- sr-users - kamailio.org

Documentation DMQ module
by Jan Hazenberg 29 Jun '16

29 Jun '16

Hi, Found a error in the documentation of the DMQ module (http://kamailio.org/docs/modules/devel/modules/dmq.html#dmq.p.num_workers). In the documentation there is a example: modparam("dmq", "num_threads", 4) If i'm correct this should be: modparam("dmq", "num_workers", 4) Thanks, Jan

2 1

rtpengine allow_op issue
by david escartin 28 Jun '16

28 Jun '16

hello all i have observed an issue using this parameter, which is useful in case you deactivate a node from the rtpengine sets, to let the calls to finish the negotiation. but i have tested that if a node fails down in the middle of a call, with that option activated, the kamailio gets blocked waiting until the rtpengine node is again up, to finish handling with the BYE request for instance. Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:55 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:55 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26571]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26571]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> could it be possible to make the kamailio leave that blocking state, or at least be able to activate the allow_op param by kamctl fifo? thanks a lot and regards david escartin

1 0

Kamailio - PSTN (kamailio) - rtpproxy issue
by Filippo Graziola 28 Jun '16

28 Jun '16

Hello everybody, I have a local kamailio+rtpproxy under NAT listening on multiple local interfaces (vlans) which works fluenty for all local user indipendently and engages rtpproxy correctly. I have another working kamailio+rtpproxy which is accessible from everywhere at a public address and I would like to route to it some specific calls from the local server. So, in the local kamailio configuration file I defined WITH_PSTN (all the necessary option) and INVITE/OK/BYE messages are going correctly to the destination. The problem is that I have no audio on both ways. After some packet sniffing, I tried killing local rtpproxy process and now calls to PSTN have also audio both ways. >From these tests, if my configuration is correct, when routing the call to PSTN gateway I should disable the local rtpproxy, and I'm in doubt about how I can do this. Could you help me? Should I change NAT flags in the PSTN route? Thanks

1 2

Kamailio v4.4.2 Released
by Daniel-Constantin Mierla 28 Jun '16

28 Jun '16

Hello, Kamailio SIP Server v4.4.2 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.2. For more details about version 4.4.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/06/kamailio-v4-4-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Enjoy the summer holidays! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

acc time mode 2 wrong timestamp
by pars3c 28 Jun '16

28 Jun '16

Hello, i've installled the new version of kamailio 4.4.1 . in 2 different server , with acc module and these parameters: modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "seconds") should be seconds.millisec On the first server in acc table , in field seconds i have values like this 1466493457.85, this is correct timestamp ( this server have centos 86_64) In the other server , i have values like 1913113.702 , that is wrong timestamp . (this is old centos 32 bit ) What can be the problem? The different version 32,64 bit ? or some compiler option ? Thanks

2 11

Planning release of v4.4.2 and other v4.x.y
by Daniel-Constantin Mierla 28 Jun '16

28 Jun '16

Hello, I am considering to release kamailio v4.4.2 next week along with the corresponding patch releases for branches 4.3 and 4.2. It will likely be the last release in series 4.2 to mark its end of official packaging. I would also consider a new patch release in 4.1.x, if that is still used by many out there to get packages that fix the libssl version matching -- Victor should say if the packages for this branch can still be generated in order to make sense doing this release. Anyhow, report any issues you are aware and are not yet announced for branches 4.4 and 4.3 to the bug tracker as soon as possible to get a chance to be fixed by release date. Likely 4.4.2 will be next Tuesday, June 29, with the other branches done in the following days. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

userblacklist use avp for check_blacklist
by Jan Hazenberg 28 Jun '16

28 Jun '16

Hello, I'm trying to use the userblacklist module to implement a global blacklist. I have a few questions about the module. Is it possible to use a avp for the check_blacklist function? If i'm correct it will only parse the request URI, but i'm rewriting the URI in a branch route to allow different formats to different destinations. Or is this only possible with the check_user_blacklist function? I also have a issue with numbers in E164 format with a leading +. I tried to escape this with a \ in the database but it will not load the entry's when i use the + in front of the number. Is the + supported by this module or should i strip it before the function and add it again afterwards? Thanks, Jan

2 12

cgrates - cdrs_tables.sql error
by Infinicalls Infinicalls 27 Jun '16

27 Jun '16

Hi, I am trying to install cgrates , but face this issue with database. " ERROR 1067 (42000) at line 7: Invalid default value for 'deleted_at' " is encountered in create_cdrs_tables.sql . Can somebody let me know how to overcome this? regards Ganesh Kumar

2 1

Re: [SR-Users] S-CSCF (kamailio) crashes after deregistration of client
by Christoph Valentin 27 Jun '16

27 Jun '16

2 1

behaviour of uac_redirect module
by Dirk Teurlings - Signet B.V. 27 Jun '16

27 Jun '16

Hi, Testing with latest stable 4.4 Kamailio at the moment, and running into the following issue. I have a setup that sends a 302 to do a Moved Temporarily to another extension. To update the Contact header accordingly, I use the get_redirects function in the failure_route. get_redirects('*', 'Redirects') This results in the following packet U 192.168.10.246:5060 -> 192.168.10.245:5060 SIP/2.0 302 Moved Temporarily. Via: SIP/2.0/UDP 192.168.10.245:5060;branch=z9hG4bK40b58e09;rport=5060;received=192.168.10.245. From: "+31........" <sip:+31........@192.168.10.245>;tag=as42745abe. To: <sip:+31........@192.168.10.246>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-f67d. Call-ID: 576349c40a7cb77a06230e181831672d@192.168.10.245:5060. CSeq: 102 INVITE. Contact: <sip:06........@sip.example.net>, <sip:06........@sip.example.net>;q=0.01. Server: sip.example.net. Content-Length: 0. . This works, but I don't want ACC to account for anything, so I tried: get_redirects('*') But this results in the following packet U 192.168.10.246:5060 -> 192.168.10.245:5060 SIP/2.0 302 Moved Temporarily. Via: SIP/2.0/UDP 192.168.10.245:5060;branch=z9hG4bK185f4fdd;rport=5060;received=192.168.10.245. From: "+31........" <sip:+31........@192.168.10.245>;tag=as13f23689. To: <sip:+31........@192.168.10.246>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-b5fe. Call-ID: 336a16912db412d418f7f7495e2574f5@192.168.10.245:5060. CSeq: 102 INVITE. Contact: <sip:accountname@sip.example.net>, <sip:06........@sip.example.net>;q=0.01, <sip:06........@sip.example.net>;q=0.01. Server: sip.example.net. Content-Length: 0. . Notice the extra accountname Contact, this results in a forward that doesn't work for the mediaserver. To me the documentation about the function states that there should be no difference between using it with or without accounting as far as SIP messages go. Is this a bug, or is this by design? Cheers, Dirk

2 4

Question about gzcompress module
by Joel Serrano | VOZELIA 26 Jun '16

26 Jun '16

Hi, When using the gzcompress module, is it possible to compress/decompress only one specific header, or will it compress/decompress the whole body? Thanks, Joel.

2 2

Get full body from http POST request
by Joel Serrano | VOZELIA 26 Jun '16

26 Jun '16

Hi, I'm using http_query (POST method) to interact with an API. This API responds with a JSON. Is there a way to get the full body and after parse that JSON? Example response: { "error": { "code": 404, "message": "Destination not Found" } } Currently, I'm only being able to get the first line, in the previous example it would be: "{" According to the doc, that is the expected behavior: If HTTP server returns a class 2xx, 3xx or 4xx reply, the first line of the reply's body (if any) is stored in “result” parameter, which must be a writable pseudo variable. I don't have the ability to get the response in 1 single line like: { "error": { "code": 404, "message": "Destination not Found" } } How can I get the full body instead of only the first line? Thanks. Best regards, Joel.

1 1

Parsing diversion headers correctly
by Joel Serrano | VOZELIA 26 Jun '16

26 Jun '16

Hi, If you receive an INVITE with 2 diversion headers such as (in this order): Diversion: <sip:9975@X.X.X.X:5060>;privacy=off;screen=no; reason=user-busy; counter=1 Diversion: <sip:1234567890@Z.Z.Z.Z:5060>;privacy=off;screen=no; reason=unknown; counter=1 How is a nice way of extracting the user part from the URI of the last one? (in this case: 1234567890) Problems I have found: $(di{uri.user}) ---> Give mes the user part of the URI of the first header, in this case: 9975 Is there a way to tell $di to get its value from the last header instead of the first one? This example works: $(hdr(Diversion)[-1]{param.name,0}{s.strip,1}{s.striptail,1}{uri.user}) But I think it is very hacky... Any better approach anyone can suggest me? Thanks in advance. Best regards, Joel.

2 2

Public vs Internal IPs
by Colin Morelli 25 Jun '16

25 Jun '16

Hey all, When using Kamailio at the edge - what's the best practice around how to advertise your Record-Route? I assume it's going to involve the use of a double-RR with both the public and private IPs. However, I'm running in AWS where the host doesn't have two interfaces with both a public and private IP. So, I set the host's public IP as the advertised address while Kamailio binds to the local IP. But now how do I get Kamailio to know that it needs to double-RR in this case? Thanks, Colin

2 9

Specify alternative OpenSSL engine for Kamailio
by Joel Serrano | VOZELIA 25 Jun '16

25 Jun '16

Hi, This is a similar question on a previous thread: "[SR-Users] Compiling kamailio with custom openssl" Is it possible to specify a different OpenSSL engine for kamailio to use? For example: On nginx you have the config param: ssl_engine <engine> On apache you have the config param: SSLCryptoDevice <engine> Is there anything similar on Kamailio? We are using debian 8 with openssl-1.0.1t and these are the available engines: root@debian:/usr/lib/x86_64-linux-gnu/openssl-1.0.0/engines# ls -l total 404 -rw-r--r-- 1 root root 19512 Jun 21 11:07 lib4758cca.so -rw-r--r-- 1 root root 19784 Jun 21 11:07 libaep.so -rw-r--r-- 1 root root 15576 Jun 21 11:07 libatalla.so -rw-r--r-- 1 root root 6104 Jun 21 11:07 libcapi.so -rw-r--r-- 1 root root 24232 Jun 21 11:07 libchil.so -rw-r--r-- 1 root root 19864 Jun 21 11:07 libcswift.so -rw-r--r-- 1 root root 6104 Jun 21 11:07 libgmp.so -rw-r--r-- 1 root root 93304 Jun 21 11:07 libgost.so -rw-r--r-- 1 root root 15432 Jun 21 11:07 libnuron.so -rw-r--r-- 1 root root 6104 Jun 21 11:07 libpadlock.so -rw-r--r-- 1 root root 24096 Jun 21 11:07 libsureware.so -rw-r--r-- 1 root root 19784 Jun 21 11:07 libubsec.so root@debian:/usr/lib/x86_64-linux-gnu/openssl-1.0.0/engines# The reason is we have another server with a Cavium SSL card, which provides a dynamic engine: -rw-r--r-- 1 root root 127968 Jun 23 15:18 libcavium.so If it is not possible via config parameter, is there a way to specify the openssl engine to use before compiling? Thanks in advance. Best regards, Joel.

