3X.2X:5060 -> 6X.2X1.4.1X5:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 6X.2X1.4.1X5;branch=z9hG4bKcff3.da8920b2.0;rport=5060. Via: SIP/2.0/UDP 6X.2X1.4.9X;branch=z9hG4bKcff3.66c425c4.0. Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK06Bbcfc5fb404048790. From: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. To: <sip:+18X88X441X1@6X.2X1.4.9X>. Call-ID: 1443253238_103264169(a)192.168.47.68. CSeq: 525 INVITE. Server: kamailio (4.0.4 (x86_64/linux)). Content-Length: 0. . U 2013/11/28 09:06:10.672460 2X8.1X1.3X.2X:5060 -> 6X.2X1.4.1X5:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 6X.2X1.4.1X5;rport=5060;branch=z9hG4bKcff3.da8920b2.0. Via: SIP/2.0/UDP 6X.2X1.4.9X;branch=z9hG4bKcff3.66c425c4.0. Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK06Bbcfc5fb404048790. Record-Route: <sip:2X8.1X1.3X.2X;lr;did=873.7d91;nat=yes>. Record-Route: <sip:6X.2X1.4.1X5;lr;ftag=gK0662d671>. Record-Route: <sip:6X.2X1.4.9X;lr;ftag=gK0662d671>. Call-ID: 1443253238_103264169(a)192.168.47.68. From: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. To: <sip:+18X88X441X1@6X.2X1.4.9X>;tag=3pOAt1HRTOy9Xfi4Rsbbz9gf73MsuXge. CSeq: 525 INVITE. Contact: <sip:1X5.6X.1X6.2X3:9439>. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. . U 2013/11/28 09:06:12.931892 2X8.1X1.3X.2X:5060 -> 6X.2X1.4.1X5:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 6X.2X1.4.1X5;rport=5060;branch=z9hG4bKcff3.da8920b2.0. Via: SIP/2.0/UDP 6X.2X1.4.9X;branch=z9hG4bKcff3.66c425c4.0. Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK06Bbcfc5fb404048790. Record-Route: <sip:2X8.1X1.3X.2X;lr;did=873.7d91;nat=yes>. Record-Route: <sip:6X.2X1.4.1X5;lr;ftag=gK0662d671>. Record-Route: <sip:6X.2X1.4.9X;lr;ftag=gK0662d671>. Call-ID: 1443253238_103264169(a)192.168.47.68. From: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. To: <sip:+18X88X441X1@6X.2X1.4.9X>;tag=3pOAt1HRTOy9Xfi4Rsbbz9gf73MsuXge. CSeq: 525 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Contact: <sip:1X5.6X.1X6.2X3:9439>. Supported: replaces, 100rel, timer, norefersub. Content-Type: application/sdp. Content-Length:238. . v=0. o=- 3594636366 3594636367 IN IP4 2X8.1X1.3X.2X. s=pjmedia. i=(o=IN IP4 10.0.193.243). c=IN IP4 1X5.6X.1X6.2X3. t=0 0. m=audio 10246 RTP/AVP 0. c=IN IP4 2X8.1X1.3X.2X. a=rtcp:10247. a=sendrecv. a=rtpmap:0 PCMU/8000. a=nortpproxy:yes. U 2013/11/28 09:06:12.960342 6X.2X1.4.1X5:5060 -> 2X8.1X1.3X.2X:5060 ACK sip:1X5.6X.1X6.2X3:9439 SIP/2.0. Record-Route: <sip:6X.2X1.4.1X5;lr=on;ftag=gK0662d671>. Record-Route: <sip:6X.2X1.4.9X;lr=on;ftag=gK0662d671>. Via: SIP/2.0/UDP 6X.2X1.4.1X5;branch=z9hG4bKcff3.da8920b2.2. Via: SIP/2.0/UDP 6X.2X1.4.9X;branch=z9hG4bKcff3.66c425c4.2. Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK06Bbd0344f60e48a125. From: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. To: <sip:+18X88X441X1@6X.2X1.4.9X>;tag=3pOAt1HRTOy9Xfi4Rsbbz9gf73MsuXge. Call-ID: 1443253238_103264169(a)192.168.47.68. CSeq: 525 ACK. Max-Forwards: 68. Route: <sip:2X8.1X1.3X.2X:5060;lr;did=873.7d91;nat=yes>. Content-Length: 0. . ------------------------ call goes on a for a while --------------------------- U 2013/11/28 09:06:24.358786 2X8.1X1.3X.2X:5060 -> 6X.2X1.4.1X5:5060 BYE sip:+18X16X620X3@6X.2X1.4.1X5:5060 SIP/2.0. Via: SIP/2.0/UDP 2X8.1X1.3X.2X;branch=z9hG4bKbb1b.267a7326.0. Via: SIP/2.0/UDP 1X5.6X.1X6.2X3:9439;rport=9439;branch=z9hG4bKPj6XR.AckX-G8lRIN4ZuDvjaPB6ZnUpHUa. Max-Forwards: 16. From: <sip:+18X88X441X1@6X.2X1.4.9X>;tag=3pOAt1HRTOy9Xfi4Rsbbz9gf73MsuXge. To: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. Call-ID: 1443253238_103264169(a)192.168.47.68. CSeq: 5685 BYE. Route: <sip:6X.2X1.4.1X5;lr;ftag=gK0662d671>. Route: <sip:6X.2X1.4.9X;lr;ftag=gK0662d671>. User-Agent: CSipSimple_m0-16/r2330. Content-Length: 0. . U 2013/11/28 09:06:24.372741 6X.2X1.4.1X5:5060 -> 2X8.1X1.3X.2X:5060 SIP/2.0 500 Invalid routeset. Via: SIP/2.0/UDP 2X8.1X1.3X.2X;branch=z9hG4bKbb1b.267a7326.0. Via: SIP/2.0/UDP 1X5.6X.1X6.2X3:9439;rport=9439;branch=z9hG4bKPj6XR.AckX-G8lRIN4ZuDvjaPB6ZnUpHUa. From: <sip:+18188244111@6X.2X1.4.9X>;tag=3pOAt1HRTOy9Xfi4Rsbbz9gf73MsuXge. To: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. Call-ID: 1443253238_103264169(a)192.168.47.68. CSeq: 5685 BYE. Server: Bandwidth.com CLEC. Content-Length: 0. . Kelvin Chua

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Safe Maximum for Shared Memory?
by elactrum＠jamailca.com 28 Nov '13

28 Nov '13

Hi, I have Kamailio running on a 32-bit Linux server with 4 GB of memory. Is there any performance impact on Kamailio when increasing shared memory size? Or is the only concern that other processes have sufficient memory left over? The highest recommendation I have found on the mailing list was 1 GB [1], and one user reports using 2 GB [2]. Thanks! -Andrew [1] http://lists.sip-router.org/pipermail/sr-users/2011-October/070458.html [2] http://lists.sip-router.org/pipermail/users/2008-November/020646.html

2 1

rtpproxy / sdp / problem with signalling
by Oliver Roth 28 Nov '13

28 Nov '13

Problem with signalling - RTP gets lost! Rtpproxy not working properly? I am absolutely stuck ... cause this happens in a live environement. I have the following situation A calls B over carrier 1 - number is not valid and I get back error 404 from carrier and now freeswitch should play a message saying: "number not valid". But from carrier 1 I get back an RTP stream that is useless [1] - and if the correct streams opens from freeswitch - this does not get back to A [2]. I tested with rtpproxy on Kamailio - and all the rtp streams arrive at the Kamailio - but they cannot be "connected" correctly. I guess the problem is the 183 I get back from carrier 1 - after whitch rtp is opened. Or there is a wrong sdp singallisation if the "correct" stream arrives [3]. Sorry - I cannot get a solution - but I could provide various tcpdumps and pcaps. A Kamailio Carrier 1 Freeswitch INVITE ---------------------> 100 Your call is important <--------------------- INVITE ------------------------> 100 Trying <------------------------ 183 Session Progress SDP <------------------------ 183 Session Progress SDP <------------------------ RTP <================= RTP <================= RTP =================> RTP =================> [1] 404 not found <--------------------------- ACK ----------------------------> INVITE ---------------------------------------------------------> 100 Trying <-------------------------------------------------------- 200 OK SDP <-------------------------------------------------------- [3] 200 OK SDP <--------------------------- RTP (Announcment - number not valid") [2] <=================================== ACK ----------------------------> ACK ----------------------------------------------------------> INFO ----------------------------> INFO ----------------------------------------------------------> 200 OK <-------------------------------------------------------- 200 OK <--------------------------- BYE ----------------------------> BYE ----------------------------------------------------------> 200 OK <-------------------------------------------------------- 200 OK <---------------------------

2 1

Regarding Kamailio Sip Server
by Naveen Chand 28 Nov '13

28 Nov '13

Dear Sir, well i would like to know,is there any web interface for kamailio to configure complete configuration part,database,modules and upgrades similar like Astrix server ,Kamailio Sip server is Really Rock Solid server,when i first visited your site,i saw the name called Rock Solid ,i thought it just the call like that ,but after testing it ,it satisfied my needs,that is Video calling 2 way ,in excellent manner,well i would like to is Kamailio Sip server is Peer to Peer or not,if it is Peer to Peer,i have tested it ,i removed the connectivity from Server between Clients when call established,it worked for 20 mins and after it discards the call ,so i was bit confusing that kamailio sip server is Peer to Peer or not,pls confirm me,if it is peer to Peer,pls send me the step by step instruction how to configure the Peering module ,But i request you if there is any Web interface Like Astrix it would be great to control all the elements in Kamailio Best Regards T.NAVEEN CHAND

2 1

Books
by Joli Martinez 27 Nov '13

27 Nov '13

Hello, Is there any books you would recommend reading so I can learn more on Kamailio? thanks,

2 2

Log files
by Joli Martinez 27 Nov '13

27 Nov '13

I am running Kamailio in CentOS. I ran tcpdump and noticed that we are getting attacked from IP 188.138.32.72. I have already blocked it on IPtables, but he keeps on attacking the server. If I look at "/var/log/secure" there are no SIP messages. My question is where is the log file for Kamailio and how can I prevent this type of attacks in the future. Thanks,

5 10

voice prompts / early media and kamailio
by Oliver Roth 27 Nov '13

27 Nov '13

Hi all Based on my problem reported with subject "error handling<http://sip-router.1086192.n5.nabble.com/error-handling-tp123240.html>" I have some other questions. I think it is a conceptual question - and I do not see any solution. I would like to handle Kamailio with carrierroute / carrierfailureroute module different errors. Like 404, 403, busy - or whatever sip error occurs. Some of them need to be sent to a freeswitch playing an announcement (like "this number is blocked", "no more credit", ...). So if an error occur (lets say 403) then the call is routed by carrierfailurroute to fresswitch playing message for 403. If I am listening the whole message - I get back error 403 at the end and the call is logged in missed calls as error 403 sent from the freeswitch - everything ok. If I cancel the listening by hanging up - 487 is stored in missed_calls - cause I terminated the call before getting error 403 back from freeswitch. ==> so I loose this important information How can I get back error 403 - play an announcement and make sure, it is logged as 403 in database? Thanks for helping ... Oli

3 2

Re: [SR-Users] Problems with Siremis Register Page
by kamailio＠AaronLux.com 27 Nov '13

27 Nov '13

I am having the exact same problems as issue as this user. There is nothing in the siremis/log/ folder when this happens. I'm running Siremis 4.0.0 with Kamailio 4.0.4. Has this issue been fixed? Hello, On 5/18/13 3:22 AM, Alexander Albert wrote: > Hi there, i hope you can help me, i am having trouble with the > register page of siremis, strange thing is, i can open the register > page, but only after i have logged my self in, if not, it redirects me > directly to the login page.. how can i setup the register page to be > accessible without the need of logging in ? > Also i am getting the info after trying to register, that "Public > registration is not enabled!" > but i have set siremis/modules/ser/config/common.Main.php to > $cfg_siremis_public_registrations = true; > What is the problem, please help. can you look in siremis/log/ folder, there are some files printing debug messages, maybe you get hints about what is wrong. Cheers, Daniel

1 0

adding new column in the CDRS table?
by arun Jayaprakash 26 Nov '13

26 Nov '13

Hello, can someone let me know how I can add a new field in the cdrs table? I would like to set a flag depending if a call is make to another extension number of if the call is made to a PSTN line. Thank you for the help. Arun

2 2

Loosing rtp only in carrierroute failureroute
by Oliver Roth 26 Nov '13

26 Nov '13

Hi all We do have a strange problem with loosing rtp in case of carrierroute - failureroute. If we send traffic directly to the failure gateway, we do have rtp without any problem Situation: Ua --> Freeswitch --> kamailio 3.3 --> gw 1 Error 404 --> failureroute --> media gateway voiceprompt for error 404 ==> this does not work Ua --> Freeswitch --> kamailio 3.3 rewrite to dest for voiceprompt 404 -->media gateway voiceprompt for error 404 ==> this works!! Any idea? For me it seems being a problem with branching or so. Regards, Oli Configs below: Main routing block request_route { --- do all the checks route(CARRIERROUTE); route(RELAY); } route[RELAY] { xlog(, "L_INFO", "RELAY: Outbound sent via $avp(s:trunk_out)"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); xlog(,"L_INFO", "RELAX - Manage branch ...\n"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # carrierroute route[CARRIERROUTE] { xlog(,"L_INFO","Carrierroute module start \n"); $avp(s:tree) = "default"; # lookup from user and from domain cr_user_carrier("$fU", "$avp(s:trunk_in)", "$avp(s:cr_pref_carr)"); if($avp(s:cr_pref_carr) == 0){ # lookup from domain cr_user_carrier("", "$avp(s:trunk_in)", "$avp(s:cr_pref_carr)"); } if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", "call_id", "$avp(s:todirection)" )){ xlog("cr_route 403 no route found $avp(s:cr_pref_carr)"); sl_send_reply("403", "Not allowed"); exit; } $avp(s:trunk_out) = $avp(s:todirection); route(ALTERHEADER); } failure_route[MANAGE_FAILURE] { if (t_grep_status("486")){ xlog("L_INFO", "Status 486 - busy"); t_reply("486", "Busy"); } #revert_uri(); route(NATMANAGE); if (t_is_canceled()) { exit; } xlog("L_INFO", "failure_route $rd $T_reply_code $avp(s:trunk_in), $avp(s:tree), $rU, $rd, $T_reply_code, $avp(s:tree) \n"); if(!cr_next_domain("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$avp(s:trunk_in)", "$T_reply_code", "$avp(s:tree)" )){ xlog("cr_next_domain failed"); exit; } if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", "call_id", "$avp(s:todirection)" )){ xlog("cr_route failed"); exit; } $avp(s:trunk_out) = $avp(s:todirection); if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting setflag(FLT_ACCMISSED); # oro 28.10.13 } route(ALTERHEADER); t_on_failure("MANAGE_FAILURE"); if (!t_relay()) { xlog("failureroute t_relay failed"); exit; } }