2 2

Count number of headers of type
by Joel Serrano | VOZELIA 25 Jun '16

25 Jun '16

Hi, It there an easy way to count the number of existing headers of certain type? For example: - How many 'via' headers: $hdrcount(Via) ? - How many 'diversion' headers: $hdrcount(Diversion) ? The $hdrcount is an example, just wondering if there is an easy way to get the number. Thanks. Best regards, Joel.

2 2

t_on_failure route not being called
by Colin Morelli 24 Jun '16

24 Jun '16

Hey all, I'm using a combination of http_async_client and rtjson to query my API and retrieve a JSON target route set for an incoming SIP request that Kamailio will forward to. The HTTP portion of it works great. I'm able to hit my API and get back a JSON document. Additionally, the rtjson parsing appears to work, because the request is hitting my freeswitch instance. However, FS is responding with a 503 (I'm intentionally doing this for testing), and Kamailio isn't calling my t_on_failure route. I've pasted the relevant portion of my config below - would be great if anyone has any insights. tl;dr: the "REROUTE" route section below isn't being reached even when FS responds with a 503. Best, Colin route[ROUTING_REPLY] { if $http_ok && $http_rs == 200 { xlog("L_INFO", "$ci|log|loaded target route set from document $http_rb\n"); rtjson_init_routes("$http_rb"); rtjson_push_routes(); xlog("L_INFO", "$ci|log|attempting to relay request towards $du\n"); t_on_failure("REROUTE"); t_set_fr(0, 1000); t_relay(); } else { send_reply("480", "Temporarily Unavailable"); } } route[REROUTE] { xlog("L_WARN", "$ci|log|relay to destination $du failed, checking next route\n"); if (rtjson_next_route()) { xlog("L_INFO", "$ci|log|additional route found, directing request to $du\n"); t_on_failure("REROUTE"); t_set_fr(0, 1000); t_relay(); } else { xlog("L_WARN", "$ci|log|route list exhausted, failing request\n"); send_reply("480", "Temporarily Unavailable"); exit; } }

1 1

S-CSCF (kamailio) crashes after deregistration of client
by Christoph Valentin 24 Jun '16

24 Jun '16

4 12

Kamailio behind ELB AWS
by yann christophe 24 Jun '16

24 Jun '16

Hello Everyone, I would like to know, if i can use ELB and route53 behind some Kamailio servers for the high availibility ? I use Kamailio behind Freeswitch. I would like deploy these servers on AWS. What do you suggest me for the high availibility on AWS with Kamailio. Thanks Best regards Yann

2 1

rtpengine installation delete ethernet interface on Centos
by Yasin CANER 24 Jun '16

24 Jun '16

1 0

How to understad that user is de-registrating
by Alexandru Covalschi 24 Jun '16

24 Jun '16

Hello list, I need to send to an external API events when user is registrated and de-registrated. As far as I understand standart behaviour is as follows: If user is not registered, he sends REGISTER and he is registrated (I can catch that because I make the auth). If user is registered and sends REGISTER he is de-register. (Please correct me if I'm wrong.) How can I catch that? Can I use event_route[usrloc:contact-expired]? Thanks in advance! -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator tel: +37367398493 web: http://abriss.solutions/ <http://abs-telecom.com/>

3 4

uacreg flags field not changing
by Yuriy Gorlichenko 24 Jun '16

24 Jun '16

Hi. Im using kamailio 4.4.1 kamailio -v flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 99e1c3 compiled on 13:34:38 Jun 14 2016 with gcc 4.9.2 I using uacreg table for register trunks and want to know their status For now i see status onlu from kamcmd uac.reg_dump/info But if i trying to take flags from uacreg table (flags field) it always 0.

1 0

Compiling kamailio with custom openssl
by Cibin Paul 24 Jun '16

24 Jun '16

Hi, Can any one guide me to compile kamailio with custom openssl path please. Thanks & Regards Cibin

2 12

Dialog timeout before answer
by Alex Balashov 24 Jun '16

24 Jun '16

Hi, Will the dialog timeout - and specifically, $dlg_ctx(timeout_bye) - also apply to dialogs which have not yet been established? If so, does that mean the dialog module will send a CANCEL instead for a dialog in an early/progress state? -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

kamailio + fax "other fax not responding"
by Mx R 23 Jun '16

23 Jun '16

Hi there, We use kamailio 4.3.3. And we have one problem with the fax. We send and receive from and to our network, the fax does not work. Fax reports "other fax not responding". When we send through our network to another operator they receive fax. What could be wrong?

2 5

Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by Mx R 23 Jun '16

23 Jun '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0

3 5

Dialog in delete state is too old
by Emilio Soler Muñoz 23 Jun '16

23 Jun '16

Hello! Kamailio process has stopped. Anyone else has happened? Versión: 4.4 This is what I have in the logs. kamailio.log: 2016-06-23T10:10:33.983494+02:00 kam01 /usr/sbin/kamailio[2907]: NOTICE: dialog [dlg_hash.c:245]: dlg_clean_run(): dialog in delete state is too old (0x7f092e15b190 ref 3) 2016-06-23T10:10:34.005015+02:00 kam01 /usr/sbin/kamailio[2897]: WARNING: dialog [dlg_handlers.c:1348]: dlg_onroute(): inconsitent dlg timer data on dlg 0x7f092e15b190 [3554:5262] with clid 'L, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PU' and tags 'as6b8e8af9' 'as25960d84' 2016-06-23T10:10:34.007989+02:00 kam01 /usr/sbin/kamailio[2913]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 17 2016-06-23T10:10:34.008119+02:00 kam01 /usr/sbin/kamailio[2823]: ALERT: <core> [main.c:739]: handle_sigs(): child process 2897 exited by a signal 11 2016-06-23T10:10:34.008128+02:00 kam01 /usr/sbin/kamailio[2823]: ALERT: <core> [main.c:742]: handle_sigs(): core was not generated kern.log: [363406.951756] kamailio[2897] general protection ip:7f092c9413a5 sp:7ffdd1065530 error:0 in dialog.so -- Emilio Soler Muñoz <emiliosoler(a)gmail.com>

1 0

Add a user through a http request
by Jay Li 22 Jun '16

22 Jun '16

Dear All, I wonder if Kamailio has the functionality to add a subscriber dynamically. For example, if I use event_route[xhttp:request] to expect a http get request, then based on the received request, add a user in the DB. Not sure what the best approach would be. Any suggestion is appreciated. Thanks. Regards,Jay

2 1

Dispatcher - Route call to alternate server if there is delay to answered call
by Amit Patkar 22 Jun '16

22 Jun '16

Hi I have a scenario where dispatcher is used to balance calls. Assume there are 2 media servers M1 & M2. Call are routed using round robin algorithm. There is scenario where call is handed over to M1 but it is not answered in defined time period (say 5 seconds due to load on server or network congestion), I wish to send this call to M2. What are the options to achieve it? Thanks & Regards, Amit Patkar

2 1

Re: [SR-Users] uac.reg_refresh not removing records from memory
by DanB 22 Jun '16

22 Jun '16

Hey Daniel, Thanks for so fast feedback! No difference as far as I can tell before and after removing the record from db. I have also not disabled specifically the record inside uacreg table but completely removed it from db. I should mention that I have just upgraded to latest 4.4.1 so I should have the flags also saved in db. Here is the record before removal: """ root@iPBXDev1:/etc/ipbxrp# kamcmd -s tcp:127.0.0.1:2046 uac.reg_dump { l_uuid: rsua1_172.16.254.101 l_username: rsua1 l_domain: 172.16.254.101 r_username: cust_sua1 r_domain: 172.16.254.102 realm: auth_username: cust_sua1 auth_password: check123 auth_proxy: sip:172.16.254.102:5060 expires: 3600 flags: 20 diff_expires: 3588 timer_expires: 1466349066 reg_init: 1466345451 reg_delay: 0 } """ and the one after: """ root@iPBXDev1:/etc/ipbxrp# kamcmd -s tcp:127.0.0.1:2046 uac.reg_dump { l_uuid: rsua1_172.16.254.101 l_username: rsua1 l_domain: 172.16.254.101 r_username: cust_sua1 r_domain: 172.16.254.102 realm: auth_username: cust_sua1 auth_password: check123 auth_proxy: sip:172.16.254.102:5060 expires: 3600 flags: 20 diff_expires: 3526 timer_expires: 1466349066 reg_init: 1466345451 reg_delay: 0 } """ Also here is the command refreshing the record (after that is not longer in db): """ POST /jsonrpc HTTP/1.1. Host: 172.16.254.101:5090. User-Agent: Go-http-client/1.1. Content-Length: 85. Content-Type: application/json. Accept-Encoding: gzip. . {"jsonrpc":"2.0","method":"uac.reg_refresh","params":["rsua1_172.16.254.101"],"id":7} # T 2016/06/19 16:17:47.275473 172.16.254.101:5090 -> 172.16.254.102:58172 [AP] HTTP/1.1 200 OK. Sia: SIP/2.0/TCP 172.16.254.102:58172. Content-Type: application/json. Server: Sipean-RP 4.3.2. Content-Length: 36. . {"jsonrpc":"2.0","result":{},"id":7} """ Ta, DanB On 16.06.2016 12:00, sr-users-request(a)lists.sip-router.org wrote: > Message: 10 Date: Thu, 16 Jun 2016 08:17:53 +0200 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] uac.reg_refresh not removing records from memory > Message-ID: <76d9f2eb-26f7-c0db-8b74-37c93a5d3ef4(a)gmail.com> > Content-Type: text/plain; charset=utf-8 Hello, On 14/06/16 16:52, DanB > wrote: >> > >> >Hey Guys, >> > >> >I was wondering if there is any dynamic way today to remove a single >> >registration out of memory (and implicitly to force un-REGISTER) >> > >> >Eg: I add one record using uac.reg_refresh (which works fine) but after >> >removing the record from the database and executing uac.reg_refresh >> >again, the record stays in memory until expire. >> >Is there a better way to enforce un-register on remote side? >> > > do the flags in memory of the uac_reg account show that it is disabled? > > Cheers, > Daniel

4 3

WS to WS calls - No ACK received for 200OK
by Amit Patkar 21 Jun '16

21 Jun '16

Hi I am using Kamailio as Websocket proxy. User 1 & User 2 are registered on Kamailio over WebSocket. When User 1 calls User 2, User 2 gets ring and answers the call. 200 OK message is received by User 1 but ACK response sent by User 1 does not reach User 2. Since User 2 didn't get ACS, after 30 sec timeout it drops the call. Media is exchanged for 30 seconds, which means ICE is successful. I am running kamailio 4.3.3 on Kamailio is behind firewall and running on private IP. rtpengine is configured to handle media. All ports are forwarded to kamailio server and Kamailio is allowed to send data on any public IP. I used jssip & sipml framework to test. Result was same for both frameworks. No ACK received I can see following errors in log file. What could be the reason which indicate ACK was not forwarded Jun 21 14:55:41 acstemplate /usr/sbin/kamailio[6584]: NOTICE: acc [acc.c:317]: acc_log_request(): ACC: transaction answered: timestamp=1466501141;method=INVITE;from_tag=7926B9NR7U0X99ZfAr1T;to_tag=w3D2JGd2EUFJpobTcFyo;call_id=1edc8e15-f1f1-584d-df81-71c3d59d713d;code=200;reason=OK;src_user=10001;src_domain=xxxx.com;src_ip=xxx.yyy.zzz.aaa;dst_ouser=10002;dst_user=10002;dst_domain=df7jal23ls0d.invalid Jun 21 14:55:41 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:41 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:55:42 acstemplate /usr/sbin/kamailio[6584]: INFO: <script>: WS to WS Jun 21 14:55:42 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:42 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:55:46 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:46 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:55:51 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:51 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:55:57 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:57 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:56:13 acstemplate /usr/sbin/kamailio[6581]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:56:13 acstemplate /usr/sbin/kamailio[6581]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) I searched list for similar issues but could not find match any Help to resolve this issue is appreciated. Thanks & Regards, Amit Patkar

3 4

Kamailio consult CNAM server through SIP Subscribe.
by José Seabra 21 Jun '16

21 Jun '16

Hello there, I need to use kamailio to consult an CNAM server in order to get the caller id name and number to the call, the diagram bellow represents what i need implement: Can I implement this behavior through the following kamailio modules: 1. tm and tmx module (call the function t_suspend and t_continue in order to stop INVITE transaction and resume it). 2. UAC module (generate the SUBSCRIBE sip message) 3. Some kamailio scripting logic to manage this behavior. what are your advices to implement this scenario? Thank you for your help Regards -- Cumprimentos José Seabra

4 17

kamailio question
by Irakli Nanobashvili 21 Jun '16

21 Jun '16

dear sr_users i have a little problem about kamailio sip server and who can help me ?

2 1

Equivalent of OpenSIP's binary interface in Kamailio
by Gholamreza Sabery 21 Jun '16

21 Jun '16

In OpenSIPS binary interface ( https://www.opensips.org/Documentation/Interface-Binary-2-2) provides a more efficient way of replicating state between multiple OpenSIPs instances. What is the equivalent feature in Kamailio? Regards

2 2

RTPENGINE rtp errors
by Dmitry 21 Jun '16

21 Jun '16

Hi 1)I encountered such errors when I make an internal call between 2 yate client SIP softphones:rtpengine is used and no audio: | Jun 19 10:02:35 kazootest2 rtpengine[2066]: [918508371(a)kazootest2.siplabs.local port 30942] Error parsing RTP header: short packet (header) when I turn off the rtpengine:THe Freeswitch handles such RTP packets correctly and audio is ok. I compared manually rtp packets from Yate: they comply to the RTP specification. 2) A call from WS client(jssip) to an ordinary sip softphone: | Jun 19 09:58:36 kazootest2 rtpengine[2066]: [s2m2k70tu0d12bjnnamr port 30814] Error parsing RTP header: invalid header version | How to fix these errors? The rtpengine looks through all incoming packets while looking for a new RTP stream. May be packets parsing is too strict? |

1 1

No other running SIP apllication on the machine.//答复: Kamailio ERROR "Address already in use" in tcp_init().
by joey.ren＠yulong.com 20 Jun '16

20 Jun '16

2 1

Kamailio dispatcher module - keep alives
by José Seabra 20 Jun '16

20 Jun '16

Hello there, I have a Kamailio server using dispatcher module to monitor several peers, some of them are in the same network, others are in an external network, for the peers that are in the external network kamailio has to send the OPTIONS(dispatcher keep alives) through another kamailio located in the border and acting as loadbalancer. If i set the dispatcher option "outbound_proxy" all SIP OPTIONS are sent to the outbound proxy configured and it isn't what i need, I need to have by some way a possibility to set outbound_proxy only for the peers that are located on the external network. How can I do that? Regards -- Cumprimentos José Seabra

3 5

Kamailio ERROR "Address already in use" in tcp_init().
by joey.ren＠yulong.com 20 Jun '16

20 Jun '16

3 2

Why SDPOPS does not remove attributes in SDP
by Dmitry 18 Jun '16

18 Jun '16

Hi allI have the following code: if($T_reply_code=="200") { if(has_body("application/sdp")) { xlog("L_INFO", "RTPENGINE received internal reply $T_reply_code $rr SDP extra lines will be removed"); set_rtpengine_set("0"); rtpengine_manage(); sdp_remove_line_by_prefix("a=rtcp"); sdp_remove_line_by_prefix("a=ssrc"); sdp_remove_line_by_prefix("a=ice"); sdp_remove_line_by_prefix("a=candidate"); xlog("L_INFO", "RTPENGINE received internal reply $T_reply_code $rr SDP extra lines removed with SDPOPS"); } }When I look through traces - I see that 200 ok(with SDP) has all these attributes and they are not removed. Why SDPOPS does not remove these attributes?

4 5

Help with routing block and Dispatcher Module
by pablo rosales 17 Jun '16

17 Jun '16

Hi everyone! I am a newbie with Kamailio. I'm trying to connect a kamailio server to an asterisk gateway. We are using basic Kamailio 4.4.x configuration file with mysql and rtpproxy we are having same issues with dispatcher module. we can not call between local subscribers. Basically, If I call to a local subscriber, the module always execute the route(DISPATCHER) code on the routing block, trying to send the call to my gateway, but if I try to call to an external number, it works perfect. All calls are being send to my gateway. Here is my dispatcher code: # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } ### xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); xlog(" --- DISPATCH: going to <$ru> via <$du>n"); t_on_failure("RTF_DISPATCH"); return; } ## Sample failure route failure_route[RTF_DISPATCH] { xlog(" --- DISPATCH: Starts RTF_DISPATCH \n"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } Can anybody help to see what I am doing wrong??

2 2

http_client module configuring User-Agent
by Jurijs Ivolga 17 Jun '16

17 Jun '16

Hi All, I tried today new module http_client. I just made GET request using http_client_query in following way: http_client_query("http://cool.api.com", "$var(result)"); It is working, but I can't see User-Agent in request. Below you can find my module options: modparam("http_client", "useragent", "kamailio") modparam("http_client", "connection_timeout", 1) Is it a bug, or am I doing something wrong? Thank you! Jurijs

2 3

Kamailio Problem carrying media
by Sirvan Paraste 16 Jun '16

16 Jun '16

Hi there We have got a situation that needed your help to overcome that. We have setup a Kamailio with voice, video call and text messaging and even push notification ability. it was working and till these days that we have lost media (voice and video and calling ability). seems that the clients do not see each other but strange is that they can send text messages and even push notification is work as well. Could you please give me an idea how to find the problem, or if you know how to solve it? Best regards. Sirvan Parasteh

2 1

Kamailio on multiple ports
by huseyin kalyoncu 16 Jun '16

16 Jun '16

Hello, We have kamailio running as proxy in front of a freeswitch box. This kamailio also acts as a registrar listening both 5060 and 5090 ports. We have two devices registered to kamailio: Device A on port 5060 Device B on port 5090 When Device B calls Device A everything is fine. Call released successfully. When Device A calls Device B, call setup is ok and they can talk. But when Device A tries to release the call, Device B remains open which cannot get a BYE message. When I dug up for the cause of the problem, I realized that: When Kamailo receives 200OK reply on port 5090 for the UPDATE message from Device B, it relays this reply internally and then sends this reply to freeswitch box on port 5060. This results in loosing alias in the contact field of the reply message and then which causes freeswitch to return BYE message with the wrong IP which is the local IP of the Device B. As a result this BYE message cannot be delivered to Device B. Is this a normal behaviour ? If it is so what changes should I make in the kamailio config to fix this? You can find the ngrep outputs of the both calls as attachment. Thanks Huseyin Kalyoncu

1 1

just a quick glance at my kamailio.cfg (suspect a rtpproxy_manage behavior)
by Sébastien Brice 16 Jun '16

16 Jun '16

Hi guys i am bridging call on a private ip asterisk behind kamailio/rtpproxy i am calling rtpproxy with the route [FROMASTERISK] route[FROMASTERISK] { if(ds_is_from_list()){ xlog("L_INFO","$dd Call from Media-Server Cluster\n"); rtpproxy_manage("cawei"); SIP INVITE with sdp always report a corrupt body!!! Session Name (s): Asterisk PBX 13.9.1 Connection Information (c): IN IP4 PUBLICIPPUBLICIP Connection Address: PUBLICIPPUBLICIP thats really strange since i am calling rtpproxy_manage only once i can't understand it appears twice. what do you think about that ? thx you