1 1

route all calls to a wav file
by Joli Martinez 26 Nov '13

26 Nov '13

Hello, I am getting a lot of hits on port 5060. I would like all registered users to be sent to the correct location (currently working), but all unwanted users allow them to connect and any number they dial it will alway play a wav file. Is this possible if so how would I implement this. Thanks

3 2

Use of t_cancel_callid() to drop an early dialog
by Guillaume Bour 26 Nov '13

26 Nov '13

Hi All We wan't to prevent our users to make more than one call at time, so we choose to disconnect the previous call. When the previous call is established, we use dlg_bye(), and its ok. But when it is in early state, we use t_cancel_callid() to cancel its INVITE transaction. We face 2 issues: 1) we use local-request event route to account calls on timeout. Sometimes this route is called for the cancelled call (after default timeout of 1 hour) 2) t_cancel_callid() cancel previous call, but also _make current dialog disappear_: call is still ongoing and we can answer and talk to each other, but the dialog does not appear in 'kamctl stats dialog' and 'kamctl mi dlg_list' commands Is there a known limitation, or do we misuse t_cancel_callid() ? Regards, Guillaume

3 10

Setting up Conference Calls in Kamailio
by Mark Zeman 26 Nov '13

26 Nov '13

Hello everyone, We're thinking of setting up conference call capability for our Kamailio server, but we can't find any tutorials or documentation. Could someone point us in the right direction or maybe even tell us how to do it? Thanks!

2 1

Re: [SR-Users] Call hold and call conference implementation in kamailio
by Olle E. Johansson 26 Nov '13

26 Nov '13

26 nov 2013 kl. 14:22 skrev "Surendra" <surendra.pullaiah(a)plintron.com>: > Hi, > > Can we implement Call hold and Call conference features in kamailio. If how we can, else what will be the way to implement. > Can anyone help on this. These are media features better handled by end points and media servers. Kamailio by default doesn't handle media, it's a signalling server. Call HOld is between phones, so that's no issue. Call conference can be handled by using Asterisk, FreeSwitch or SylkServer and directing the calls that way in your Kamailio routing script. /O

1 0

Calls via Kamailio - Audio delay
by Mark Zeman 26 Nov '13

26 Nov '13

Hello everybody, having set up our Kamailio server and test env with Jitsi and CSipSimple successfully, we've noticed a delay of ~5 seconds in the audio. We are using IPv6 and encrypting everything (control and data channel) but this should still not lead to such a big delay, as far as we understand. Does anyone know why there would be this delay? Attached is our config, if that is of any help in fixing this delay. Cheers, Mark Zeman

2 1

Call hold and call conference implementation in kamailio
by Surendra 26 Nov '13

26 Nov '13

Hi, Can we implement Call hold and Call conference features in kamailio. If how we can, else what will be the way to implement. Can anyone help on this. Thanks in advance @surendra

1 0

ACL based on User-Agent
by Igor Potjevlesch 26 Nov '13

26 Nov '13

Hello, I was using an old unofficial module called 'check_ua.so' (with an old SER version) to check if an User-Agent is allowed or not. I don't find something similar to do this job in Kamailio. Maybe someone has already something workable with the existing modules? Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

2 1

Problem with loose_route for a BYE
by Igor Potjevlesch 25 Nov '13

25 Nov '13

Hello, I work on interconnecting an old SER version with Kamailio. When a BYE is sent by the old SER, the "if (loose_route)" instruction returns false and then a "404 Not here" is sent. So the BYE is not relay to the UA. If the BYE is sent by Kamailio, no problem. Here are the BYE headers received by Kamailio: Record-Route: sip:OLD_SER_IP_ADDRESS;ftag=e97b7b77c5a3a9cc;lr=on Via: SIP/2.0/UDP OLD_SER_IP_ADDRESS;branch=z9hG4bKc9e5.3cbad207.0 Via: SIP/2.0/UDP UA_IP_ADDRESS_CONNECTED_TO_OLD_SER:5060;rport=5060;branch=z9hG4bK1fbe0a49fe3 44e6af1efd45b74305a9a Route: sip:KAMAILIO_IP_ADDRESS;lr=on I looked at the RR module documentation and it's clearly said that "The loose_route function analyzes the Route: headers in the requests. [.]If a Route: header is found, the function returns 1 and behaves as described in section 16.12 of RFC 3261.". Someone has an idea? Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

3 6

error handling
by Oliver Roth 24 Nov '13

24 Nov '13

Hi all Question about error handling with kamailio. We send call to carrier and get back error 404. In carrierfailureroute we catch up this error and send call to an internal freeswitch that plays a voiceprompt saying: "destination not available" In accounting this calls is collected like a "normal" call - cause the internal freeswitch did the connection. I would like to play the voiceprompt but get the error 404 and see the call in the missed calls acc table. With 486 (busy) it is simple because we do not need an rtp response ... just fast busy. How can we handle this for errors we need to play a voiceprompt? What we do in failure route: failure_route[MANAGE_FAILURE] { sip_trace(); setflag(22); if (t_grep_status("486")){ xlog("L_INFO", "Status 486 - busy"); t_reply("486", "Busy"); exit; } revert_uri(); route(NATMANAGE); if (t_is_canceled()) { exit; } xlog("L_INFO", "failure_route rd: $rd replCode: $T_reply_code trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU: $rU, todirection $avp(s:todirection) fu: $fu\n"); if(!cr_next_domain("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$avp(s:trunk_in)", "$T_reply_code", "$avp(s:tree)" )){ xlog("cr_next_domain failed"); exit; } if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", "call_id", "$avp(s:todirection)" )){ xlog("cr_route failed"); exit; } xlog("L_INFO", "failure_route rd: $rd replCode: $T_reply_code trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU: $rU, todirection $avp(s:todirection) fu: $fu \n"); $avp(s:trunk_out) = $avp(s:todirection); xlog(, "L_INFO", "RELAY - FailureRoute: Outbound sent via $avp(s:trunk_out) rU $rU -------------------------------------------------------- "); if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting setflag(FLT_ACCMISSED); # oro 28.10.13 } route(ALTERHEADER); t_on_failure("MANAGE_FAILURE"); #append_branch(); if (!t_relay()) { xlog("failureroute t_relay failed"); exit; } }