2 2

ACK not sent out
by Nagorny, Dimitry 16 Jun '16

16 Jun '16

Hello List, I have a setup of two WebRTC-Clients registered to kamailio. RegA calls RegB. Signalling itself for connection seems to work until the point RegA receives the 200Ok and wants to send the ACK, kamailio throws errors. Here are the 200Ok with the following ACK and the errors that kamailio writes: 200OK (from RegB to RegA, successful): SIP/2.0 200 OK Record-Route: <sip:zfHrSW4KTMuuxQYK+gVKH5MK+gUR1jc=@10.250.5.74:8083;transport=ws;lr=on;ftag=3bege11omg> Via: SIP/2.0/WSS 192.0.2.1;rport=54839;received=10.250.5.17;branch=z9hG4bK415866 To: <sip:dnagorny@r5dev-kam1.robot5.de>;tag=ecmmlvc4ee From: "Testaccount" <sip:4@r5dev-kam1.robot5.de>;tag=3bege11omg Call-ID: vu8rkkgki2uupuf2cqf4 CSeq: 2017 INVITE Contact: <sip:l1jevtog@192.0.2.59;alias=10.250.5.17~54838~6;alias=10.250.5.17~54838~6;transport=ws> Supported: ice,replaces,outbound Content-Type: application/sdp Content-Length: 1001 v=0 o=mozilla...THIS_IS_SDPARTA-47.0 4652302965311172452 0 IN IP4 0.0.0.0 s=- t=0 0 a=sendrecv a=fingerprint:sha-256 1A:88:2D:6E:9F:82:14:E0:A2:DB:A7:16:56:00:2C:25:1B:D2:D1:EE:B2:D5:ED:6F:E1:7C:5C:B4:1A:30:9C:85 a=group:BUNDLE audio a=ice-options:trickle a=msid-semantic:WMS * m=audio 30710 UDP/TLS/RTP/SAVPF 111 c=IN IP4 10.250.5.74 a=candidate:0 1 UDP 2122252543 10.250.5.17 52581 typ host a=candidate:1 1 UDP 1686052863 87.158.246.174 49807 typ srflx raddr 10.250.5.17 rport 52581 a=end-of-candidates a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=fmtp:111 maxplaybackrate=48000;stereo=1 a=ice-pwd:4d722747e7d3b3caa360ebe9e646fb44 a=ice-ufrag:2b7e23b8 a=mid:audio a=msid:{5cea7c38-cc3e-4bd2-9a3b-5d2381f0acf9} {0236e1cf-dfdc-4c94-8ce0-59f52a23ebb1} a=rtpmap:111 opus/48000/2 a=setup:active a=ssrc:565594393 cname:{7df4b0da-093c-4c2d-b154-0b860b43f575} a=sendrecv a=rtcp:30710 a=rtcp-mux a=candidate:Y2XhN5oQNoH6nRph 1 UDP 2122252799 10.250.5.74 30710 typ host ACK (from RegA to RegB, unsuccessfull): ACK sip:l1jevtog@192.0.2.59;alias=10.250.5.17~54838~6;transport=ws SIP/2.0 Route: <sip:zfHrSW4KTMuuxQYK+gVKH5MK+gUR1jc=@10.250.5.74:8083;transport=ws;lr=on;ftag=3bege11omg> Via: SIP/2.0/WSS 192.0.2.1;branch=z9hG4bK2821443 Max-Forwards: 69 To: <sip:dnagorny@r5dev-kam1.robot5.de>;tag=ecmmlvc4ee From: "Testaccount" <sip:4@r5dev-kam1.robot5.de>;tag=3bege11omg Call-ID: vu8rkkgki2uupuf2cqf4 CSeq: 2017 ACK Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: outbound User-Agent: JsSIP 1.0.1 Content-Length: 0 Thrown Errors: kamailio[5009]: INFO: <script>: Is natted kamailio[5009]: INFO: <script>: From: sip:4@r5dev-kam1.robot5.de To: sip:l1jevtog@192.0.2.59;alias=10.250.5.17~54838~6;transport=ws @ <null> Via: SIP/2.0/WSS 192.0.2.1;branch=z9hG4bK4386608 Route: <null> vu8rkkgki2uupuf2cqf4 kamailio[5009]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket kamailio[5009]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) I don't understand these errors. Can anybody explain and point me what needs to be fixed? Using Kamailio 4.3.5 on CentOS7.2, funny info if I call from outside(PSTN/FreeSWITCH) to kamailio into WebRTC-Client it works w/o problems. Respectfully Dimitry Nagorny Trainee

1 1

OpenSIPS balcklist/whitelist related question
by eyatsko 16 Jun '16

16 Jun '16

Hello! Could you, please, help me to make the above?.. I already have rough understanding about blst, permission modules etc. But I am about to compose CFG to count wrong (erroneous) attempts to REGISTER or INVITE from the same IP through several SIP dialogs. For example, some malicious user attempes to brute-force password. I would like to move it permanently into permissions' database after several (predefined) attempts. Kind regards, Ellad

1 0

uac.reg_refresh not removing records from memory
by DanB 16 Jun '16

16 Jun '16

Hey Guys, I was wondering if there is any dynamic way today to remove a single registration out of memory (and implicitly to force un-REGISTER) Eg: I add one record using uac.reg_refresh (which works fine) but after removing the record from the database and executing uac.reg_refresh again, the record stays in memory until expire. Is there a better way to enforce un-register on remote side? Thanks in advance! DanB

2 1

Question about testing RTPENGINE
by Dmitry 16 Jun '16

16 Jun '16

Hello I am trying to make a test setup to test the RTPENGINE. Case 1: Now my setup is Kazoo . i.e.: Subsriber A ---Asterisk (192.168.175.136)-----via NAT(10.111.101.3)-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) Case 2: When A calls to B - the sound both ways, so it is ok. Subsriber A ---Asterisk (10.20.20.101)-----via NAT(10.11.97.12)-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) WITH RTPENGINE: When A calls to B - the sound is absent, no sound. I compared both dumps the difference only in addressesAnd what I read in nathelper module: - 1 - Contact header field is searched for occurrence of RFC1918 addresses. - 2 - the "received" test is used: address in Via is compared against source IP address of signaling - 4 - Top Most VIA is searched for occurrence of RFC1918 addresses - 8 - SDP is searched for occurrence of RFC1918 addresses - 16 - test if the source port is different from the port in Via - 32 - test if the source IP address of signaling is a RFC1918 address All flags can be bitwise combined, the test returns true if any of the tests identified a NAT. in nat-traversal-role.cfgI see: if (nat_uac_test("3")) { force_rport(); fix_nated_contact(); } if (has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); } As I understand - both addresses are RFC1918but in case 2 the RTPENGINE is not kernalized Does the RTPENGINE distinguish the ip address? why it was not kernalized in case 2? Any help is appresiated.

1 1

RTPENGINE not kernalizing in some cases
by Dmitry 16 Jun '16

16 Jun '16

Hi I have the Kamailio and FS behind it (in Kazoo). Both are in one ip address 10.34.101.104 i use rtpengine_manage(). When I call from Asterisk which is behind NAT :1) Asterisk in VMware player (NAT, local ip 192.168.175.136) and public 10.34.101.103 - the voice was ok? in logs I saw Kernalizing media 2)Asterisk in KVM(NAT, local 10.28.40.101 public 10.101.29.45) and no Kernalizing in /var/log/messages and no voice. The Asterisk configs are absolutely the same. Kamailio log shows no errors In both cases the INVITE's SDP connection is 10.34.101.104 - the address of the Kamailioin the ' case 1' I see all RTP streams : 2 rtp streams - between asterisk and kamailio and vice versa, 2 -rtp streams between kamailio/rtpengine - Freeswitch and vice versa in the second case I see the following RTP streams: 1 rtp stream - from Kamailio to Asterisk, 1 rtp stream - from Freeswitch to Kamailio How to debug such cases? Any help is appreciated.

1 1

HTTP query at the end of a phone call
by Jay Li 15 Jun '16

15 Jun '16

Dear All, I'd like to send a HTTP call to a HTTP server whenever a call routed through a Kamailio server ends. By looking at the document, seems I should use http_query() function in the utils module. I wonder which route function I should put the http_query() in assuming we use the default kamailio.cfg. Thanks. Regards,Jay

4 3

404", "Not Found location"
by Sébastien Brice 15 Jun '16

15 Jun '16

Hi guys when i try to register i have route[LOCATION] that returns "404 Not Found location" due to this statement: $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found location"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } thats weird i did follow ubuntu realtime trying to get sip account registered and then uac_req_send to asterisk but the first step already failed. :-( i noticed there is no Username in my RURI (whatever UAC i am using) maybe this is the root of evil please find attache my kamailio.cfg i am using kamailio v 4.4 Any idea very welcome

2 1

Kamailio + ToS
by Marcelo Bonin 15 Jun '16

15 Jun '16

Hello, I need to know what are the available parameters to set ToS on Kamailio. I couldn't found much information about that, only this https://www.kamailio.org/wiki/cookbooks/4.0.x/core#tos but there is no reference for the available parameters. Thanks, Marcelo.

2 1

question about RTPengine
by Dmitry 14 Jun '16

14 Jun '16

hello i read the documentation about RTPengine.and the documentation says that: If INVITE with SDP, when the tm module is loaded, mark transaction with internal flag FL_SDP_BODY to know that the 1xx and 2xx are for rtpengine_answer() what does it mean? actually - according to 3261 RTP media (both ways) may be turned on after ACK

3 5

IP TOS
by Daniel W. Graham 14 Jun '16

14 Jun '16

Could I get some clarification on tos global setting as referenced by documentation below? Specifically, are DSCP values allowed? Or is this simply for scheduling purposes at the OS level? I can change the value once packets hit the network but if possible would like 0x60 to originate from Kamailio. tos The TOS (Type Of Service) to be used for the sent IP packages (both TCP and UDP). Example of usage: tos=IPTOS_LOWDELAY tos=0x10 tos=IPTOS_RELIABILITY Thanks! -Dan

2 3

Fwd: [Free-RTC] GSoC progress, more mentors?
by Daniel Pocock 14 Jun '16

14 Jun '16

This report was sent to the Free-RTC list https://lists.fsfe.org/mailman/listinfo/free-rtc Most of the projects have an emphasis on SIP If anybody wants to facilitate testing the new Telepathy/Empathy SIP connection manager code against Kamailio or other SIP proxies, registrars and presence servers that would be very welcome. -------- Forwarded Message -------- Subject: [Free-RTC] GSoC progress, more mentors? Date: Mon, 13 Jun 2016 22:53:46 +0200 From: Daniel Pocock <daniel(a)pocock.pro> Reply-To: Free real-time-communications discussion list <free-rtc(a)lists.fsfe.org> To: Free-RTC(a)lists.fsfe.org Hi all, The GSoC students have officially been working for 3 weeks now. I created a summary of all the students here: https://wiki.debian.org/SummerOfCode2016/RTCProjects They've been publishing weekly reports on the debian-outreach list: https://lists.debian.org/debian-outreach/ It would be really helpful to increase the size of the mentoring team for the students working on RTC projects: - we probably need at least one more Python developer and one more C++ developer to help - other people can also help test the projects, for example, if you are using Wordpress, please try Mesut's new WPCall module: https://github.com/mesutcang/wpcall As a bonus, some mentors are selected every year for a fully paid trip to California for Google's GSoC mentor summit. If anybody has more questions about it or would like to be more involved please contact me privately. Regards, Daniel _______________________________________________ Free-RTC mailing list Free-RTC(a)lists.fsfe.org https://lists.fsfe.org/mailman/listinfo/free-rtc

1 0

Kamilio as Scalable Registrar + Proxy
by Colin Morelli 13 Jun '16

13 Jun '16

Hello, I currently have Freeswitch acting as a B2BUA, handling registrations, routing, etc for a prototype voice application I have built. Now I'm at the point where it's time to actually scale it out, and I'm looking at a few different options for the SIP proxy + registration later. Essentially, what I would like to have (I think), is: 1) Stateless SIP proxy with consistent hashing to route to: 2) Transaction-stateful SIP proxy handling registrations and initial routing Ideally, the only thing that's stored across the cluster (if possible) would be registrations, in something like memcache/redis. Then, of course, each node might have its own internal state of transactions that it's tracking. I'd prefer the only thing to care about dialog state to be Freeswitch. Finally, the decisions about which FS instance to route to would be made by my application as well, based on runtime rules. Is this something people here have done before? Are there other suggestions on how to scale out kamailio to multiple instances. Thanks in advance, Colin

2 1

BLF lights
by Bill 13 Jun '16

13 Jun '16

I have spent a couple days beating my head against the Kamilio BLF feature. But cannot seem to get presence signals from one ua to another. I am testing with a Zoiper softphone and a Yealink hardware phone, both registering to Kamailio and able to make and receive calls. But getting absoultely nowhere with BLF. Have gone through all the suggestions in whitepaper 43, but still no results. Anybody had success with BLF, please steer me in the ritght direction. Thanks, Bill

2 1

Kamailio benchmark app_python + OPTIONS
by Anonim Stefan 13 Jun '16

13 Jun '16

Hi, I am doing some basic benchmarks with Kamailio. I am using a VM with 1 CPU i7 2,50 GHz and Kamailio with children=1. This is what I measure for each request, inside the benchmark block: - app_python processing (some euclidian distance based on a dlg_var input) -> same for all requests - uac_req_send() OPTIONS -> same for all requests I am running SIPp tests for 1000 calls with 1,10,50,100,150,200 calls per second. What I see is that the Global average processing time per request decreases with the increasing rate; I expected it to increase with the increasing rate. Is there an explanation for this, or am I doing something wrong ?! Thanks, Stefan

3 5

BYE message to an individual client
by kai.ohnacker＠cbc.de 13 Jun '16

13 Jun '16

Hello Community, is it possible to send a BYE message to an individual client via mi command line (siremis)? Cheers, Kai

2 1

TM Module t_suspend question
by gmele 10 Jun '16

10 Jun '16

Hello, we are using the TM Module* t_suspend *function in our kamailio to suspend transactions (mainly INVITE requests). We do this as we send some push notification to the callee that will force it to re-REGISTER. Once we get the REGISTER, we resume the transaction with a *t_continue()*. If the callee never re-REGISTER, the FR Timer will terminate the INVITE transaction. This is working fine for us when the callee has a phone that can receive the push notification. Now, we want to be able to contact users having multiple devices: phone and desktop VoIP clients. The problem with current implementation is that if we wait for the re-REGISTER from the mobile phone, we may never send the INVITE to the desktop VoIP client if the mobile phone is shutdown. So, I'd like to now if it is possible to set a timer to the t_suspend command which will automatically resume the suspended transaction (without issuing an ERROR like the FR timer is doing)? I didn't see such possibility in the TM module, but is there another way to implement this? Hope I was clear enough Regards Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/TM-Module-t-suspend-question-tp1493… Sent from the Users mailing list archive at Nabble.com.

3 6

Re: [SR-Users] Push notification question
by Vasiliy Ganchev 10 Jun '16

10 Jun '16

Hi! To cope with your specific cases try connection-oriented transport (tcp/tls). e.g. if app is closed - the connection is closed as well and the usrloc record will be removed. After this use standard approach (with late forking - tsilo module) imho, it is not a good idea to suspend the call if there is records in location. Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/Push-notification-question-tp149025… Sent from the Users mailing list archive at Nabble.com.

1 0

Handling ReINVITEs
by Grant Bagdasarian 09 Jun '16

09 Jun '16

Hello, How should reINVITEs be handled in the kamailio script and also in combination with rtpengine? I was reading the RR docs and loose_route function pretty much handles this if I'm correct. We're currently receiving reINVITEs for checking if a session is still active, not for session modifications like SDP changes. Does kamailio handle reINVITEs correctly out of the box or should extra logic be implemented? Also, what happens when rtpengine_manage is called for a reINVITE inside WITHNDLG or in onreply_route for 200 OK? Is the module smart enough to ignore port (de-)allocations when the SDP is still the same as the initial INVITE or 200 OK? Regards, Grant

2 2

forward calls from kamailio(RTPengine) to FREESWITCH
by Ankitt Sharma 09 Jun '16

09 Jun '16

hello i am using using kamailio + RTPengine in my setup and i want to forward every single call to freeswitch(for recording purpose), all three are running on the same machine (testing in the local network) my requirement for media flow is: UA1-->RTPproxy-->>freeswitch-->RTPproxy-->UA2 my user agents are using webrtc media profile so my rtp engine flags are : rtpengine_manage("trust-address replace-origin replace-session-connection RTP/SAVPF"); i've attached my configuration file i've configured freeswitch for media_webrtc=true ,still when i try calling the freeswitch send 488 without sending call to other end should media-address flag be used to forward media through, if yes how it should be done ??

2 1

ACK does not match transaction after stripping Route header
by Francisco Valentin Vinagrero 09 Jun '16

09 Jun '16

Hi there, I'm having an issue in a SBC (ACME) -> KAMAILIO -> Asterisk scenario with an ACK that gets ignored in Kamailio because it does not match any transaction. The INVITE coming from the SBC looks like this (only relevant headers and hidden numbers for simplicity - SBC has IP .12 , Kamailio .30 and Asterisk .34) INVITE sip:mynumber@10.15.1.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 To: <sip: mynumber@10.15.1.30:5060> From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 P-Asserted-Identity: <sip: a-number @10.15.1.12> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 INVITE Contact: <sip:41754112601@10.15.1.12:5060;transport=udp> And its forwarded to Asterisk with the Record-Route header: INVITE sip: mynumber @10.15.1.30:5060 SIP/2.0 *Record-Route: <sip:10.15.1.30;lr=on;ftag=SDkbo9901-42090> *Via: SIP/2.0/UDP 10.15.1.30;branch=z9hG4bK1c02.7dc1b94be22d8780df5141f9ba3c5b7b.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 To: <sip:mynumber@10.15.1.30:5060> From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 P-Asserted-Identity: <sip:a-number@10.15.1.12> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 INVITE Contact: <sip:a-number@10.15.1.12:5060;transport=udp> Then, 200 OK from Asterisk: SIP/2.0 200 OK *Via: SIP/2.0/UDP 10.15.1.30;rport=5060;received=10.15.1.30;branch=z9hG4bK1c02.7dc1b94be22d8780df5141f9ba3c5b7b.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 *Record-Route: <sip:10.15.1.30;lr;ftag=SDkbo9901-42090> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 To: <sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a CSeq: 1 INVITE Server: Asterisk PBX 13.8.0 Contact: <sip:10.15.1.34:5060> Which is sent to the SBC like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 *Record-Route: <sip:10.15.1.30;lr;ftag=SDkbo9901-42090> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 To: <sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a CSeq: 1 INVITE Server: Asterisk PBX 13.8.0 Contact: <sip:10.15.1.34:5060> And finally the SBC sends the ACK: ACK sip:10.15.1.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdt7p9k00dounet8ic600.1 To: <sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 ACK Contact: <sip:a-number@10.15.1.12:5060;transport=udp> *Route: sip:10.15.1.30;lr;ftag=SDkbo9901-42090 The problem: this ACK gets not retransmitted to Asterisk At first, I thought it was some sanity check but after disabling that I realized that it was in the WITHINDLG route. For the incoming ACK I get in the logs: Jun 8 11:56:47 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53240]: ALERT: <script>: Inside LOOSE route for ACK proto=UDP trans=4194304 from=sip:00754112601@10.15.1.12;user=phone route=sip:10.15.1.30;lr;ftag=SDkbo9901-42090 src_ip=10.15.1.12 And once the ACK is ready to be sent to Asterisk, the Route header has been removed and no Record-Route has been added so it fails. Jun 8 11:56:44 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53238]: INFO: rr [rr_mod.c:402]: pv_get_route_uri_f(): No route header present. Jun 8 11:56:44 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53238]: ALERT: <script>: ACK does not match transaction!! proto=UDP trans=4194304 from=sip:00754112601@10.15.1.12;user=phone route= src_ip=10.15.1.30 My WITHINDLG route looks like this: # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { xlog("L_ALERT","Inside LOOSE route\n"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { xlog("L_ALERT","Inside LOOSE route for ACK proto=$rP trans=$mf from=$fu route=$route_uri src_ip=$si \n"); # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_ALERT","ACK does not match transaction!! proto=$rP trans=$mf from=$fu route=$route_uri src_ip=$si \n"); exit; } } sl_send_reply("404","Not here"); } exit; } } Thanks for reading this :) Any idea about how to validate the transaction? t_check_trans is not being validated... Cheers, Francisco.