3 5

Re: [SR-Users] SEG FAULT dialog
by Kelvin Chua 24 Nov '13

24 Nov '13

I can reproduce it consistently. I am doing serial forking into a non-existent host, when the caller gets impatient, it cancels the call and send another invite. this happens after that. full backtrace: #0 qm_detach_free (frag=0x7fb2b552e2b0, qm=<optimized out>) at mem/q_malloc.c:269 prev = 0x7fb2b5192058 next = 0x0 #1 qm_malloc (qm=0x7fb2b5192000, size=16) at mem/q_malloc.c:386 f = 0x7fb2b552e2b0 hash = -1256644520 #2 0x00007fb2b44f7e93 in set_dlg_variable_unsafe () from /usr/local/lib64/kamailio/modules/dialog.so No symbol table info available. #3 0x00007fb2b44fa3aa in pv_set_dlg_variable () from /usr/local/lib64/kamailio/modules/dialog.so No symbol table info available. #4 0x000000000047be59 in lval_pvar_assign (rv=0x7fb2bf8348f8, lv=0x7fb2bf834750, msg=0x7fb2bf8c7a00, h=<optimized out>) at lvalue.c:353 r_avp = <optimized out> ret = 1 destroy_pval = 1 pvar = 0x7fb2bf834758 pval = {rs = { s = 0x92750b "+1xx882xx111(a)2xx.xx1.3x.2xSIP/2.0\r\nRecord-Route: <sip:6x.x1.4.195;lr=on;ftag=gK085a1dec>\r\nRecord-Route: <sip:6X.2X1.8.8X;lr=on;ftag=gK085a1dec>\r\nAccept: application/sdp\r\nAllow: INVITE,ACK,CANCEL,BYE"..., len = 12}, ri = 0, flags = 4} avp_val = {n = 0, s = {s = 0x0, len = 0}, re = 0x0} v = <optimized out> #5 lval_assign (h=<optimized out>, msg=0x7fb2bf8c7a00, lv=0x7fb2bf834750, rve=0x7fb2bf8348f0) at lvalue.c:401 rv = 0x7fb2bf8348f8 ret = 0 __FUNCTION__ = "lval_assign" #6 0x000000000041cf14 in do_action (h=0x7fffe1b8f710, a=0x7fb2bf8332b0, msg=0x7fb2bf8c7a00) at action.c:1453 ret = -5 v = <optimized out> dst = {send_sock = 0x7fb2bdb0f080, to = {s = {sa_family = 0, sa_data = "\000\000\000\000\000\000\200-\210\000\000\000\000"}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\200-\210\000\000\000\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {__in6_u = { __u6_addr8 = "\200-\210\000\000\000\000\000\016\000\000\000\000\000\000", __u6_addr16 = {11648, 136, 0, 0, 14, 0, 0, 0}, __u6_addr32 = {8924544, 0, 14, 0}}}, sin6_scope_id = 13}}, id = 0, proto = 58 ':', send_flags = {f = 180 '\264', blst_imask = 138 '\212'}} tmp = <optimized out> new_uri = <optimized out> end = <optimized out> crt = <optimized out> cmd = <optimized out> len = <optimized out> user = <optimized out> uri = {user = {s = 0x7fffe1b8eef0 "/", len = 0}, passwd = { s = 0x3000000010 <Address 0x3000000010 out of bounds>, len = -507973920}, host = { s = 0x7fffe1b8ee20 "", len = 17022112}, port = {s = 0x7fb2bffbd720 "", len = 8208}, params = { s = 0x2010 <Address 0x2010 out of bounds>, len = 17030320}, sip_params = { s = 0x2010 <Address 0x2010 out of bounds>, len = -1077387804}, headers = { s = 0x7fb2bffbdfb0 "0z\001\001", len = -507974144}, port_no = 55072, proto = 49147, type = 32690, flags = 17022128, transport = {s = 0x95 <Address 0x95 out of bounds>, len = -1241200896}, ttl = { s = 0x3734000000a0 <Address 0x3734000000a0 out of bounds>, len = -1081931184}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x63371f "", len = -1}, method = {s = 0x7fb2b519cb30 "\220\240H\265\262\177", len = 4310469}, lr = {s = 0x3000000018 <Address 0x3000000018 out of bounds>, len = -507973440}, r2 = { s = 0x7fffe1b8f000 "\210u\223", len = 8187}, gr = {s = 0x2 <Address 0x2 out of bounds>, len = 1}, transport_val = {s = 0x1 <Address 0x1 out of bounds>, len = 0}, ttl_val = { s = 0xffffffffffffffa0 <Address 0xffffffffffffffa0 out of bounds>, len = 17022112}, user_param_val = { s = 0x7fb2bffbd720 "", len = -507974064}, maddr_val = {s = 0x7fb2bf8c7a00 "\016", len = -1081931184}, method_val = {s = 0x2010 <Address 0x2010 out of bounds>, len = -507971824}, lr_val = {s = 0x0, len = -507971824}, r2_val = {s = 0x7fffe1b8f580 "\001", len = -1}, gr_val = { s = 0x7fb2b519cb30 "\220\240H\265\262\177", len = 4346308}} next_hop = {user = {s = 0x0, len = 1}, passwd = {s = 0x7fffe1b8f710 "\002", len = 1237187532}, host = { s = 0x7fffe1b8f710 "\002", len = -507972224}, port = { s = 0xffffffff <Address 0xffffffff out of bounds>, len = -1256600784}, params = { s = 0x3542f9ae92fdfbcc <Address 0x3542f9ae92fdfbcc out of bounds>, len = -589431860}, sip_params = { s = 0x0, len = 17022128}, headers = {s = 0x15 <Address 0x15 out of bounds>, len = 1}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = 4294967200, transport = {s = 0x7fffe1b8efe8 "l\345N", len = -1076936721}, ttl = {s = 0x1 <Address 0x1 out of bounds>, len = -1114716768}, user_param = { s = 0x1c0000002f <Address 0x1c0000002f out of bounds>, len = 7}, maddr = { s = 0x710000000a <Address 0x710000000a out of bounds>, len = 2}, method = {s = 0x0, len = -18000}, lr = {s = 0x1017a30 "EST", len = 6323844}, r2 = {s = 0x7fb2bd8ed356 "tm [t_lookup.c:716]: ", len = -507972224}, gr = {s = 0x7fffe1b8f710 "\002", len = 5148368}, transport_val = {s = 0x0, len = 0}, ttl_val = {s = 0x7fb2bf8c7a00 "\016", len = -507973348}, user_param_val = { s = 0x103bcb0 "<135>Nov 19 07:28:47 /usr/local/sbin/kamailio[15556]: DEBUG: dialog [dlg_hash.c:602]: dlg_lookup(): dialog id=7094 found on entry 2049\n", len = 0}, maddr_val = { s = 0x103bcb0 "<135>Nov 19 07:28:47 /usr/local/sbin/kamailio[15556]: DEBUG: dialog [dlg_hash.c:602]: dlg_lookup(): dialog id=7094 found on entry 2049\n", len = -1115163744}, method_val = { s = 0x99 <Address 0x99 out of bounds>, len = 1384864127}, lr_val = {s = 0x882d80 "", len = -1081312768}, r2_val = {s = 0x7fb2bdb0f080 "\016", len = 8924544}, gr_val = { s = 0xffffffff <Address 0xffffffff out of bounds>, len = -1081930768}} u = <optimized out> port = <optimized out> dst_host = <optimized out> i = <optimized out> flags = <optimized out> avp = <optimized out> st = {flags = 3213654528, id = 32690, name = {n = -507971824, s = {s = 0x7fffe1b8f710 "\002", len = -507972224}, re = 0x7fffe1b8f710}, avp = 0x4ee56c} sct = <optimized out> sjt = <optimized out> rve = <optimized out> mct = <optimized out> rv = <optimized out> rv1 = <optimized out> c1 = {cache_type = 3213020352, val_type = 32690, c = {avp_val = {n = 0, s = { s = 0x7fb200000000 <Address 0x7fb200000000 out of bounds>, len = -1081312768}, re = 0x7fb200000000}, pval = {rs = {s = 0x7fb200000000 <Address 0x7fb200000000 out of bounds>, len = -1081312768}, ri = -1124131338, flags = 32690}}, i2s = "\377\377\377\377\060", '\000' <repeats 16 times>} s = {s = 0x7fffe1b8f0d8 "\260\062\203\277\262\177", len = -507973632} srevp = {0x937588, 0x7fb2bfc4f1fc} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, { type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = { s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action" #7 0x000000000041c5c5 in run_actions (h=0x7fffe1b8f710, a=0x7fb2bf8313a0, msg=0x7fb2bf8c7a00) at action.c:1573 t = <optimized out> ret = <optimized out> mod = <optimized out> ms = 0 __FUNCTION__ = "run_actions" #8 0x000000000041d92e in do_action (h=0x7fffe1b8f710, a=0x7fb2bf8396d0, msg=0x7fb2bf8c7a00) at action.c:1065 ret = 1 v = 1 dst = {send_sock = 0x0, to = {s = {sa_family = 0, sa_data = "\000\000\001\000\000\000\000\000\000\000\000\000\000"}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 1}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = { sin6_family = 0, sin6_port = 0, sin6_flowinfo = 1, sin6_addr = {__in6_u = { __u6_addr8 = "\000\000\000\000\000\000\000\000\001\000\000\000\000\000\000", __u6_addr16 = {0, 0, 0, 0, 1, 0, 0, 0}, __u6_addr32 = {0, 0, 1, 0}}}, sin6_scope_id = 0}}, id = 0, proto = -80 '\260', send_flags = {f = 188 '\274', blst_imask = 3 '\003'}} tmp = <optimized out> new_uri = <optimized out> end = <optimized out> crt = <optimized out> cmd = <optimized out> len = <optimized out> user = <optimized out> uri = {user = {s = 0x3635353531000002 <Address 0x3635353531000002 out of bounds>, len = 1}, passwd = { s = 0x0, len = 17022128}, host = {s = 0x7fffe1b8f460 "/", len = 0}, port = { s = 0x3000000010 <Address 0x3000000010 out of bounds>, len = -507972528}, params = { s = 0x7fffe1b8f390 "\260\337\373\277\262\177", len = -1076936721}, sip_params = { s = 0xa2 <Address 0xa2 out of bounds>, len = -1114690816}, headers = { s = 0x1c0000002f <Address 0x1c0000002f out of bounds>, len = 7}, port_no = 10, proto = 0, type = 113, flags = URI_SIP_USER_PHONE, transport = {s = 0x0, len = -18000}, ttl = {s = 0x1017a30 "EST", len = -1076935806}, user_param = {s = 0x7fb2b5531d90 "P\213\065\265\262\177", len = 48}, maddr = { s = 0x3734e1b8f3b8 <Address 0x3734e1b8f3b8 out of bounds>, len = -507972896}, method = { s = 0x3531320000000002 <Address 0x3531320000000002 out of bounds>, len = -1076363582}, lr = { s = 0x103bcbc "07:28:47 /usr/local/sbin/kamailio[15556]: DEBUG: dialog [dlg_hash.c:602]: dlg_lookup(): dialog id=7094 found on entry 2049\n", len = 17022140}, r2 = {s = 0x7fb2bffbb3c0 "", len = 7}, gr = { s = 0x3000000018 <Address 0x3000000018 out of bounds>, len = -507972048}, transport_val = { s = 0x7fffe1b8f570 "\001", len = 8187}, ttl_val = {s = 0x2 <Address 0x2 out of bounds>, len = 1}, user_param_val = {s = 0x1 <Address 0x1 out of bounds>, len = 0}, maddr_val = { s = 0xffffffffffffffa0 <Address 0xffffffffffffffa0 out of bounds>, len = 17022112}, method_val = { s = 0x7fb2bffbd720 "", len = 8208}, lr_val = {s = 0x2010 <Address 0x2010 out of bounds>, len = 17030320}, r2_val = {s = 0x2010 <Address 0x2010 out of bounds>, len = -1077387804}, gr_val = { s = 0x7fb2bffbdfb0 "0z\001\001", len = -1077164088}} next_hop = {user = {s = 0x7fb2bffbd720 "", len = 8180}, passwd = { s = 0x3635353531000002 <Address 0x3635353531000002 out of bounds>, len = 17022112}, host = {s = 0x0, len = -1077381378}, port = {s = 0x7fffe1b8f5f0 "", len = 0}, params = { s = 0x3000000010 <Address 0x3000000010 out of bounds>, len = -507972128}, sip_params = { s = 0x7fffe1b8f520 "\002", len = -1077164088}, headers = {s = 0x0, len = 8180}, port_no = 2, proto = 12544, type = 909456693, flags = 17022128, transport = {s = 0x0, len = 1}, ttl = { s = 0x7fffe1b8f650 "0\313\031\265\262\177", len = 0}, user_param = { s = 0x3000000010 <Address 0x3000000010 out of bounds>, len = -507972032}, maddr = { s = 0x7fffe1b8f580 "\001", len = 6526864}, method = { s = 0x37340000002f <Address 0x37340000002f out of bounds>, len = 7}, lr = { s = 0x320000710000000a <Address 0x320000710000000a out of bounds>, len = -1076363582}, r2 = { s = 0x103bcbc "07:28:47 /usr/local/sbin/kamailio[15556]: DEBUG: dialog [dlg_hash.c:602]: dlg_lookup(): dialog id=7094 found on entry 2049\n", len = 17022140}, gr = {s = 0x7fb2bffbb3c0 "", len = 7}, transport_val = { s = 0x3000000018 <Address 0x3000000018 out of bounds>, len = -507971648}, ttl_val = { s = 0x7fffe1b8f700 "0\313\031\265\262\177", len = 8187}, user_param_val = { s = 0x373400000002 <Address 0x373400000002 out of bounds>, len = 1}, maddr_val = { s = 0x1 <Address 0x1 out of bounds>, len = -1076363582}, method_val = { s = 0x103bcbc "07:28:47 /usr/local/sbin/kamailio[15556]: DEBUG: dialog [dlg_hash.c:602]: dlg_lookup(): dialog id=7094 found on entry 2049\n", len = 17022140}, lr_val = {s = 0x7fb2bffbb3c0 "", len = 7}, r2_val = { s = 0x3000000018 <Address 0x3000000018 out of bounds>, len = -507971552}, gr_val = { s = 0x7fffe1b8f760 "", len = 8187}} u = <optimized out> port = <optimized out> dst_host = <optimized out> i = <optimized out> flags = <optimized out> avp = <optimized out> st = {flags = 2, id = 0, name = {n = 1, s = {s = 0x1 <Address 0x1 out of bounds>, len = 1}, re = 0x1}, avp = 0x0} sct = <optimized out> sjt = <optimized out> rve = <optimized out> mct = <optimized out> rv = <optimized out> rv1 = <optimized out> c1 = {cache_type = 137, val_type = RV_NONE, c = {avp_val = {n = 17022112, s = {s = 0x103bca0 "`\002", len = 160}, re = 0x103bca0}, pval = {rs = {s = 0x103bca0 "`\002", len = 160}, ri = -1077381378, flags = 32690}}, i2s = " \327\373\277\262\177\000\000\000\000\000\000\001\000\000\000\340\366\270\341\377\177"} s = {s = 0xffffffffffffffa0 <Address 0xffffffffffffffa0 out of bounds>, len = 17022112} srevp = {0x7fb2bffbd720, 0x2010} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, { type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = { s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action" #9 0x000000000041c5c5 in run_actions (h=0x7fffe1b8f710, a=0x7fb2bf82d4c0, msg=0x7fb2bf8c7a00) at action.c:1573 t = <optimized out> ret = <optimized out> mod = <optimized out> ms = 0#0 qm_detach_free (frag=0x7fb2b552e2b0, qm=<optimized out>) at mem/q_malloc.c:269 prev = 0x7fb2b5192058 next = 0x0 ... cut .... Kelvin Chua On Tue, Nov 19, 2013 at 9:40 PM, Daniel-Constantin Mierla <miconda(a)gmail.com > wrote: > Hello, > > can you get the output for 'bt full'? Probably you have to install the > debug symbols (kamailio-dbg package) for getting something useful. > > Another question, can you reproduce it? Or it happens sporadically? > > Cheers, > Daniel > > > > > On 11/19/13 1:11 PM, Kelvin Chua wrote: > > kamailio 4.0.4 > has anybody experience this using dialog module? > > backtrace: > #0 0x000000000057af45 in qm_malloc () > #1 0x00007f268ac0fe93 in set_dlg_variable_unsafe () from > /usr/local/lib64/kamailio/modules/dialog.so > #2 0x00007f268ac123aa in pv_set_dlg_variable () from > /usr/local/lib64/kamailio/modules/dialog.so > #3 0x000000000047be59 in lval_assign () > #4 0x000000000041cf14 in do_action () > #5 0x000000000041c5c5 in run_actions () > #6 0x000000000041d92e in do_action () > #7 0x000000000041c5c5 in run_actions () > #8 0x0000000000425250 in run_top_route () > #9 0x00000000004bfcce in receive_msg () > #10 0x000000000056a2df in udp_rcv_loop () > #11 0x0000000000481b43 in main_loop () > #12 0x000000000041c081 in main () > > logfile: > DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: > DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <ACK> > DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: > <sip:+1XX882XX111@2XX.XX1.X9.2X> > DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> > DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param > type 232, <branch> = <z9hG4bKe97b.5a4397f3.0>; state=16 > DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header > reached, state=5 > DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: > Via found, flags=2 > DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: > this is the first via > DEBUG: <core> [receive.c:151]: receive_msg(): After parse_msg... > DEBUG: <core> [receive.c:192]: receive_msg(): preparing to run routing > scripts... > DEBUG: <core> [parser/parse_to.c:176]: parse_to_param(): DEBUG: add_param: > tag=bbd932f8f9dbf9743f9b7cadcbf622ac.0dc7 > DEBUG: <core> [parser/parse_to.c:799]: parse_to(): end of header reached, > state=29 > DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: > get_hdr_field: <To> [74]; uri=[sip:+1XX882XX111@6X.X31.X.8X] > DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [ > <sip:+1XX882XX111@6X.X31.X.8X>] > DEBUG: sl [sl_funcs.c:415]: sl_filter_ACK(): SL local ACK found -> > dropping it! > DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: > destroying list (nil) > last message repeated 5 times > DEBUG: <core> [xavp.c:447]: xavp_destroy_list(): destroying xavp list (nil) > DEBUG: <core> [receive.c:295]: receive_msg(): receive_msg: cleaning up > segfault at 0 ip 000000000057af45 sp 00007fff30e44ac0 error 4 in > kamailio[400000+27c000] > > Kelvin Chua > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio Advanced Trainings - Berlin, Nov 25-28 > - more details about Kamailio trainings at http://www.asipto.com - > > > _______________________________________________ > 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 > >

2 4

kamailio 4.0.4 package for ubuntu
by Freddy Andersen 23 Nov '13

23 Nov '13

I’m currently running Kamailio 4.0.3 in production and think I’m hitting bugs that looks like they are fixed in 4.0.4 but deb.kamailio.org does not have 4.0.4 yet? Any timeline when this will get released? -- - Freddy

1 0

ubuntu package for kamailio 4.0.4
by Freddy Andersen 23 Nov '13

23 Nov '13

I’m currently running Kamailio 4.0.3 in production and think I’m hitting bugs that looks like they are fixed in 4.0.4 but deb.kamailio.org does not have 4.0.4 yet? Any timeline when this will get released? -- - Freddy

1 0

Kamailio returns "405 Method Not Allowed" for SIP MESSAGE request
by Karthikeyan R 22 Nov '13

22 Nov '13

Is Kamailio supports SIP MESSAGE request? Please let me know how to configure. Regards, Karthikeyan R DISCLAIMER: This email message and all attachments are confidential and may contain information that is Privileged, Confidential or exempt from disclosure under applicable law. If you are not the intended recipient, you are notified that any dissemination, distribution or copying of this email is strictly prohibited. If you have received this email in error, please notify us immediately by return email to mailadmin(a)spanservices.com and destroy the original message. Opinions, conclusions and other information in this message that do not relate to the official of SPAN, shall be understood to be nether given nor endorsed by SPAN.