2 2

rtpproxy question
by Albert Petit 09 Jun '16

09 Jun '16

Hi list I have installed a newer version of rtpproxy (2.0) in our development server , i was happy because it brings lot of performance improvements However i have found an issue when clients use TLS+SRTP In that scenario i do not want rtpproxy to decrypt/reencrypt the traffic as my B2BUA does that or in some scenarios we do plan to add end to end encryption. RTPProxy 1.X was doing that perfectly Instead since i use rtpproxy 2.0 all SRTP and SRTCP traffic is decrypted by rtpproxy and i receive it clear in the B2BUA How can i disable this new feature and that rtpproxy just keeps sending the received SRTP/SRTCP stream without decrypting it?

3 3

IPv6 Only client-rtp proxy configuration
by aman jeff 09 Jun '16

09 Jun '16

Hi team, How to do the Configuration of Kamailio+RTP-Proxy for SER in IPv6, Currently I can establish SIP Session,while rtp start call are not audible.So Some configuration need to change,Am bit confused on it,Please help to resolve the issue,For my proper VoIP-Communication in IPv6 Surveillance. Regards Arnab

3 2

kamailio mysql+ldap+radius authentication
by wanmon 08 Jun '16

08 Jun '16

Dear list, I have Kamailio 4.4.1 running with local mysql authentication (WITH_MYSQL/WITH_AUTH). i have RADIUS and LDAP services running. I would like to use all of them (MYSQL+RADIUS+LDAP) for Authentication. How is the best "kamailio.cfg" approach to use all defined users on RADIUS and LDAP? so, when a user try to login, kamailio will try MYSQL, them LDAP, them RADIUS. If user/password are not define on any of those 3 system, authentication will fail. Do i need modify my current LDAP tree? Im trying to using standart "userPassword" attribute. This is my reference: http://www.kamailio.org/wiki/tutorials/mini-howto-admin/ldap-user-auth i would like to "add" route[LDAPAUTH], not to "replace" route [AUTH]. Kind Regards

2 1

A: README files for modules generated automatically
by Daniel-Constantin Mierla 08 Jun '16

08 Jun '16

Hello, following past discussions about different formats resulted when generating readme files for modules on various systems, which typically led to conflicts when backporting, I put in place a cron job running on kamailio.org to fetch from github the docbook files, generate the readme and push them back. The cron job is running every 15min. >From now on it is no longer needed to generate the readme yourself. Just commit the changes to xml files and push them to git repository. The readme will be refreshed automatically in a bit. The commits in this case are done by a generic git account named kamailio-dev. The commit message looks like: modules: readme files regenerated - app_mono ... Only the first module that has the readme updated is listed (in this example being app_mono). The readmes are automatically generated for master and 4.4 branches. If you do consistent changes to the docbook file, you can generate the readme locally to be sure there is no error, then restore the old version before committing - relevant commands: make modules-readme modules=modules/async rm modules/async/README git checkout modules/async/README Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

4.4 compiler warnings on debian jessie
by Juha Heinanen 08 Jun '16

08 Jun '16

I'll come back to mem/tlsf_malloc.c issue. Camille asked if the compiler warnings on mem/tlsf_malloc.c were occurring on x86 32 bit machine. I now tried and, yes, they do not appear on i386 jessie, but not on amd64 jessie. So looks like this is the same issue that Ovidiu has reported. -- Juha

2 3

4.4 fails to start on debian jessie due to tls issue
by Juha Heinanen 08 Jun '16

08 Jun '16

Was this fixed on master? Jun 8 08:12:46 sars /usr/bin/sip-proxy[19017]: : tls [tls_init.c:557]: init_tls_h(): ERROR: tls: init_tls_h: installed openssl library version is too different from the library the Kamailio tls module was compiled with: installed "OpenSSL 1.0.1t 3 May 2016" (0x1000114f), compiled "OpenSSL 1.0.1k 8 Jan 2015" (0x100010bf).#012 Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) Jun 8 08:12:46 sars /usr/bin/sip-proxy[19017]: CRITICAL: <core> [main.c:2514]: main(): could not initialize tls, exiting... If so, please backport to 4.4 since it is now impossible to start kamailio on debian jessie. -- Juha

2 6

ack with cscall-id header
by Rawad Khalil 07 Jun '16

07 Jun '16

Dear kamailio community, I have a problem that seems really weird. i have the same problem using kamailio 4.3 and 4.4.1. The scenario is the following: I make a call from a sip phone to kamailio. kamailio routes the Invite to an asterisk server. After a while asterisk sends a Bye request to kamailio. Now instead of routing the Bye to the sip phone, I want to create a re-invite request from kamailio and send it to the sip phone. The purpose of the re-invite is to connect the sip phone to another sip phone that already has an active call. This part is working fine but the problem occurs after the sip phone sends the ‘200 OK’ message for the re-invite. Kamailio sends an ACK for the 200 message but this ACK has a ‘CSCall-id’ header instead of the ‘Call-id’ header. To create the re-invite I am using the UAC module with one change to the source code. In the uac.c file found in the tm module folder, I added 2 lines of code to the request function in order to stop the UAC module from creating a from tag(attached). I also attached a trace that shows the ack message with cscall-id. Thank you, Rawad

2 1

Re: [SR-Users] Regarding kamailio-4.4.0 msrp module
by Daniel-Constantin Mierla 07 Jun '16

07 Jun '16

Hello, what is missing? What is highlighted seems to be in the code. Can you provide a patch or make a pull request on github project? That will make it simpler to understand what you want to change. Cheers, Daniel On 07/06/16 11:51, Surendra.P wrote: > > Hello Daniel, > > > > Switch case control is missing while parsing MSRP > pseudo variables. Please find the code snippet below: > > > > int pv_parse_msrp_name(pv_spec_t *sp, str *in) > > { > > if(sp==NULL || in==NULL || in->len<=0) > > return -1; > > > > switch(in->len) > > { > > case 3: > > if(strncmp(in->s, "buf", 3)==0) > > sp->pvp.pvn.u.isname.name.n = 1; > > else goto error; > > break; > > case 4: > > if(strncmp(in->s, "body", 4)==0) > > sp->pvp.pvn.u.isname.name.n = 2; > > else if(strncmp(in->s, "code", 4)==0) > > sp->pvp.pvn.u.isname.name.n = 3; > > else if(strncmp(in->s, "hdrs", 4)==0) > > sp->pvp.pvn.u.isname.name.n = 4; > > else goto error; > > break; > > case 5: > > if(strncmp(in->s, "msgid", 5)==0) > > sp->pvp.pvn.u.isname.name.n = 5; > > else if(strncmp(in->s, "conid", 5)==0) > > sp->pvp.pvn.u.isname.name.n = 21; > > else goto error; > > case 6: > > if(strncmp(in->s, "method", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 6; > > else if(strncmp(in->s, "buflen", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 7; > > else if(strncmp(in->s, "sessid", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 8; > > else if(strncmp(in->s, "reason", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 9; > > else if(strncmp(in->s, "crthop", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 12; > > else goto error; > > break; > > > > > > Regards > > Surendra > > > > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Regarding kamailio-4.4.0 msrp module
by Surendra.P 07 Jun '16

07 Jun '16

Hello Daniel, Switch case control is missing while parsing MSRP pseudo variables. Please find the code snippet below: int pv_parse_msrp_name(pv_spec_t *sp, str *in) { if(sp==NULL || in==NULL || in->len<=0) return -1; switch(in->len) { case 3: if(strncmp(in->s, "buf", 3)==0) sp->pvp.pvn.u.isname.name.n = 1; else goto error; break; case 4: if(strncmp(in->s, "body", 4)==0) sp->pvp.pvn.u.isname.name.n = 2; else if(strncmp(in->s, "code", 4)==0) sp->pvp.pvn.u.isname.name.n = 3; else if(strncmp(in->s, "hdrs", 4)==0) sp->pvp.pvn.u.isname.name.n = 4; else goto error; break; case 5: if(strncmp(in->s, "msgid", 5)==0) sp->pvp.pvn.u.isname.name.n = 5; else if(strncmp(in->s, "conid", 5)==0) sp->pvp.pvn.u.isname.name.n = 21; else goto error; case 6: if(strncmp(in->s, "method", 6)==0) sp->pvp.pvn.u.isname.name.n = 6; else if(strncmp(in->s, "buflen", 6)==0) sp->pvp.pvn.u.isname.name.n = 7; else if(strncmp(in->s, "sessid", 6)==0) sp->pvp.pvn.u.isname.name.n = 8; else if(strncmp(in->s, "reason", 6)==0) sp->pvp.pvn.u.isname.name.n = 9; else if(strncmp(in->s, "crthop", 6)==0) sp->pvp.pvn.u.isname.name.n = 12; else goto error; break; Regards Surendra

1 0

IPv6 Only client-rtp proxy configuration
by aman jeff 07 Jun '16

07 Jun '16

Hi Team, The SIP packets are fine ,we are able to establish SIP session , As long the RTP Proxy not get involved and we are not able to hear audio,So How to do the proper configuration RTP Proxy,what to do in RTP Proxy to get the voice for the connected sip session Regards Admin Arnab

1 0

IPv6 Only client-rtp proxy configuration
by aman jeff 07 Jun '16

07 Jun '16

Hi Team, How to configure rtp proxy for IPv6 Only client,now we are not able to get the audio, Please help Regards Arnab

2 1

Kamailio Sorce code Documents
by aman jeff 07 Jun '16

07 Jun '16

Hi team, I like to know the source code documention of kamailio, I like to customize textops.c,like to get the variables and structure documents gparam_p, etc.. Regards Jeff