3 2

siptrace & sipcapture - write to multiple tables
by Grant Bagdasarian 22 Nov '13

22 Nov '13

Hello, Is it possible to have the sipcapture module write the duplicated messages to multiple tables? If not, is it possible to have the siptrace module duplicate the message to multiple Sip Capture instances? Regards, Grant

3 3

add carrier
by Joli Martinez 21 Nov '13

21 Nov '13

Hello, I have Kamailio installed as an SBC with Freeswitch. I would like to add a SIP trunk to Kamailio. Is there any documentation on how to do this? Also, is it recommended to add trunks to Kamailio vs Freeswitch if so why? Thanks

3 5

REGISTER request's "Via" address used to send INVITE
by Karthikeyan R 21 Nov '13

21 Nov '13

Hi All, I'm trying to configure the Kamailio Server (V 4.0.3) as Registrar with OverSIP server (V 1.4.1) as a Proxy for my WebRTC client; While calling UserB from UserA, the Kamailio is trying to find the UserB using address from Via address in UserB's REGISTER request; I'm also using GRUU and enabled the same in configuration file. Can someone help me what's wrong in my configuration / is Kamailio requires any additional settings to resolve this issue? Attached Files.zip contains the following files: OverSip-Kamailio.pcapng - Contains the packet capture between the OverSIP and Kamailio Servers. Kamailio-Logs.txt - Shows the error message that Kamailio is trying to resolve address and it's failed. User-records.txt - Output of "kamctl ul show". Kamailio.cfg - Configuration file for Kamailio. Note: I have configured MySQL as DBENGINE, but haven't created any DB tables. Thanks in Advance, Karthikeyan R DISCLAIMER: This email message and all attachments are confidential and may contain information that is Privileged, Confidential or exempt from disclosure under applicable law. If you are not the intended recipient, you are notified that any dissemination, distribution or copying of this email is strictly prohibited. If you have received this email in error, please notify us immediately by return email to mailadmin(a)spanservices.com and destroy the original message. Opinions, conclusions and other information in this message that do not relate to the official of SPAN, shall be understood to be nether given nor endorsed by SPAN.

1 0

Problem with 4.0 nightlies and IMS - sem_post undefined
by Barry Flanagan 21 Nov '13

21 Nov '13

Hi I am trying to run the Ubuntu nightlies from http://deb.kamailio.org/kamailiodev-nightly and loading the p-cscf config files. Things were working fine while I was on the 20130110 nightlies, but when I upgraded to 20130122 I get the following errors. Any have an idea why sem_post is undefined and how to fix this? 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/lib64/kamailio/modules/ims_registrar_pcscf.so>: /usr/lib64/kamailio/modules/ims_registrar_pcscf.so: undefined symbol: sem_post 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/lib64/kamailio/modules/cdp.so>: /usr/lib64/kamailio/modules/cdp.so: undefined symbol: sem_post 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/lib64/kamailio/modules/ims_qos.so>: /usr/lib64/kamailio/modules/ims_qos.so: undefined symbol: sem_post Thanks -Barry Flanagan

4 10

append_hf to CANCEL
by Grant Bagdasarian 21 Nov '13

21 Nov '13

Hello, I'm executing the following code for both the BYE and CANCEL in request_route: if (is_method("BYE|CANCEL")) { #finished $var(finished) = $TS; append_hf("X-Finished: $var(finished)\r\n"); } For some reason, the X-Finished header is appended to the BYE, but not the CANCEL. The CANCEL is forwarded further. I have the setflag(22); set, so all outbound messages are sent to the Capture server. Does append_hf not work for CANCEL requests?

4 9

Transaction Matching Not Working
by Samuel Ware 21 Nov '13

21 Nov '13

I am currently working on building a type of outbound proxy. I am having an issue with t_check_trans() not finding the associated transaction. I thought maybe this was because I was manipulating the RURI and other headers so I turned on the called_matching parameter and turned off ruri_matching and via1_matching. I get the same result as before making that change of the t_check_trans() not returning a positive value. # Check source address/port of request if ($si != "xxx.xxx.xxx.xxx") { # This means that the server communicating is not an internal # sip signaling server. We must determine in the message is # part of an existing Call Dialog. If not, we drop the message. # If so, relay the message on presumably to the appropriate internal # server. if ( t_check_trans() ) { # Confirmed part of an existing dialog, so we relay xlog("L_INFO","External Server ($si:$sp) - Sent Relay Message (Call ID: $ci METHOD: $rm)"); route(RELAY); exit; } else { # Message is not part of an existing dialog, so drop it with # no response xlog("L_WARN","External Server ($si:$sp) - Sent Non-Authorized Message (Call ID: $ci METHOD: $rm)"); exit; } }; SIP Capture: U 2013/11/20 17:08:16.87172 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 INVITE sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:mod_sofia@xxx.xxx.xxx.xxx:5060> User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130723T192601Z~1dbcfbfa02 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 205 X-VSIP-CUST-IP: 162.243.64.6 X-AUTH-IP: 162.243.64.6 X-VSIP-OUTBOUND: DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060 X-FS-Support: update_display,send_info P-Asserted-Identity: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx> v=0 o=FreeSWITCH 390420642 390420643 IN IP4 xxx.xxx.xxx.xxx s=FreeSWITCH c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 19532 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 U 2013/11/20 17:08:16.87340 yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport=5060;branch=z9hG4bKceQ2e5g58evvD From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Server: kamailio (4.0.4 (x86_64/linux)) Content-Length: 0 U 2013/11/20 17:08:16.87387 yyy.yyy.yyy.yyy:5060 -> zzz.zzz.zzz.zzz:5060 INVITE sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060 SIP/2.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;branch=z9hG4bK9dcc.06fa6927.0 ;rport=5060Max-Forwards: 14 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:mod_sofia@xxx.xxx.xxx.xxx:5060> User-Agent: VSIP-OP v0.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 205 X-AUTH-IP: 162.243.64.6 X-FS-Support: update_display,send_info P-Asserted-Identity: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx> v=0 o=FreeSWITCH 390420642 390420643 IN IP4 xxx.xxx.xxx.xxx s=FreeSWITCH c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 19532 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 U 2013/11/20 17:08:16.117314 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Server: MERA MVTS3G v.4.3.0-46 Content-Length: 0 U 2013/11/20 17:08:18.383674 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 183 Progress Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Server: MERA MVTS3G v.4.3.0-46 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:18.570976 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:19.71773 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:20.72864 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:22.74578 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:26.73588 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:30.72173 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:32.187227 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:32.686940 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:33.688358 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:34.71527 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:35.688296 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:38.70512 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:39.691038 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:42.74073 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:43.689345 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:46.77664 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:47.689095 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:50.76714 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:51.688846 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:55.689470 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:55.903509 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:08:56.904308 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:08:58.903685 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:08:59.689703 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:09:02.903777 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:09:03.690873 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:09:06.902640 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:09:10.904529 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:09:14.903813 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:09:18.906135 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:09:22.902976 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:09:26.905600 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0

3 2

WebRTC - XOR-MAPPED-ADDRESS and bridge mode
by Kelvin Chua 21 Nov '13

21 Nov '13

here is the scenario: kamailio 4.0.4 running in bridge mode eth0 : 10.17.0.202 eth1 : 100.200.30.40 a call comes in from ice enabled webrtc client (chrome) from public internet, kamailio processed the call, no audio. if i replace kamailio with asterisk, audio ok. upon investigating using wireshark, i found out that chrome sends a "Binding Request" for the IP/port in ICE candidate. Kamailio/rtpproxy sends back Binding Success XOR-MAPPED-ADDRESS but the IP is the internal one 10.17.0.202:port this is not right, the response should be the remote IP/port, not the local IP/port Kelvin Chua

1 1

Does $hdr() accept PVs?
by Alex Balashov 20 Nov '13

20 Nov '13

That is to say, can I do? $var(x) = 'P-Asserted-Identity'; xlog("L_INFO", "PAI: $hdr(var(x))\n"); What about remove_hf("$var(x)"); -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 1

EOF Error in pass_fd
by Ricardo Martinez 20 Nov '13

20 Nov '13

Hello List. The last weekend our kamailio process crashed with this error : Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10661]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 29 Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: ALERT: <core> [main.c:751]: child process 10601 exited by a signal 11 Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: ALERT: <core> [main.c:754]: core was generated Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD Can someone tell me what seems to be the problem here? We’re using : version: kamailio 3.2.2 (x86_64/linux) 98ba92-dirty flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 98ba92 -dirty compiled on 12:45:36 Feb 2 2012 with gcc 4.4.6 And this is the bt full from gdb Core was generated by `/usr/local/sbin/kamailio -m 1024'. Program terminated with signal 11, Segmentation fault. #0 mi_profile_list (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_profile.c:789 789 if ( mi_print_dlg( rpl, ph->dlg, 0)!=0 ) Missing separate debuginfos, use: debuginfo-install filesystem-2.4.30-2.1.el6.x86_64 glibc-2.12-1.25.el6_1.3.x86_64 keyutils-libs-1.4-1.el6.x86_64 krb5-libs-1.9-9.el6_1.2.x86_64 libcom_err-1.41.12-7.el6.x86_64 libselinux-2.0.94-5.el6.x86_64 mysql-libs-5.1.52-1.el6_0.1.x86_64 nss-softokn-freebl-3.12.7-1.1.el6.x86_64 openssl-1.0.0-20.el6_2.1.x86_64 pcre-7.8-3.1.el6.x86_64 zlib-1.2.3-26.el6.x86_64 (gdb) bt full #0 mi_profile_list (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_profile.c:789 node = <value optimized out> rpl_tree = 0x20a6790 rpl = 0x20a67b0 profile = 0x7fe99a2dda00 ph = 0x0 profile_name = <value optimized out> value = <value optimized out> i = <value optimized out> #1 0x00007fe9dce287b1 in run_mi_cmd (fifo_stream=<value optimized out>) at ../../lib/kmi/mi.h:77 No locals. #2 mi_fifo_server (fifo_stream=<value optimized out>) at fifo_fnc.c:509 mi_cmd = 0x208f8f0 mi_rpl = <value optimized out> hdl = 0x0 line_len = 41 file_sep = 0x2039017 "" command = 0x7fe9dfdbdef9 "profile_list_dlgs" file = 0x7fe9dfd98d78 "/tmp/openser_receiver_27809" f = 0x2039018 reply_stream = 0x20b0980 #3 0x00007fe9dce2a640 in fifo_process (rank=<value optimized out>) at mi_fifo.c:247 fifo_stream = 0x209a0b0 #4 0x00007fe9dce2a9de in mi_child_init (rank=0) at mi_fifo.c:211 pid = <value optimized out> #5 0x00000000004dba41 in init_mod_child (m=0x7fe9dfc6ba10, rank=0) at sr_module.c:886 No locals. #6 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6bd08, rank=0) at sr_module.c:883 No locals. #7 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6c248, rank=0) at sr_module.c:883 No locals. #8 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6c748, rank=0) at sr_module.c:883 No locals. #9 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6d288, rank=0) at sr_module.c:883 No locals. #10 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6d928, rank=0) at sr_module.c:883 No locals. #11 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6dee0, rank=0) at sr_module.c:883 No locals. #12 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6e1f0, rank=0) at sr_module.c:883 No locals. #13 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6e5d0, rank=0) at sr_module.c:883 No locals. #14 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6e830, rank=0) at sr_module.c:883 No locals. #15 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6ebd0, rank=0) at sr_module.c:883 No locals. #16 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6f220, rank=0) at sr_module.c:883 No locals. #17 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6f648, rank=0) at sr_module.c:883 No locals. #18 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6fa60, rank=0) at sr_module.c:883 No locals. #19 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6fe40, rank=0) at sr_module.c:883 No locals. #20 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc702c0, rank=0) at sr_module.c:883 No locals. #21 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc70ae8, rank=0) at sr_module.c:883 No locals. ---Type <return> to continue, or q <return> to quit--- #22 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc70e18, rank=0) at sr_module.c:883 No locals. #23 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc71100, rank=0) at sr_module.c:883 No locals. #24 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc716c0, rank=0) at sr_module.c:883 No locals. #25 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc71df0, rank=0) at sr_module.c:883 No locals. #26 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc723c0, rank=0) at sr_module.c:883 No locals. #27 0x000000000046077d in main_loop () at main.c:1661 i = <value optimized out> pid = <value optimized out> si = 0x0 si_desc = "udp receiver child=15 sock=64.76.154.35:5060", '\000' <repeats 20 times>, "\003", '\000' <repeats 15 times>, "\030\325\a\232\351\177\000\000\001\000\000\000\002", '\000' <repeats 11 times>, "\001\000\000\000\000\000\000\000\003\000\000\000\000\000\000\000)\000\000\000\000\000\000" #28 0x0000000000461f13 in main (argc=<value optimized out>, argv=0x7fff2f2b2848) at main.c:2475 cfg_stream = <value optimized out> c = <value optimized out> r = <value optimized out> tmp = 0x7fff2f2b3f76 "" tmp_len = 0 options = 0x5a3178 ":f:cm:M:dVhEb:l:L:n:vrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 1850310455 rfd = <value optimized out> debug_save = <value optimized out> debug_flag = <value optimized out> dont_fork_cnt = <value optimized out> n_lst = <value optimized out> p = <value optimized out> (gdb) Regards, Ricardo.-

4 7

change callerid, kamailio
by vladimir tarasov 20 Nov '13

20 Nov '13

Substituted by callerid mobile number - the higher the operator beats, and other provider passes. How to write so that the directions to change the phone number that is displayed with customers from 7452393 to 3493357505

1 0

change callerid
by vladimir tarasov 20 Nov '13

20 Nov '13

How do I keep calling with an asterisk on the kamailio, changed callerid on kamailio and then called on the gw?

1 0

remove a line from SDP
by Laurent Schweizer 20 Nov '13

20 Nov '13

Dear all, I try to remove a line from an SDP with: subst_body('/^a=fmtp:8 vad=no(.*)$//g'); unfortunately the "0a" of the end of line ( 0d0a ) is not removed, so the "$" remove only the 0d any idea how to remove the last "0a" Best regards Laurent

2 2

TLS questions
by Fabian Borot 20 Nov '13

20 Nov '13

Hi, please help me out with these questions: Is it possible to configure kamailio to use different certificates, one per peer? Do we just add them to the same "calist.pem" file? Is is possible to configure kamailio to accept a certificate from another device? I mean instead of Kamailio creating the certificate/key and give it to customer/provider. If that is the case, do we also add it to the "calist.pem" file as well? If all the above is true, what happens when the certificates have different expiration dates? thank you very much

3 7

kamailio and db_cassandra
by jay binks 20 Nov '13

20 Nov '13

Hi all, can anyone comment on the lack of a debian package for the db_cassandra module ?? I understand there may be reasons to not distribute the binary, but I was really hoping to find a control file so I can build my own package. any hints / pointers on the best way to achive this and the reasons behind it would be great. -- Sincerely Jay

3 7

need help regarding with kamailio3.3
by Manav Kumar 20 Nov '13

20 Nov '13

Hi , I need some help regarding kamailio 3.3 presence issue. Below I mentioned the scenario which I'm doing with kamailio3.3. I simply install kamailio3.3 with configuration file(as per http://kb.asipto.com/kamailio:presence:k31-made-simple site) which I have attached with this mail. I have started my kamailio on tcp as well as UDP because I want getting presence with xcap configuration and after configured jitsi when I open the jitsi client it makes the tcp connection with kamailio server and fetch users list from server by following request ( http://ip:port/ xcap-root/resource-lists/users/sip:manav@172.21.183.138/index) and jitsi client gives pop up message i.e http://ip:port/xcap-root/xcap-caps/global/index resource cannot be read . and when I see in wireshark traces it closes the tcp connection . Can you help me this issue as I need some kind of resolution on high priority ? Thanx in advance for your support. Regards, Manav =============================================================================== Please refer to http://www.aricent.com/legal/email_disclaimer.html for important disclosures regarding this electronic communication. ===============================================================================

1 0

record_route_preset() Question
by kamailio＠AaronLux.com 20 Nov '13

20 Nov '13

Why did 'transport=tls' 'r2=on' 'lr=on' and 'nat=yes' all disappear after just adding 'record_route_preset("myurl.com");' (I had to edit my routing logic to fix a problem with kamailio sending record-route to WAN clients telling them to respond on my private NAT IP address) # I'm concerned my edits to the routing logic is incorrect and would like a second opinion. (I added record_route_preset on line 41 and commented out line 42): http://pastebin.com/3azFStmE # Here is what record-route looked like Before adding record_route_preset (3 lines): Record-Route: <sip:66.111.222.111:5061;transport=tls;r2=on;lr=on;nat=yes> Record-Route: <sip:66.111.222.111;r2=on;lr=on;nat=yes> Record-Route: <sip:myurl.com;lr=on> # And now here is what they look like After adding record_route_preset: (1 line) Record-Route: <sip:myurl.com;lr> MISC INFO: # kamailio -v version: kamailio 4.0.4 (i386/linux) cabe58 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: cabe58 compiled on 01:07:11 Nov 2 2013 with gcc 4.4.7

1 0

SEG FAULT dialog
by Kelvin Chua 19 Nov '13

19 Nov '13

kamailio 4.0.4 has anybody experience this using dialog module? backtrace: #0 0x000000000057af45 in qm_malloc () #1 0x00007f268ac0fe93 in set_dlg_variable_unsafe () from /usr/local/lib64/kamailio/modules/dialog.so #2 0x00007f268ac123aa in pv_set_dlg_variable () from /usr/local/lib64/kamailio/modules/dialog.so #3 0x000000000047be59 in lval_assign () #4 0x000000000041cf14 in do_action () #5 0x000000000041c5c5 in run_actions () #6 0x000000000041d92e in do_action () #7 0x000000000041c5c5 in run_actions () #8 0x0000000000425250 in run_top_route () #9 0x00000000004bfcce in receive_msg () #10 0x000000000056a2df in udp_rcv_loop () #11 0x0000000000481b43 in main_loop () #12 0x000000000041c081 in main () logfile: DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <ACK> DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:+1XX882XX111@2XX.XX1.X9.2X> DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKe97b.5a4397f3.0>; state=16 DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via DEBUG: <core> [receive.c:151]: receive_msg(): After parse_msg... DEBUG: <core> [receive.c:192]: receive_msg(): preparing to run routing scripts... DEBUG: <core> [parser/parse_to.c:176]: parse_to_param(): DEBUG: add_param: tag=bbd932f8f9dbf9743f9b7cadcbf622ac.0dc7 DEBUG: <core> [parser/parse_to.c:799]: parse_to(): end of header reached, state=29 DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [74]; uri=[sip:+1XX882XX111@6X.X31.X.8X] DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:+1XX882XX111@6X.X31.X.8X>] DEBUG: sl [sl_funcs.c:415]: sl_filter_ACK(): SL local ACK found -> dropping it! DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) last message repeated 5 times DEBUG: <core> [xavp.c:447]: xavp_destroy_list(): destroying xavp list (nil) DEBUG: <core> [receive.c:295]: receive_msg(): receive_msg: cleaning up segfault at 0 ip 000000000057af45 sp 00007fff30e44ac0 error 4 in kamailio[400000+27c000] Kelvin Chua

2 1

kamdbctl create ?
by Thomas Secula 18 Nov '13

18 Nov '13

I seem to be in a catch 22, sure it's something I've done wrong. I set dbhost in kamctlrc to be the ip of my remote mysql server I run kamdbctl create and I get root@pcscf:/etc/kamailio# kamdbctl create MySQL password for root: database engine 'mysql' loaded INFO: test server charset INFO: creating database kamailio ... INFO: granting privileges to database kamailio ... ERROR 1044 (42000) at line 1: Access denied for user 'root'@'172.16.101.27' to database 'kamailio' ERROR: granting privileges to database kamailio failed! root@pcscf:/etc/kamailio# Before I run the create from mysql: mysql> select user,host from mysql.user; +------+---------------+ | user | host | +------+---------------+ | root | % | | root | 172.16.101.27 | | root | 172.16.101.28 | | root | 172.16.101.29 | | root | localhost | +------+---------------+ 5 rows in set (0.00 sec) mysql> show grants for 'root'@'%'; +--------------------------------------------------------------------------- -----------------------------------+ | Grants for root@% | +--------------------------------------------------------------------------- -----------------------------------+ | GRANT ALL PRIVILEGES ON *.* TO 'root'@'%' IDENTIFIED BY PASSWORD '*FE04940F3F9B339AB9361B2D01AD3D940B215B52' | +--------------------------------------------------------------------------- -----------------------------------+ 1 row in set (0.00 sec) If I rerun it the create scripts says: root@pcscf:/etc/kamailio# kamdbctl create MySQL password for root: database engine 'mysql' loaded INFO: test server charset INFO: creating database kamailio ... ERROR 1007 (HY000) at line 1: Can't create database 'kamailio'; database exists ERROR: Creating database kamailio failed! root@pcscf:/etc/kamailio# any ideas?

2 3

kamailio tls.reload core dump
by Ding Ma 18 Nov '13

18 Nov '13

Hi, all This is related to the previous tls.reload not safe email chain. Now we have a detailed gdb output that shows the stack trace of the core dump. Please take a look. This looks like a bug. Please let me know if you have any insights on how to fix this. Thanks, Ding

2 6

kamailio 4.0 crash at urecord wb_timer
by jh＠tutpro.com 18 Nov '13

18 Nov '13

this kind of crash has occurred a few times now with kamailio 4.0: Program terminated with signal 11, Segmentation fault. #0 0xb56b41bd in wb_timer (_r=<optimized out>) at urecord.c:319 319 urecord.c: No such file or directory. (gdb) where #0 0xb56b41bd in wb_timer (_r=<optimized out>) at urecord.c:319 #1 0xb56a9602 in mem_timer_udomain (_d=0xa59574e8, istart=0, istep=1) at udomain.c:879 #2 0xb569ae3e in synchronize_all_udomains (istart=0, istep=1) at dlist.c:742 #3 0xb56b0fff in destroy () at ul_mod.c:489 #4 0x0811408c in destroy_modules () at sr_module.c:790 #5 0x0809c6ec in cleanup (show_status=1) at main.c:573 #6 0x0809d529 in shutdown_children (show_status=1, sig=<optimized out>) at main.c:715 #7 0x0809e800 in handle_sigs () at main.c:806 #8 0x0809fe18 in main_loop () at main.c:1767 #9 0x080a1bbe in main (argc=17, argv=0xbff02f74) at main.c:2566 would be nice to get it fixed. -- juha

2 6

Siremis Getting Started - error when logging in
by Mark D. Montgomery II 18 Nov '13

18 Nov '13

I got Siremis installed today but when I try to login it pops up a debug window with the error: Fatal error: Cannot find the class with name as service. in /var/www/admin/siremis-4.0.0/openbiz/bin/ObjectFactory.php on line 157 Any suggestions on what I need to check to fix this so I can actually login and start using it? Thanks. Mark II -- Mark D. Montgomery II http://www.techiem2.net

1 0

Timezone in acc table
by Igor Potjevlesch 16 Nov '13

16 Nov '13

Hi, I'm wondering how can I change the timezone when INVITE, ACK, BYE and CANCEL are inserted into acc table? Indeed, today, each entrie has the same datetime than the server. I would prefer to have UTC time in "time" column instead. I know that it was the behaviour with old SER releases. Thank for your help. Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

3 3

Fwd: sr-users Digest, Vol 102, Issue 53
by santosh satapathy 16 Nov '13

16 Nov '13

---------- Forwarded message ---------- From: santosh satapathy <santosh.satapathy(a)gmail.com> Date: Fri, Nov 15, 2013 at 4:23 PM Subject: Fwd: sr-users Digest, Vol 102, Issue 53 To: Daniel-Constantin Mierla <miconda(a)gmail.com> Appreciate for the input Daniel. I was looking here little different scenario where in weights are dynamic. Load distribution should happen in such a way that first it round robins among the nodes with weight 3 until they are equal with other sets of nodes with weight 2. Once this stage is reached where in all the nodes in both the sets are equally weighted, then LB should round robin among all the nodes uniformly. This means that dynamically nodes with higher weights will get more calls until their weight difference is substantiated and after that all nodes are treated as equal. Is it possible to configure script for the above scenario? Thanks , Santosh ---------- Forwarded message ---------- From: <sr-users-request(a)lists.sip-router.org> Date: Thu, Nov 14, 2013 at 2:26 AM Subject: sr-users Digest, Vol 102, Issue 53 To: sr-users(a)lists.sip-router.org Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Kamailio/Radius problem (Daniel-Constantin Mierla) 2. Re: Variables in event_route[sl:local-response] (Daniel-Constantin Mierla) 3. Re: Query regd the dispatcher algorithm support (Daniel-Constantin Mierla) 4. WORDS_BIGENDIAN error in auth_diameter.h (Sotas Development) ---------------------------------------------------------------------- Message: 1 Date: Thu, 14 Nov 2013 10:06:20 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Kamailio/Radius problem Message-ID: <5284928C.9080300(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" On 11/14/13 10:02 AM, Daniel-Constantin Mierla wrote: > Hello, > > On 11/14/13 4:31 AM, jonathan augenstine wrote: >> I am working on the configuration to enable RADIUS in Kamailio 4.0. >> The basic install worked successfully and the daemon started without >> problems. I then install radiusclient-ng and configured RADIUS in >> the Kamailio build/configuration. All seemed OK until the Kamailio >> restart and I encountered the following error during the restart: >> >> Nov 13 06:52:23 lax-vproxy2-a kernel: [ 348.808624] kamailio[5346]: >> segfault at 0 ip 00007f375d249a5d sp 00007fff4d42eed0 error 4 in >> libradiusclient-ng.so.2.0.0[7f375d243000+c000] >> Nov 13 06:52:23 lax-vproxy2-a kamailio: ERROR: <core> >> [daemonize.c:307]: daemonize(): Main process exited before writing to >> pipe >> >> After some research it seemed that others encountered a similar error >> and traced it to the dictionary. I have gone through the dictionary >> configuration and all seems to be correct. Any guidance on what >> might be the issue or some troubleshooting steps? > did you get a core dump? Looking at the backtrace may show where is > the problem in libradiusclient-ng. Just found this one that seems to be similar: - http://sip-router.org/tracker/index.php?do=details&task_id=310&project=1 It mentions that acctserver was not set in radius config. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

1 0

Kamailio + freeswitch BLF is not working
by José Seabra 15 Nov '13

15 Nov '13

Hello there, Im trying implement BLF with kamailio and snom phones, with the following modules: presence, presence_xml, presence_mwi, pua_dialoginfo, presence_dialoginfo, pua and dialog calls between phones registed in kamailio, BLF works fine. The issue is, when the call comes from freeswitch --> Kamailio --> snom phone, Kamailio creates a Publish but doesn't send notify's to the phones that are monitoring the ringing phone. the logs shows the following: DEBUG: presence [notify.c:1249]: publ_notify(): Could not find subs_dialog Nov 15 16:36:47 vx00-kml01 /usr/local/sbin/kamailio[13953]: DEBUG: <core> [msg_translator.c:2026]: build_res_buf_from_sip_res(): build_res_from_sip_res: copied size: orig:666, new: 596, rest: 302 msg=#012SIP/2.0 180 Ringing#015#012Via: SIP/2.0/UDP 94.46.248.80;received=10.0.20.10;rport=5060;branch=z9hG4bKDgmN81t5NZrDg#015#012Record-Route: <sip:94.46.XX.XX:5060;r2=on;lr=on;ftag=ZvNgee3Uy81tr;lb=yes>#015#012Record-Route: <sip:10.0.20.7:5062;r2=on;lr=on;ftag=ZvNgee3Uy81tr;lb=yes>#015#012Record-Route: <sip:10.0.20.13;lr=on;ftag=ZvNgee3Uy81tr;proxy=yes;did=213.ae12;nat=yes>#015#012From: "256370XXX" <sip:25637XXX@94.46.2X.XXX>;tag=ZvNgee3Uy81tr#015#012To: < sip:+3513045XXXX@admin.com>;tag=fvt7h9az6c#015#012Call-ID: 3ef55a52-4e14-11e3-85cd-71b7082538d9#015#012CSeq: 51930666 INVITE#015#012Contact: <sip:203@192.168.1.180:32768;alias=10.0.20.7~5062~1;alias=213.228.167.67~32768~1;line=sg3y775p>;reg-id=1#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE#015#012Allow-Events: talk, hold, refer, call-info#015#012Content-Length: 0#015#012P-VOICIS-Src-Ip: 213.228.167.67#015#012P-VOICIS-Src-Port: 32768#015#012P-VOICIS-Src-Proto: udp#015#012#015#012 Nov 15 16:36:47 vx00-kml01 /usr/local/sbin/kamailio[13957]: DEBUG: presence [notify.c:1249]: publ_notify(): Could not find subs_dialog Nov 15 16:36:47 vx00-kml01 /usr/local/sbin/kamailio[13955]: DEBUG: presence [notify.c:1249]: publ_notify(): Could not find subs_dialog Anyone can help on this? -- Regards José Seabra