2 1

Yealink BLF
by Bill 06 Jun '16

06 Jun '16

Has anyone been successful in getting Yealink BLF's to work with Kamailio presence It appears that the xml format being sent is not compatible with Yealink, butr I can't find how to change the format of the body. Bill

2 1

RTPENGINE not kernalizing in some cases
by Dmitry 06 Jun '16

06 Jun '16

is it possible that RTPENGINE failes because 2 rtpengine_offer (2 invites during autharization) and the 183 Reply as I see in the log - is seen as an ANSWER Both INVITES are with SDP RTPENGINE_MANAGEThis call is not answered Received command 'offer' from 127.0.0.1:54490 Creating new call] offer time = 0.000134 sec Replying to 'offer' from 127.0.0.1:54490 Received command 'offer' from 127.0.0.1:53879 offer time = 0.000060 secReplying to 'offer' from 127.0.0.1:53879 Received command 'answer' from 127.0.0.1:40099 answer time = 0.000169 sec Replying to 'anser' from 127.0.0.1:40099 ICE negotiated: local interface 10.34.101.104Confirmed peer address as 10.34.101.104:7458Confirmed peer address as 10.34.101.103:62608

1 0

IPv6 Only client-rtp proxy configuration
by aman jeff 06 Jun '16

06 Jun '16

Hi Team, The SIP packets are fine ,we are able to establish SIP session , As long the RTP Proxy not get involved and we are not able to hear audio,So How to do the proper configuration RTP Proxy,what to do in RTP Proxy to get the voice for the connected sip session Regards Arnab

1 1

Wrong Publish status
by Aleš Šturm 06 Jun '16

06 Jun '16

Hello, Kamailio 4.3: I am testing Presence module with more complex scenario. I have 2 SIP clients (presentity) with equal identities (3918) and one watcher (2025). All my clients are Monster 0.9.25. My Kamailio Presence module is configured with "retrieve_order=1" (presentity records are retrieve by priority order). I have situation when wrong status is send to the watcher (2025). 1. First all three clients registers and publish its own status. Below is print from my MySQL database (presentity table) mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84481 | 3918 | 40 | 1464340695 | 1464337095 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84480 | 3918 | 40 | 1464340687 | 1464337087 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 2. Then first 3918 client publishes his mood >11111<. The status in >Presentity< table is refreshed under the id=84480. NOTIFY is sent to 2025 with status with mood >11111<. mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84480 | 3918 | 40 | 1464340703 | 1464337103 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>11111</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84481 | 3918 | 40 | 1464340695 | 1464337095 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 3. After that the second 3918 client publishes his mood >22222<. The status in >Presentity< table is refreshed under the right id=84481. NOTIFY is sent to 2025 with status with mood >22222<. mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84481 | 3918 | 40 | 1464340716 | 1464337116 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>22222</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84480 | 3918 | 40 | 1464340703 | 1464337103 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>11111</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 4. Then finally, the first 3918 changes his mood to "33333". The status in >Presentity< table is refreshed under the right id=84480. BUT, to 2025 is sent NOTIFY with mood "22222". "received_time" timestamp of status with mood "33333" is higher than timestamp of status with mood "22222". BUT the "id" attribute of "presentity" table of status with mood "33333" is smaller. I think that statuses with equal "priority" are retrieved by "id" and not by "received_time"!? mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84480 | 3918 | 40 | 1464340729 | 1464337129 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>33333</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84481 | 3918 | 40 | 1464340716 | 1464337116 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>22222</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | The Wireshark trace is attached. Can anyone help. Please? We have a big installation in the Gazprom's (Rusia) network, where Kamailio is used as a Presence server only. Thanks Ales

2 8

Yealink BLF
by Bill 05 Jun '16

05 Jun '16

Has anyone been successful in getting Yealink BLF's to work with Kamailio presence It appears that the xml format being sent is not compatible with Yealink, butr I can't find how to change the format of the body. Bill

1 0

rtpproxy ipv6 problem
by aman jeff 03 Jun '16

03 Jun '16

Hi Team, The SIP packets are fine ,we are able to establish SIP session , As long the RTP Proxy not get involved and we are not able to hear audio,So How to do the proper configuration RTP Proxy,what to do in RTP Proxy to get the voice for the connected sip session Regards Arnab

1 0

Kamailio 4.4.1 REGISTER Parse Error in PCSCF
by Mihail Dakov 02 Jun '16

02 Jun '16

Hi Folks, I have the strangest problem just after installing latest stable version. I had tried both ways using the repo (http://deb.kamailio.org/kamailio44) and downloading code with local compile (https://www.kamailio.org/w/2016/05/kamailio-v4-4-1-released/) The initial scenario is to get M2M with commercial IP and subsequently use the Rx interface to do the same. We are blocked because we cannot even achieve clean register for the UEs. UE1(galaxy s4 zoiper) -------- Kamailio IMS aio | UE2(galaxy s4 zoiper) -------- | What happens is that the first register message triggers some quasi-endless loop in the PCSCF. Hence it cannot be forwarded to SCSCF, ICSCF and HSS. But this doesn't always happen and between all the parse problems there are some register messages which get passed to the scscf icscf and hss. Somehow the parser feeds itself from the previous parse run and thus loops quite intensively. The message I get is SIP/2.0 400 CSeq method does not match request method. I must say that reiterating the registration process sometimes i get the register and sometimes i don't. Has anyone encounter similar issue for similar use case and can share some thoughs about a fix? I attached relevant log file, pcaps and configuration files. Let me know if more config files are needed.

3 15

Nonce extract from SIP Meesage
by Sergio Serrano 02 Jun '16

02 Jun '16

Hi all, I just try integrate Kamailio with external software and We want taht kamailio doesn't authenticate users. I want that the autehtication process would be done by external software. I want To receive a REGISTER for a user. Then I send a 407 and then when I receive all paramters for autehtincation(nonce, response, digestURI, etc, etc etc) send that data to external software and receive if data is correct, then sent 200 OK and save location in external software. Bu t I can see any pseudovariable or enything else to obtain that dato to provide to external software. Anyone could think that this process could be done? Regards, Sergio

3 2

NSQ libevbuffsock dependency
by smititelu 02 Jun '16

02 Jun '16

Hi, Upon trying to compile latest Kamailio from sources I got an: " nsq.h:9:24: fatal error: evbuffsock.h: No such file or directory #include <evbuffsock.h> ^ " Looking on the new NSQ module documentation I see the libevbuffsock dependency. Is there other recommended way, to install the library besides compiling from sources at [1]? Thx, Stefan [1] https://github.com/mreiferson/libevbuffsock

2 2

sip trace
by Jeremy Betts 01 Jun '16

01 Jun '16

I have a new carrier that wants sip traces of certain scenarios. They also require IPSEC tunnelling for all SIP messages. So when I use tcpdump or any other similar tool I can see the incoming SIP messages but outgoing SIP messages are ESP encoded. I've played with the siptace module but can't seem to get it to log replies. Does anyone have any solutions?

2 2

Dynamic advertised ip for rtpengine/rtpproxy
by Serge Yuriev 01 Jun '16

01 Jun '16

Hi, We have several exits from our network and all of them natted. Could we set adv ip for rtpproxy/rtpengine from db in realtime or we must define them on daemon run time only? -- Wbr, Serge via mobile -- Serge via mobile

4 3

Send Register from Kamailio
by Grant Bagdasarian 01 Jun '16

01 Jun '16

Hello, Which module do I need to have Kamailio send REGISTER messages to our carrier? Is it the UAC module or REGISTRAR module? It just needs to register with our carrier in order to receive inbound calls. Regards, Grant

2 4

advertise xxx.xxx.xxx.xxx:5060 is weird
by Sébastien Brice 01 Jun '16

01 Jun '16

Hi guys, I have a local proxy running on private ip 192.168.3.97 and advertising the wan ip of my box (THEPUBLICIPOFMYBOX) this local Kamailio is a stateful proxy that TM REGISTERS and INVITES to a Kamailio registrar(PUBLICIPKAMAILIOREGISTRAR) (please have a look at the kamailio.cfg attached to this email.) I want to force the signalling and sessions through this local proxy (some kind of outbound proxy) I have jitsi and xlite behind (ip 192.168.3.39 and 192.168.3.16) that tries to call themselves. user agents keep retransmitting 200 OK because they never get the ACK following the 200 OK pickup. If i remove the advertise THEPUBLICIPOFMYBOX of the kamailio.cfg the three handshake is working ok and the call is no longer dropped after a few second! i have a true bidirectionnal session and that's working ok. The bad thing is RTP flows straight through router (thanks to rtpproxy i guess) but i want incoming and outgoing RTP traffic to go on central local node 192.168.3.97 How can i achieve that? thanks. PS: i am not using path header because i am not skilled enough and kamailio is really hard horse to train

2 5

DHCP renewal conflicts with kamailio?
by Joonas Harjumäki 01 Jun '16

01 Jun '16

Hi! The attachment contains a log trace of kamailio 3.3.2. There is some errors in the log and I wonder if it is normal operation? The errors appeared in the middle of DHCP lease renewal and I wonder if the configuration (of dhclient/kamailio) could be improved to prevent such errors? Or to make sure kamailio retries udp_send sufficiently many times. The kamailio operated normally before renewal procedure began and after the procedure finished. Joonas Harjumäki

2 1

how to set kamctl to find ctl anf fifo that are not at /var/run
by Uri Shacked 01 Jun '16

01 Jun '16

Hi, I am using centos 7. By default on centos 7, the var/run file system is tmpfs. So, when kamailio runs with user and group other than root, it can not create the fifo file. I set the kamailio service and config files to work on usr/local/var/run/kamailio. Now, how do i set the kamctl to work with the fifo on /usr/local/var/run/kamailio ? Uri

3 3

Want dialog.default_timeout NOT to reset on subseq. req
by Andrey Utkin 01 Jun '16

01 Jun '16

Hi! I'd like to restrict max call duration. Currently we have something that partially works: dialog.default_timeout. But, as described in "How it works" section http://www.kamailio.org/docs/modules/devel/modules/dialog.html#idp2029360 "The dialog timeout is reset each time a sequential request is processed." So this option works for us when the useragents don't send sequential UPDATEs or reINVITEs. I am not sure if it is adequate to ask for this feature in dialog module. I am aware of possibility to terminate any call by management command, and of cnxcc module which has call duration control. I would love to find something that simple - a single option for max call duration, without external mechanism (cronjob etc.) or full-blown billing. Any suggestions are very welcome. Thanks in advance. Please reply to all recipients, I am not subscribed to this maillist.