1 0

Presentation of Kamailio v4.1.0 today on VUC
by Daniel-Constantin Mierla 15 Nov '13

15 Nov '13

Hello, a reminder for those that want to participate later today to the VoIP Users Conferences presenting what is new in Kamilio's upcoming release v4.1.0 as well as discussing the future plans. For interactive participation, you can join: - for voice - dial in via sip, skype or pstn - see http://vuc.me - for text chat - join irc channel #vuc on server freenode.net For passive participation, you can: - watch the video streaming - listen the audio streaming The streaming links will be published on http://vuc.me when the session starts (based on past sessions, they should be http://vuc.me/live or http://live.vuc.me for video and http://vuc.me/listen or http://vuc.me/m3u for audio). The session is scheduled to start at 17:00GMT - to learn the time in your zone, click on http://vuc.me/next In the video session will be a consistent group of developers, more details are at: - http://www.kamailio.org/w/2013/10/kamailio-update-on-vuc-nov-15-2013/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

1 0

roni@pomvan.co.uk
by Roni | 15 Nov '13

15 Nov '13

Hi I would like to please join the mailing list

2 1

Kamailio/Radius problem
by jonathan augenstine 15 Nov '13

15 Nov '13

I am working on the configuration to enable RADIUS in Kamailio 4.0. The basic install worked successfully and the daemon started without problems. I then install radiusclient-ng and configured RADIUS in the Kamailio build/configuration. All seemed OK until the Kamailio restart and I encountered the following error during the restart: Nov 13 06:52:23 lax-vproxy2-a kernel: [ 348.808624] kamailio[5346]: segfault at 0 ip 00007f375d249a5d sp 00007fff4d42eed0 error 4 in libradiusclient-ng.so.2.0.0[7f375d243000+c000] Nov 13 06:52:23 lax-vproxy2-a kamailio: ERROR: <core> [daemonize.c:307]: daemonize(): Main process exited before writing to pipe After some research it seemed that others encountered a similar error and traced it to the dictionary. I have gone through the dictionary configuration and all seems to be correct. Any guidance on what might be the issue or some troubleshooting steps? Jonathan

2 4

Kamailio Segfault Dialog.so and http_query inside event_route
by Juan Luis Faylon 15 Nov '13

15 Nov '13

Hi, I need help. I encounter a segfault error in dialog.so when I do an http_query containing shared pseudovariables in the event_routes of dialog:start and dialog:end. How can I fix this?

2 1

kamailio sending qop in auth header
by Daniel W. Graham 15 Nov '13

15 Nov '13

It may be my lack of understanding, but I moved from using auth_check function to www_authenticate function and now see kamailio sending qop in auth header. It's my understanding this is not necessary (based on UA types in use) so I would like to remove it. Per module doc, it is "" by default, I tried adding auth param as below but made no difference. What could be causing this? U x.x.x.x:5060 -> y.y.y.y:5060 REGISTER sip:y.y.y.y SIP/2.0. Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-bd3965f2. From: user <sip:user@y.y.y.y>;tag=682bdce181bb953eo0. To: user <sip:user@y.y.y.y>. Call-ID: 1b886b6c-b283de31(a)x.x.x.x. CSeq: 612 REGISTER. Max-Forwards: 70. Contact: user <sip:100@x.x.x.x:5060>;expires=1800. User-Agent: agent Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. . U y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-bd3965f2;rport=5060. From: user <sip:user@y.y.y.y>;tag=682bdce181bb953eo0. To: user <sip:user@y.y.y.y>;tag=02500e456b34f8655aebe61627730a5a.852c. Call-ID: 1b886b6c-b283de31(a)x.x.x.x. CSeq: 612 REGISTER. WWW-Authenticate: Digest realm="y.y.y.y", nonce="UoV8q1KFe38amcNXXHXmuDU83y1OXg2n", qop="auth". Server: SIP Proxy. Content-Length: 0.. version: kamailio 4.0.3 (i386/linux) 090035 # ----- auth params ----- modparam("auth", "qop", "") # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "id") modparam("auth_db", "use_domain", MULTIDOMAIN) modparam("auth_db", "version_table", 0) ......... if (!www_authenticate("$td", SIP_TABLE)) { www_challenge("$td", "1"); exit; } ......... [cid:image001.gif@01CEE17B.A12FBB50]

1 1

(no subject)
by Roni | 14 Nov '13

14 Nov '13

may i join

4 3

GIT Branch 4.1 has been created for release series 4.1.x
by Daniel-Constantin Mierla 14 Nov '13

14 Nov '13

Hello, the stable branch from where we are going to package the release series 4.1.x has been created - its name: 4.1 There is now a 'kamailio' alias for the repository name, therefore it can be used instead of 'sip-router' in the git path. Getting the git branch 4.1 with read only access: git clone --depth 1 --no-single-branch git://git.sip-router.org/kamailio kamailio cd kamailio git checkout -b 4.1 origin/4.1 For developers that have write access: git clone --depth 1 --no-single-branch ssh://_USERID_@git.sip-router.org/kamailio kamailio cd kamailio git checkout -b 4.1 origin/4.1 Note that older versions of git application may not have the option --no-single-branch. Just remove it when is the case. I created also the wiki tutorial about installing kamailio 4.1.x from git, it is available at: - http://www.kamailio.org/wiki/install/4.1.x/git After few more days of testing, we can loot at nailing down the release date for v4.1.0. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

1 0

Development for v4.2.0 is open
by Daniel-Constantin Mierla 14 Nov '13

14 Nov '13

Hello, a dedicated branch 4.1 has been created for upcoming major release. From now on, the master can take again code with new features to be part of future release series 4.2.x. Any new bug fix that is applicable to stable branches has to be committed first in master and then backported. I will follow with another email describing how to use branch 4.1 for those that want to switch testing from master to it. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

1 0

Kamailio VUC Session on Nov 15
by Daniel-Constantin Mierla 14 Nov '13

14 Nov '13

Hello, we are preparing for a new VUC session to give an update about Kamailio project - a perfect timing as we are just about to release a new major version. We will like to get many developers involved to be able to highlight properly what is new, especially those that did new development for v4.1 - new modules or enhancement to existing modules. However, any developer and community member is welcome to join, we will appreciate it very much, in support of the project. Those that have a google hangout account can participate with video, otherwise there are options to join via sip or pstn audio calls - you can see more details at: - http://vuc.me The number of participants with video is limited, therefore if you plan to do it, let me know to be able to coordinate and send you the invite link when the session starts. No matter you participate with audio/video, you can join the IRC channel #vuc on freenode.net for text chatting during the event. I made quickly a news about the event: - http://www.kamailio.org/w/2013/10/kamailio-update-on-vuc-nov-15-2013/ I will add names of other participants as I get confirmations. Do suggest topics to highlight/discuss there as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

3 3

Disable NAT for given IP range
by David Dean 14 Nov '13

14 Nov '13

Hello all, I have a kamailio server on a public IP address with a private interconnect into my office network. The private interconnect allows the server to see requests from individual 10.x.x.x IP addresses instead of everything coming through the public NAT. This avoids timeout issues and reduces keepalive traffic. The problem is the SIP headers from clients on the 10.x.x.x IP range match nat_uac_test because of the RFC1918 address in the via and contact headers and so NAT support is enabled regardless of no NAT being present. Is it possible to disable NAT support for a given IP range, in my case 10.x.x.x ? Thanks, John

3 2

WORDS_BIGENDIAN error in auth_diameter.h
by Sotas Development 14 Nov '13

14 Nov '13

Hello All, We are currently cross compiling Kamailio for powerpc and are encountering an "issue", of which we want to know whether or not it is intentional. In the modules/auth_diameter/diameter_msg.h of Kamailio 4.0.4 (latest) we found the following line: 72: #error BIG endian detected!! But it looks like if all the code is in place to make big endian function as desired. Is this a leftover debug message? The reason we are asking is because we previously encountered another error with a similar check for WORDS_BIGENDIAN, in MD5.h. It appears as the check is built in on multiple places for this preprocessor directive, however it is not set anywhere. Are we missing a necessary build step somewhere? We have solved the issue for now by patching the diameter_msg.h to remove the #error statement, and start the makefile using: CC_EXTRA_OPTS = "-DWORDS_BIGENDIAN" This seems to solve our initial issue, which was an md5 mismatch on authentication, however we are still unsure if this is a proper workaround, or if we are missing something else. It would be very nice if someone could shed some light on this. Thanks in advance, Sotas Team (Pascal, Yuri).

1 0

Query regd the dispatcher algorithm support
by santosh satapathy 14 Nov '13

14 Nov '13

Does the dispatcher module allow combination like "weighted round robin" algorithm. Example: 3 of 5 nodes are in better condition so I set them as weight 1 and the other two as weight 2. Then the dispatcher will round robin between the 3 nodes with weight 1 until all 5 are with the same weight. That also means that the weight should be dynamic and updated as conditions of the node changes. -- Thanks Santosh

2 2

Variables in event_route[sl:local-response]
by Sergey Okhapkin 14 Nov '13

14 Nov '13

Which variables I can use in event_route[sl:local-response] route to xlog proto:ip:port to which the response will be sent? event_route[sl:local-response] { xlog("L_INFO","Local response to *$??*\n$mb\n"); }

3 2

Kamailio 4.0.4 crash t_load_contact();
by Julia 14 Nov '13

14 Nov '13

Hello, After upgrade from 3.3 to 4.0.4 Kamailio crashed when the first call after application start is a call with failure route: get_redirect(); t_load_contact(); t_next_contacts(); Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:78]: get_redirect(): resume branch=0 Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:85]: get_redirect(): checking branch=0 (added=0) Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:89]: get_redirect(): branch=0 is a redirect (added=0) Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:165]: sort_contacts(): sort_contacts: <sip:39988202@10.25.147.241:5060;user=phone> q=250 Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:165]: sort_contacts(): sort_contacts: <sip:39988202@10.25.148.41:5060;user=phone> q=500 Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:297]: shmcontact2dset(): adding contact <sip:39988202@10.25.147.241:5060;user=phone> Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: <core> [sruid.c:176]: sruid_next(): new sruid is [rdir-5280b7a9-2874-1] (1 / 20) Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:297]: shmcontact2dset(): adding contact <sip:39988202@10.25.148.41:5060;user=phone> Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: <core> [sruid.c:176]: sruid_next(): new sruid is [rdir-5280b7a9-2874-2] (2 / 20) Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: tm [t_serial.c:166]: t_load_contacts(): nr_branches is 2 Nov 11 12:56:00 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10336]: ALERT: <core> [main.c:788]: handle_sigs(): child process 10356 exited by a signal 11 Nov 11 12:56:00 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10336]: ALERT: <core> [main.c:791]: handle_sigs(): core was generated Nov 11 12:56:00 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10336]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLD Regards, Julia.

2 6

Steps to next major release v4.1.0
by Daniel-Constantin Mierla 14 Nov '13

14 Nov '13

Hello, it's now one month since we froze the development for release of v4.1.0. No critical issues in on the table for this particular version, so perhaps next week is time to create a dedicated branch for it, to be named in GIT as 4.1, and open development for v4.2. If all goes fine for one more week or so, then we can do the release. Don't forget to add any open issues you are aware of to the tracker so we can solve them in time. If any of devs or community members have spare time, adding ids to the xml docbook files (for modules) will help getting a better alpha-numeric index for parameters and functions -- see more at: - http://www.kamailio.org/wiki/devel/module-docbook-readme#section_ids Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

3 6

Stuck dialogs
by Daniel W. Graham 13 Nov '13

13 Nov '13

I'm occasionally getting a stuck dialog in my database. I use dialog module for enforcing call limits. I found this post but still not sure what to do: http://lists.sip-router.org/pipermail/sr-users/2013-April/077681.html What should I look for / how should I troubleshoot this issue? -dan

2 8

forward audio to another sip server
by Joli Martinez 13 Nov '13

13 Nov '13

I have all user registration done through kamailio and conferences are handled by freeswitch. I would like to forward all audio to another sip server where the conference can be recorded and we can pull reports. I understand that FS can do the recording, but a box has been setup for this and this is the way my boss wants it. please let me know how to accomplish this thanks,

2 1

extract part of string / INVITE
by Oliver Roth 13 Nov '13

13 Nov '13

Hi all I need to extract a part of the INVITE msg: INVITE sip:+41796159999@82.197.185.185;user=CSC10824 SIP/2.0 I need 10824 in a avp variable. Is there a regex function where I can extract this? Or how can this be done? Something like: Avp(myCSC) = string after CSC - length 5 Please be aware, that ip and destination number in the invite can change ... Otherwise substring would be a solution. Thanks for help Oli

4 3

DBTEXT / AVPOPS and custom database
by Hugh James 13 Nov '13

13 Nov '13

Thanks Daniel, That helped my understanding further along , but it wasn't the full story . If I use $var(rc)=avp_db_load("$to/username","$avp(inbound)/LineIDs"); Then I still get no matches . But if I place entries in the uuid column of the table , then it works. I had assumed that since I don't use the uuid or the domain they could be null . In fact I tried it again with just a number in the uuid column of the entry I expected the avp_db_load to match , and it still worked . I then placed a single entry in the uuid column of a row that I didn't expect it to match and it still worked . I then placed two identical numbers in the UUID column and it still works ( see below for current table used in testing) So in summary it appears that there must be at least one value in the uuid column for it to work as expected . Cheers Hugh