2 1

513 Message too big for VIDEO calls
by Varghese Paul 31 May '16

31 May '16

Hi all, I am trying to setup WSS on opensips-2.2. I am getting "513 Message too big " when i am trying to make video calls. Audio calls and Registration is working fine.I tried to change the message_chunk_size in WSS module but still i am getting the message. Client : SIPML5 and JSSIP Opensip version : 2.2 I am attaching the sipml5 console logs. Please let me know is there any settings i can apply to resole this. Thanks in Advance Varghese Paul

2 1

Extending has_body() with additional mime types
by Giacomo Vacca 31 May '16

31 May '16

Hi all, apologies if this is an old topic but I've noticed that has_body() can provide false positives and would like to discuss a solution. As a practical example, has_body("application/vq-rtcpxr") was returning 1 for Content-Type "application/dialog-info+xml". The interpretation is that they are both unknown and so they match. Using 'if ($cT == "application/vq-rtcpxr)" provides a work around, but I wonder what's the suggested solution for this? a. Add the missing mime types (from RFC 6035 and 4235 respectively) to the list b. Instead of returning 1, return another code, e.g. 2, when the mime type is unknown c. Switch to string comparison when the mime type is unknown d. Something else? Thanks, Giacomo

2 1

Accept INVITE with transport=tcp
by Nahum Nir 31 May '16

31 May '16

Hi, I have a client that sends INVITE with transport=tcp. How can I make Kamailio to work with that? Thanks and regards, Nahum

2 1

Issue with LCR an failover routers
by Alberto Sagredo 30 May '16

30 May '16

Hi Im using LCR module and trying to handle 503/408 errors on provider to route to a second one. Actually im using if(!load_gws(1, $rU, $var(caller_uri))) { xlog("Gateways no disponibles"); sl_send_reply("500", "Server Internal Error - No gateway"); exit; } else { xlog("GW Seleccionado '$avp(i:709)'\n"); xlog("Dominio de destino: $dd\n"); xlog("To URI: $tu\n"); } if(!next_gw()) { xlog("No hay GW de Backup"); sl_send_reply("503", "Service not available - No gateway"); exit; } else { xlog("Enviado por primer GW\n"); xlog("ruri_user_avp: '$avp(i:500)'\n"); xlog("To URI after next_gw: $tu\n"); xlog("Request URI: $rU\n"); t_on_failure("FAILBACK"); } failure_route[FAILBACK] { if (t_is_canceled()) { exit; } if (t_check_status("408|50[34]")) { route(LCR_OUT_OP2); } } And using for OP2 same routing block with id=3... This second providers use AUTH so using: t_on_failure("TRUNKAUTH"); failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { if (to_gw("3")) { xlog("Autenticación "); $avp(auser) = "user"; $avp(apass) = "pass"; } uac_auth(); t_relay(); exit; } But getting this error: May 30 09:59:05 kamailio /usr/local/sbin/kamailio[21602]: ERROR: tm [t_fwd.c:1704]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding May 30 09:59:05 kamailio /usr/local/sbin/kamailio[21602]: ERROR: tm [tm.c:1466]: _w_t_relay_to(): ERROR: w_t_relay_to: t_relay_to failed Any idea how to resolve that?

2 2

converting integer to string
by Henry Fernandes 30 May '16

30 May '16

In Kamailio 4.2, I’m having problems converting an AVP integer value into a string. My Kamailio config stores the return code from auth_check() and outputs it. $avp(auth_code) = auth_check("$fd", "subscriber", "1"); xlog("L_NOTICE", “response = $avp(auth_code)”); When I check the AVP against an integer, it behaves normally. However, I cannot convert it to a string. The return value -3 (invalid user) gets converted to the string “18446744073709551613” when outputting it. My hunch is that Kamailio is treating the AVP as an unsigned integer (rather than a signed integer) when it converts it to a string. If I try this with an AVP with a positive integer, it works as expected. Can someone tell me how to convert a negative integer into a string in Kamailio? -H

3 3

Variable in dialog event routes
by Minh Phan 30 May '16

30 May '16

Hi List, I'm trying to have an external application to write CDR to a database by doing this: event_route[dialog:start] { evapi_relay("{'cdr_id':* $dlg_var(cdr_id),* 'event': 'answered'}"); } event_route[dialog:end] { evapi_relay("{'cdr_id': *$dlg_var(cdr_id)*, 'event': 'ended'}"); } event_route[dialog:failed] { evapi_relay("{'cdr_id': *$dlg_var(cdr_id)*, 'event': 'failed'}"); } $dlg_var(cdr_id) is set inside a request route. However, in these event routes, its value is always null. Its seems that $dlg_var variables are not available in those event routes. What are the variables available in dialog event route? Any suggestion on how to achieve what I'm trying to do? Best regards, Minh

2 3

http_async_client segfault
by Jason Penton 30 May '16

30 May '16

Hi, Has anyone experienced segfault using http_async_query from http_async_client module? Here is BT: #0 __GI_fgets_unlocked (buf=0x7fca23ffe660 "127.0.0.1", n=n@entry=1032, fp=0x7fca23ffe660) at iofgets_u.c:53 53 iofgets_u.c: No such file or directory. (gdb) bt #0 __GI_fgets_unlocked (buf=0x7fca23ffe660 "127.0.0.1", n=n@entry=1032, fp=0x7fca23ffe660) at iofgets_u.c:53 #1 0x00007fca68f08afa in get_contents (stream=<optimized out>, len=1032, linebuf=0x7fca23ffe660 "127.0.0.1") at nss_files/files-XXX.c:201 #2 internal_getent (result=0x7fca23ffe5c0, buffer=buffer@entry=0x7fca23ffe640 "\177", buflen=buflen@entry=1064, errnop=errnop@entry=0x7fca23ffebb0, herrnop=herrnop@entry=0x7fca23ffec00, af=af@entry=0, flags=flags@entry=0) at nss_files/files-XXX.c:246 #3 0x00007fca68f09ca3 in _nss_files_gethostbyname4_r (name=name@entry=0x7fca2e790fa0 "localhost", pat=pat@entry=0x7fca23ffebe0, buffer=buffer@entry=0x7fca23ffe640 "\177", buflen=buflen@entry=1064, errnop=errnop@entry=0x7fca23ffebb0, herrnop=herrnop@entry=0x7fca23ffec00, ttlp=ttlp@entry=0x0) at nss_files/files-hosts.c:402 #4 0x00007fca79eb7757 in gaih_inet (name=<optimized out>, name@entry=0x7fca2e790fa0 "localhost", service=<optimized out>, req=req@entry=0x7fca341c7490, pai=pai@entry=0x7fca23ffed10, naddrs=naddrs@entry=0x7fca23ffed00) at ../sysdeps/posix/getaddrinfo.c:850 #5 0x00007fca79ebae0d in __GI_getaddrinfo (name=0x7fca2e790fa0 "localhost", service=0x7fca23ffeec0 "18000", hints=0x7fca341c7490, pai=0x7fca23ffee78) at ../sysdeps/posix/getaddrinfo.c:2406 #6 0x00007fca6e0f80f4 in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #7 0x00007fca6e104fa4 in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #8 0x00007fca6e10284b in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #9 0x00007fca723c9182 in start_thread (arg=0x7fca23fff700) at pthread_create.c:312 #10 0x00007fca79ee547d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111 Camille, are you maintaining this module now?

2 2

Multiple async operations in single transaction fail (using http_async_client)
by Thom Seddon 30 May '16

30 May '16

Hi All, I've been doing some work this week with the new http_async_client module, it seems that with it's default behaviour it's not possible to execute another async query in the block immediately following the response, for example the following config: request_route { if (!t_newtran()) { sl_reply_error(); exit(); } http_async_query("http://localhost:8090/req_1", '{"user":"$fU"}', "REQ_1"); } route[REQ_1] { xlog("L_INFO", "route[REQ_1]: INIT\n"); if ($http_ok) { xlog("L_INFO", "route[REQ_1]: status $http_rs\n"); xlog("L_INFO", "route[REQ_1]: Sending REQ_2\n"); http_async_query("http://localhost:8090/req_2", '{"user":"$fU"}', "REQ_2"); } else { xlog("L_INFO", "route[REQ_1]: error $http_err)\n"); } } route[REQ_2] { xlog("L_INFO", "route[REQ_2]: INIT\n"); if ($http_ok) { xlog("L_INFO", "route[REQ_2]: status $http_rs\n"); } else { xlog("L_INFO", "route[REQ_2]: error $http_err)\n"); } } Generates the following error: 10(11832) INFO: <script>: route[REQ_1]: INIT 10(11832) INFO: <script>: route[REQ_1]: status 200 10(11832) INFO: <script>: route[REQ_1]: Sending REQ_2 10(11832) WARNING: tm [t_suspend.c:186]: t_continue(): transaction is not suspended [53195:560306836] Looking at the source, http_async_client calls t_suspend, then goes away and performs the request in the background. On response it calls t_continue and then this block <https://github.com/kamailio/kamailio/blob/4.4/modules/tm/t_suspend.c#L270-L…> in t_continue will continue execution, the second query will cause http_async_client to call t_suspend whilst sill executing from the previous t_continue. Once they query is sent, the first t_continue will complete, removing the T_ASYNC_SUSPENDED flag. When the second query returns it doesn't find the flag and so fails as shown. I'm far from an expect on this so would really appreciate some pointers, it seems to fundamental to be a race condition in tm but the usage in http_async_client seems to match usage in other modules so I'm not sure on the best tactic. I've tried peppering append_branch and async_route call's in various places to no avail. Full debug log here: http://pastebin.com/raw/rFmm0fmf Any help appreciated, *Thom Seddon*

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