1 0

Kamailio, Keepalived: Kamailio doesn't respond on virtual IP but real IP
by Nguyễn Đức Vinh 13 Nov '13

13 Nov '13

Hi all I want to setup IP failover for Kamailio: I have 2 hosts A (IP: X.X.X.32) and B (X.X.X.36), with 2 running kamailios. Using Keepalive to setup IP failover as , virtual IP *is X.X.X.43*. On host A, kamailio was binded to *listen=udp:0.0.0.0:5060 <http://0.0.0.0:5060>* and host A is MASTER with virtual IP *Problem is:* Client send REGISTER to X.X.X.43:5060. It recieved 401 respond from X.X.X.32:5060 (actual IP of A), not as expected from X.X.X.43:5060 (To check if keepalived setting OK or not, I have *nc -l 5060* on host A, and using nc X.X.X.43 5060 on client, wireshark show that responds is from X.X.X.43 -> keepalived work OK Is there any difference between *kamailio* and *nc* cause they both bind to 0.0.0.0, but repond on differ IP? Thanks Vinh Nguyen

4 3

Function to test DB connection in the cfg file?
by Allen Zhang 12 Nov '13

12 Nov '13

Hi all, This is a re-post. My email didn't seem to get on the list last time. Is there a function I can use to test database connection from the routing script in the cfg file? My dispatcher uses OPTIONS probing. When the destination receives the OPTIONS request, I'd like to detect the health status of the database connection before sending a reply back. Note the OPTIONS request doesn't have a user part in the RURI. Hence, is_subscriber() and alias_db_lookup() don't work. Any suggestion? Regards, Allen Zhang

5 5

Forwarding registrations upstream
by Spencer Thomason 12 Nov '13

12 Nov '13

Hello, I'm trying to use Kamailio as a customer facing SBC to handle NAT traversal and basic SBC functionality for a group of backend FreeSWITCH servers. What I'm trying to accomplish is this: - UAC sends REGISTER - Auth is handled by the edge proxy, backend server trusts edge proxy - If NAT is detected, expires is decreased to 60 seconds, proxy sends options ping - REGISTER is cached and forwarded upstream with Path header and longer expires, i.e. 3600 seconds - UAC reregisters every < 60 seconds but upstream registration is only changed if the timer expires (just less than 3600 seconds) or if AoR changes I good illustration is here: http://www.markholloway.com/blog/?p=1826 My thought is to use htable to store the AoR with an expires just less than 3600, and check it before crafting a new REGISTER with uac module. Is there a better way of handling this scenario? Is there an efficient way populate the $uac_req(…) pseudo variables and call uac_req_send() for multiple, i.e. 4-6 backend servers? I guess all I would need to change is the RURI? Thanks, Spencer

1 0

DBTEXT / AVPOPS and custom database
by Hugh James 12 Nov '13

12 Nov '13

Hi , I'm trying to use the db_text and avpops modules to implement a simple custom database but having problems getting it to work . I'm trying to find the Identity of the recipient for an inbound call based on the called number . I have a database file:- 1) LineIDs - contains the called numbers in the username field and an attribute for that line ID (inbound/ outbound) ,and the recipient identity in the value field I am trying to get the attribute and value pairs that matches the line number in the user name column , id(int,auto) uuid(str,null) username(str) domain(str,null) attribute(str) type(int) value(str) 1::+12125551212::inbound:0:Foster 2::012125551212::inbound:0:Foster 3::12125551212::inbound:0:Foster 4::+4412125551212::inbound:0:Foster 5::+12125551213::inbound:0:Foster 6::012125551213::inbound:0:Foster 7::013440001::inbound:0:Foster 8::call-centre::inbound:0:MBA 9::call-centre::outbound:0:MBA 10::01344665544::inbound:0:chicken 11::01344665544::outbound:0:chicken I try to load the AVPs that match a received number using the following line in the kamailio script file $var(rc)=avp_db_load("01344665544/username","s:recipient/LineIDs"); I have also tried $var(rc)=avp_db_load("01344665544/username","$avp(s:tenants)/LineIDs"); And $var(rc)=avp_db_load("01344665544","$avp(i:123)/LineIDs"); But it fails to find any matching rows ( AVPS) . Am I using the function correctly ? The debug output is shown below , there are some custom debug lines I have added in the db_text C files . DEBUG: avpops [avpops_impl.c:378]: AVPOPS: calls db_load_avp DEBUG: db_text [dbt_base.c:185]: DBTEXT: dbt_query called DEBUG: db_text [dbt_file.c:102]: request for table [LineIDs] DEBUG: db_text [dbt_file.c:109]: db is [/etc/kamailio/Tenants] DEBUG: db_text [dbt_file.c:124]: loading file [/etc/kamailio/Tenants/LineIDs] DEBUG: db_text [dbt_tb.c:187]: mtime is 1383910898 DEBUG: db_text [dbt_file.c:215]: column[0] is INT! DEBUG: db_text [dbt_file.c:210]: column[1] is STR! DEBUG: db_text [dbt_file.c:210]: column[2] is STR! DEBUG: db_text [dbt_file.c:210]: column[3] is STR! DEBUG: db_text [dbt_file.c:210]: column[4] is STR! DEBUG: db_text [dbt_file.c:215]: column[5] is INT! DEBUG: db_text [dbt_file.c:210]: column[6] is STR! DEBUG: db_text [dbt_base.c:240]: new res with 3 cols DEBUG: db_text [dbt_res.c:52]: DBTEXT: dbt_result_new called DEBUG: db_text [dbt_res.c:79]: new res with 3 cols DEBUG: db_text [dbt_res.c:100]: DBTEXT: dbt_result_new returns DEBUG: db_text [dbt_api.c:255]: DBTEXT: dbt_get_result called DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x7f334a85c390 DEBUG: <core> [db_res.c:155]: allocate 24 bytes for result names at 0x7f334a85c3e0 DEBUG: <core> [db_res.c:165]: allocate 12 bytes for result types at 0x7f334a85c1f0 DEBUG: db_text [dbt_api.c:80]: allocate 16 bytes for RES_NAMES[0] at 0x7f334a85c1c0 DEBUG: db_text [dbt_api.c:80]: allocate 16 bytes for RES_NAMES[1] at 0x7f334a85c420 DEBUG: db_text [dbt_api.c:80]: allocate 16 bytes for RES_NAMES[2] at 0x7f334a85c450 DEBUG: db_text [dbt_api.c:281]: DBTEXT: dbt_get_result returns 0 DEBUG: avpops [avpops_impl.c:389]: AVPOPS: db_load_avp returns 0 DEBUG: avpops [avpops_db.c:271]: close avp query DEBUG: <core> [db_res.c:81]: freeing 3 columns DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x7f334a85c1c0 DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x7f334a85c420 DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x7f334a85c450 DEBUG: <core> [db_res.c:94]: freeing result names at 0x7f334a85c3e0 DEBUG: <core> [db_res.c:99]: freeing result types at 0x7f334a85c1f0 DEBUG: <core> [db_res.c:136]: freeing result set at 0x7f334a85c390 DEBUG: avpops [avpops_impl.c:420]: loaded avps = 0 INFO: avp_db_load return result is -1 I was expecting it to return at least one AVP . Any suggestions gratefully received . Hugh James Hugh James | Telephony Developer [icons] * 0845 543 0822 * * Hugh.James(a)semafone.com * www.semafone.com<http://www.semafone.com> * @semafone<http://twitter.com/semafone> * SEMAFONE, 3 The Billings, Walnut Tree Close, Guildford, Surrey GU1 4UL [logo] This e-mail is intended only for the addressee named above. As this e-mail may contain confidential or privileged information if you are not the named addressee or the person responsible for delivering the message to the named addressee, please telephone us immediately. Please note that we cannot guarantee that this message, or any attachment, is virus free or has not been intercepted and amended. The views of the author may not necessarily reflect those of the Company.

2 1

MSILO: remove "Offline message" prefix
by Marc M. 12 Nov '13

12 Nov '13

Hi! I have an up and running msilo config with kamailio 4.0.4. However, all the queued messages are delivered with a prefix "[Offline message - (date)(time)]". Non queued messages are delivered unchanged. Is there a way I can remove this prefix, so also queued messages get delivered unchanged? Please advise! Any help is appreciated! Thanks you, Marc

2 1

rtpproxy-ng ipv4-ipv6 bridge mode problem
by Pavel Miskov 12 Nov '13

12 Nov '13

Hi all, I'm trying to use rtpproxy-ng module (kamailio 4.1 from GIT) and mediaproxy-ng (from https://github.com/sipwise/mediaproxy-ng) in bridge ipv4-ipv6 mode with new NG protocol but keep getting "Failed to extract streams from SDP: No address info found for stream" (log is bellow). Same setup, same SIP clients, same network but with old rtpproxy module and mediaproxy-ng without NG protocol works fine for ipv4-ipv6 bridge mode. of course, only change is in Kamailio configuration where now I load rtpproxy module and change sockets not to use new NG protocol. mediaproxy-ng configuration is untouched. Best regards /Pavel Nov 11 15:58:00 kamailio2 mediaproxy-ng[4790]: Got valid command from 183.24.6.201:57283: answer - { "sdp": "v=0#015#012o=john 0 0 IN IP6 2004:c23:d:caef:3426:666:e36c:12#015#012s=-#015#012c=IN IP6 2004:c23:d:caef:3426:666:e36c:12#015#012t=0 0#015#012m=audio 5076 RTP/AVP 96 9 97 98 100 102 0 8 103 3 104 101#015#012a=rtpmap:96 opus/48000#015#012a=fmtp:96 usedtx=1#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:97 SILK/24000#015#012a=rtpmap:98 SILK/16000#015#012a=rtpmap:100 speex/32000#015#012a=rtpmap:102 speex/16000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:103 iLBC/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:104 speex/8000#015#012a=rtpmap:101 telephone-event/8000#015#012m=video 5078 RTP/AVP 105 99#015#012a=inactive#015#012a=rtpmap:105 H264/90000#015#012a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1#015#012a=imageattr:105 send * recv [x=[0-1440],y=[0-900]]#015#012a=rtpmap:99 H264/90000#015#012a=fmtp:99 profile-level-id=4DE01f#015#012a=imageattr:99 send * recv [x=[0-1440],y=[0-900]]#015#012", "direction": [ "external", "internal" ], "flags": [ "force", "symmetric" ], "call-id": "60a061ef55655c340a4ffc536383c4ad@0:0:0:0:0:0:0:0", "via-branch": "branch_v4_v6", "received-from": [ "?", "2004:C23:D:CAEF:52e6:BB4:C6F2:1CAD" ], "from-tag": "3a21da16", "to-tag": "5668dfd9", "command": "answer" } Nov 11 15:58:00 kamailio2 mediaproxy-ng[4790]: Failed to extract streams from SDP: No address info found for stream Nov 11 15:58:00 kamailio2 mediaproxy-ng[4790]: Protocol error in packet from 183.24.6.201:57283: Incomplete SDP specification [d3:sdp771:v=0#015#012o=john 0 0 IN IP6 2004:c23:d:caef:3426:666:e36c:12#015#012s=-#015#012c=IN IP6 2004:c23:d:caef:3426:666:e36c:12#015#012t=0 0#015#012m=audio 5076 RTP/AVP 96 9 97 98 100 102 0 8 103 3 104 101#015#012a=rtpmap:96 opus/48000#015#012a=fmtp:96 usedtx=1#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:97 SILK/24000#015#012a=rtpmap:98 SILK/16000#015#012a=rtpmap:100 speex/32000#015#012a=rtpmap:102 speex/16000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:103 iLBC/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:104 speex/8000#015#012a=rtpmap:101 telephone-event/8000#015#012m=video 5078 RTP/AVP 105 99#015#012a=inactive#015#012a=rtpmap:105 H264/90000#015#012a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1#015#012a=imageattr:105 send * recv [x=[0-1440],y=[0-900]]#015#012a=rtpmap:99 H264/90000#015#012a=fmtp:99 profile-level-id=4DE01f#015#012a=imageattr:99 send * recv [x=[0-1440],y=[0-900]]#015#0129:directionl8:external8:internale5:flagsl5:force9:symmetrice7:call-id48:60a061ef55655c340a4ffc536383c4ad@0:0:0:0:0:0:0:010:via-branch14:branch_v4_v613:received-froml1:?34:2004:C23:D:CAEF:52e6:BB4:C6F2:1CADe8:from-tag8:3a21da166:to-tag8:5668dfd97:command6:answere] Nov 11 15:58:00 kamailio2 mediaproxy-ng[4790]: Returning to SIP proxy: d6:result5:error12:error-reason28:Incomplete SDP specificatione

2 2

Trunk between Asterisk and Kamailio problem
by Francisco Casasempere 12 Nov '13

12 Nov '13

Hi list, I have this structure IP Phones -> Asterisk Client (LAN) -> (router) -> Kamailio/RTPproxy(Internet) -> Asterisk (local from Kamailio) -> Trunk SIP/Pstn All is working well, I don't have problems with NAT and I managed about 6000 daily calls. For a couple of months I've the next problem.The problem is that in the Asterisk randomly received this message. [Sep 29 04:04:29] NOTICE[3393] chan_sip.c: Peer 'kamONOp' is now Lagged. (2075ms / 2000ms) [Sep 29 04:04:41] NOTICE[3393] chan_sip.c: Peer 'kamailioONO' is now Reachable. (1888ms / 2000ms) In some Asterisk Client I get this message 10 times in a hour, and usually coincide with periods of high load on the kamailo. Also I am getting these messages in the log of Kamailio: Nov 11 08:59:53 c1 /usr/local/sbin/kamailio[8079]: INFO: <core> [forward.c:788]: broken reply to forward - no 2nd via Nov 11 08:59:53 c1 /usr/local/sbin/kamailio[8082]: INFO: <core> [forward.c:788]: broken reply to forward - no 2nd via Are these two issues related? Whay can I do? Kind Regards SiscoCS (Sorry for my english)

2 2

Regd Load Balancer Algorithms
by santosh satapathy 12 Nov '13

12 Nov '13

Hi, I am new to this mailing list and evaluating various options for a good load balancer. Can somebody help let me know if we can choose two algorithms (Ex: weighted and round robin) together for load balancing. -- Thanks Santosh

1 0

SCA Crashes
by Robert Boisvert 12 Nov '13

12 Nov '13

Hi! Looking for some help as to why the SCA module is crashing randomly on 4.0.4. It has crashed several times with a backtrace that shows these stack calls: (gdb) backtrace #0 0x00000000 in ?? () #1 0x4dceca11 in sca_hash_table_slot_kv_delete_unsafe () from /usr/lib/kamailio/modules/sca.so #2 0x4dcecaaa in sca_hash_table_slot_kv_delete () from /usr/lib/kamailio/modules/sca.so #3 0x4dcecb1e in sca_hash_table_index_kv_delete () from /usr/lib/kamailio/modules/sca.so #4 0x4dcecb99 in sca_hash_table_kv_delete () from /usr/lib/kamailio/modules/sca.so #5 0x4dcf30bc in sca_appearance_unregister () from /usr/lib/kamailio/modules/sca.so #6 0x4dd0d9b9 in sca_call_info_update () from /usr/lib/kamailio/modules/sca.so #7 0x126a6a9a in do_action () #8 0x126af88e in run_actions () #9 0x126a4731 in do_action () #10 0x126af88e in run_actions () #11 0x126b007f in run_top_route () #12 0x1269dd7e in receive_msg () #13 0x12734f01 in udp_rcv_loop () #14 0x126d3f56 in main_loop () #15 0x126d76de in main () The message that seemed to trigger the crash in this particular case was the following INVITE: INVITE sip:35705@10.250.16.71;user=phone SIP/2.0 *Record-Route:*<sip:10.250.16.70;lr=on;ftag=D8B14B68-93D21A25;did=513b.a3741> *Via:* SIP/2.0/UDP 10.250.16.70;branch=z9hG4bKc43b.14d180e6.0 *Via:* SIP/2.0/UDP 10.250.88.116;branch=z9hG4bK8d3ec33e632D8AB3 *From:* PIN-1 WKL USA <sip:15862@*obscured*>;tag=D8B14B68-93D21A25 *To:* <sip:0@*obscured*;user=phone> *CSeq:* 1 INVITE *Call-ID:* c69594c-ff17a5d9-d6ed4f6a(a)10.250.88.116 *Contact:* <sip:15862@10.250.88.116> *Allow:* INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER *User-Agent:* PolycomSoundPointIP-SPIP_335-UA/4.0.3.7562 *Accept-Language:* en *Supported:* 100rel,replaces *Allow-Events:* conference,talk,hold *Max-Forwards:* 32 *Content-Type:* application/sdp *Content-Length:* 296 v=0 o=- 1383792680 1383792680 IN IP4 10.250.88.116 s=Polycom IP Phone c=IN IP4 10.250.88.116 t=0 0 a=sendrecv m=audio 2230 RTP/AVP 9 18 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 I have a pcap trace that shows all the calls from 3 hours prior and this is the first time this device appeared. I followed the code down to the sca_hash_table_slot_kv_delete_unsafe ()routine and it appears to have crashed when it tried to use e->free_entry which seems to have been NULL at that point. I'm not sure how it could be NULL since it should have been initialized in sca_hash_table_slot_kv_insert_unsafe (). I'm guessing that either e->free_entry was never initialized or somehow it was overwritten. Since several backtraces end in the same way the bug seems to have be specific to some sequence of events I have not been able to identify. Any help tracking this down would be appreciated. Thanks, Bob

2 2

uninstall kamailio
by Joli Martinez 11 Nov '13

11 Nov '13

Hello, I would like to know how to uninstall Kamailio. I will be upgrading from 3.0 to 3.3 and I would like to uninstall 3.0. I know that there is an upgrade procedure, but 3.0 now does not work and I would like to start from scratch. thanks,

1 0

kamailio dialplan
by Joli Martinez 11 Nov '13

11 Nov '13

I am new to Kamailio and am having an issue with the dialplan setup. I have Kamailio setup as an SBC to handle all user authentication and call routing. I need freeswitch to handle all conferences and voicemails. When I dial 433001 I would like to be transferred to freeswitch for conferences. Right now I have followed the following article and it when I dial 433001 call hangs up and never reaches FS. If I call 43 call does reach FS and I am able to hear FS play the VM prompt. My system is CentoOS 6.4 and FS is installed via yum, but Kamailio is complied. Both FS and Kamailio are on the same box. What commands would you suggest I use to troubleshoot these issues in the future. http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms#dokuwik… Also, since I am new could you give some pointers as far as security and documentation. thanks,

3 4

Re: [SR-Users] tls.so: mod_register failed on fedora 18 x86_64
by Ding Ma 11 Nov '13

11 Nov '13

Hi, all We have applied the openssl 1.0.1e patch described in the email chain between Roberto Fichera and Daniel-Constantin Mierla back in August. But Kamailio 4.0.4 still fails to start up and the log shows memory allocation problem. By the way, we're running kamailio on RHEL 6.5. Wonder if we have missed anything. Please advise. Thanks. Ding

1 0

407 proxy authentication required
by Joli Martinez 11 Nov '13

11 Nov '13

I am setting up Kamailio as an SBC and freeswitch to handle vm and conferences. When I dial 41 Kamailio sends the call to FS and I get the VM prompts, but when I dial 43 the call never hits FS. I am looking at FS in the cli and call never comes in. I get "407 proxy authentication required" from my sip client. I am new to Kamailio and cant seem to figure it out. My system is running CentOS 6.4. FS is installed via yum. Kamailio is complied running version 3.0 thanks,

1 0

implemented specifiers not processed
by Oliver Roth 11 Nov '13

11 Nov '13

Hi all Having some problems with the following part of the script in a timer: We have some kamailios running doing more or less the same job - they are used by a loadbalancer kamailio. Actually I wanted to check the ip or systemname of the current system do perform some actions depending on the used kamailio - based on a timer route[CDRS] { sql_query("ca","call kamailio_cdrs()","rb"); sql_query("ca","call kamailio_rating()","rb"); xlog("timer routine: time is %TF\n"); xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> 2nd via <%{via[1]}>\n"); xlog("L_INFO", "CDRS- rated Host %Hi"); if %Hi = ipOfGateway1 then do something else if %Hi = ipOfGateway2 then do something different } What I get in log Nov 10 12:54:12 sipgw21 /usr/sbin/kamailio[16231]: ERROR: <script>: timer routine: time is %TF Nov 10 12:54:12 sipgw21 /usr/sbin/kamailio[16231]: ERROR: <script>: time [%Tf] method <%rm> r-uri <%ru> 2nd via <%{via[1]}> Nov 10 12:54:24 sipgw21 /usr/sbin/kamailio[16231]: INFO: <script>: CDRS- rated Host %Hi The %-variables are not processed ... what am I doing wrong?? Thanks Oli

2 3

kamailio 4.0.4 Record-Route reply setting
by kamailio＠AaronLux.com 11 Nov '13

11 Nov '13

2 1

Carrierroute module
by Oliver Roth 11 Nov '13

11 Nov '13

Hi I get the following error when trying to load the carrierroute module load_module(): ERROR: load_module: could not open module </usr/lib64/kamailio/modules/carrierroute.so>: /usr/lib64/kamailio/modules/carrierroute.so: undefined symbol: cfg_set_error_function I installed Kamailio 4.0.3 by apt-get on Ubuntu 12.04 lts It seems, that with version 3.3 it is working ... root@sipgw22:/home/triotel# kamailio -V version: kamailio 4.0.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 17:03:55 Aug 19 2013 with gcc 4.6.3

2 3

BUG? - Dialog module timeout BYE is not ok.
by Uri Shacked 11 Nov '13

11 Nov '13

Hi, Following this issue, and the issue "BYE dialog timeout bad syntax" from Julia. I have 2 kamailio servers. srv1 creates a dlg_timeout BYE and sends it to srv2 that forwards the BYE to the next sip server (some other server...). I can definitely see that the BYE that is being forward is malformed. The TO header info is not complete (the port section for example). When a "normal" bye is received from srv1 (not dlg_timeout) all works fine. One thing that i do in this scenario, on srv2 i use uac_replace() on the TO header. So, it is OK that the final destination is different (i do translations). I attached the BYE from both servers. Any ideas? Thanks, Uri You probably meant the To and From headers, I'm guessing. Yeah, as long as there is no display name component, URIs in headers like that don't have to go in <>s. The way Kamailio sends it is grammatically valid, per RFC 3261. The user agents are at fault for not understanding it correctly. Alex Balashov <abalashov at evaristesys.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: >*"BYE headers"? Which headers? *>* *>*Uri Shacked <ushacked at gmail.com <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: *>* *>>*Hi, *>>* *>>*I think there is a bug with the BYE that is sent to the caller and *>>*callee *>>*when dialog timeout happends. *>>* *>> *The BYE headers are sent with no "<" or ">". *>>*So, some sip singaling points decline the BYE. *>>* *>>*BR, *>>*Uri *>>* *>>* *>>*------------------------------------------------------------------------ *>>* *>>*_______________________________________________ *>>*SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing *>*list *>>*sr-users at lists.sip-router.org<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> *>>*http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users *>* *>*-- *>*Sent from my mobile, and thus lacking in the refinement one might *>*expect from a fully-fledged keyboard. *>* *>*Alex Balashov - Principal *>*Evariste Systems LLC *>*235 E Ponce de Leon Ave *>*Suite 106 *>*Decatur, GA 30030 *>*United States *>*Tel: +1-678-954-0670 *>*Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ *>* *>*_______________________________________________ *>*SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list * >*sr-users at lists.sip-router.org<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> *>*http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users * -- Sent from my Nexus 10, with all the figments of autocorrect that might imply. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: < http://lists.sip-router.org/pipermail/sr-users/attachments/20130422/0c75272…>

4 11

Wrong "To" header in CANCEL, when "To" has been rewritten by uac_replace_to() in branch route
by Julia 10 Nov '13

10 Nov '13

Hello, I know it was already discussed, that not recommended to modify "To" header and no routing should ever be done on "To". But as long as there are carriers that do it, we need full support for this manipulation for all messages in the call flow. I found in list a few mails with the same problem of wrong header "To" in CANCEL after "To" rewriting. And one solution from Vitaliy Aleksandrov : http://lists.sip-router.org/pipermail/sr-dev/2012-October/016657.html "Kamailio builds "To" header from the received INVITE. Patch to fix it: --- ./kamailio-3.2.3/modules/tm/t_cancel.c 2012-08-22 10:39:21.000000000 +0300 +++ ./kamailio-3.2.3-fixed/modules/tm/t_cancel.c 2012-10-01 18:25:57.000000000 +0300 @@ -284,7 +284,7 @@ if (cfg_get(tm, tm_cfg, reparse_invite)) { /* build the CANCEL from the INVITE which was sent out */ cancel = build_local_reparse(t, branch, &len, CANCEL, CANCEL_LEN, - &t->to + NULL #ifdef CANCEL_REASON_SUPPORT , reason #endif /* CANCEL_REASON_SUPPORT */" We use header "To" manipulation in branch route (after failure route) and this patch solved our problem with "To" in CANCEL. But a patch is not committed. Is it possible to commit this solution? Thank you, Julia

2 4

Modify/Edit INVITE message.
by Aziz 09 Nov '13

09 Nov '13

Hello. I have a question about Kamailio configuration. Can I edit or modify the INVITE message content? For example, remove "To" or "Via" fields in the INVITE content. If yes, how can I do? Thanks.

2 1

#TM: timing problem in serial forking (INVITE vs. CANCEL)
by Klaus Feichtinger 08 Nov '13

08 Nov '13

Hello list, I have troubles with serial forking in kamailio 3.2.4, which is mixed with parallel forking. In the scenario that an initial INVITE message, which is addressed to sip:A, is coming in to the server, it is doing a lookup in the DB and forking (parallel) the request to e.g. 3 SIP user agents. I have set the timer to 20 seconds transaction timeout and after that timeout, the server is handling the original request in the FAILURE_ROUTE[xy]. In this failure route, the request-URI username is overwritten to an alternative one -- e.g. sip:B. Then the server is doing a DB lookup again and forking the request to the number of registered user agents. A specialiy of this scenario is that it can be possible, that user agents have registered for username "A" and username "B" -- in other words: they are members of the parallel forking group in serial forking step 1 and step 2. When the CANCEL and INVITE message would be sent out (to the user agents) in the correct order, then it would be no problem. But in my case the server is sending the "new" INVITE message (2^nd step in the serial forking procedure) to user agents BEFORE the CANCEL request. Therefore, these user agents are rejecting the INVITE message with "500". Signalisation scenario ================== INVITE -> SRV SRV -> INVITE (branch-1) UA1 SRV -> INVITE (branch-2) UA2 SRV -> INVITE (branch-3) UA3 SRV <- 180 (branch-1) UA1 SRV <- 180 (branch-2) UA2 SRV <- 180 (branch-3) UA3 ..... [timeout] SRV -> CANCEL (branch-1) UA1 SRV -> CANCEL (branch-2) UA2 SRV -> INVITE (branch-4) UA4 SRV -> INVITE (branch-5) UA5 SRV -> INVITE (branch-6) UA3 (!!!) SRV -> CANCEL (branch-3) UA3 SRV <- 500 (branch-6) UA3 It was quasi reproduceable that only the last branch of the initial transaction had that timing problem (INVITE <- vs -> CANCEL). My question is: why does the server send the (last) CANCEL message only after the INVITE message for some branch(es)? Could this behaviour be prohibited in any way? Thanks in advance! Klaus

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UUID generation
by Grant Bagdasarian 08 Nov '13

08 Nov '13

Hello, Any plans for building a module which generates a UUID or building it into the core? Daniel once told me to use $sruid, which basically returns a unique value. It works good. I haven't had any duplicates yet, even under heavy load tests.

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