- sr-users - kamailio.org

Processing of Invite Packet
by aawaise 27 Jun '14

27 Jun '14

When an invite packet is recieved by kamailio server, what processing is done on it ? Which modules and functions are called in its processing and in which sequence ? Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Processing-of-Invite-Packet-tp12867… Sent from the Users mailing list archive at Nabble.com.

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Transaction Stateful Message Handling
by AliReza Khoshgoftar Monfared 26 Jun '14

26 Jun '14

Hi, I am trying to configure a SIP proxy architecture that is stateful with regard to transaction but not the dialogues, i.e. If an INVITE takes a specific route, the corresponding 180 Ringing, 200 OK and ACK shall traverse the same set of proxies, but the corresponding BYE can take another route (and obv it's own 200 OK takes the same route again) Which modules can help me with that? In specific which functions / modules are recommended to handle requests and responses? Currently, I use "t_relay()" from TM module to forward requests and "loose_route()" from RR module to handle replies, but it looks like that if a BYE ends in a proxy which its corresponding INVITE has not traversed, it is not forwarded by that proxy. Am I making a mistake in my routing? Isn't there a simple way to do transaction stateful routing in Kamailio? Thanks, Alireza

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How to stop kamailio from send 408?
by Allen Zhang 26 Jun '14

26 Jun '14

Hi, when t_relay() times out, calling sl_reply_error() will send 408 to the client. But if I don't want to send 408, how do I interpret the error code t_relay() set? Regards, Allen

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How to find whether or not certain header exists?
by Allen Zhang 26 Jun '14

26 Jun '14

Hi, If I normally receive a INVITE that has a header 'aaa' with value 'bbb', but it's possible that some INVITEs don't have this header. How do I determine in the config file if the header exists? To check the value of the header I do if($hdr(aaa) == "bbb"). But what if the header doesn't exist? Cheer, Allen

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Kamailio Freepbx Integration Dropping Calls
by Carlos Rangel 26 Jun '14

26 Jun '14

Hello I have successfully (I believe) implemented Kamailio 4.1.4 integration with Freepbx 5.2.11 taking as a guide Daniel's tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. I just did not create the voicemail tables because voice mail is handled by Freepbx. I installed the system in a separate box for testing and connected to the Freepbx Production server via IAX trunk. The system is behind a Cisco Firewall and looks like this Remote User Internet Internal network Cisco 7960 ----DSL router ---|------Internet --------|-----Cisco ASA 5500 FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production Server --------|------ PSTN I have configured the FW to allow UDP and TCP traffic from the corresponding IP as well as tfpt that is needed for the Ciscos to pick up the configuration from the server. I have a few remotes Cisco 7960 phones that can register remotely in Kamailio as long as the user is added with kamctl add user password and as long as the extension is created in Freepbx. The problem that I have is when try to make a call from the remote Ciscos the call is dropped after 30 or 40 seconds. I can see from the logs that the problem appears to be that the server is not receiving responses from the phone 06-25 10:57:30] WARNING[1814] chan_sip.c: Retransmission timeout reached on transmission 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32001ms with no response [2014-06-25 10:57:30] WARNING[1814] chan_sip.c: Hanging up call 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Is this something that we can adjust in kamailio or could it be related to the FW configuration?? Sorry but I am very new to kamailio and sip. Thanks Carlos

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Trouble with rtpproxy
by Igor Potjevlesch 26 Jun '14

26 Jun '14

Hello, For a particular gateway, I force the RTPProxy. This work fine but on the other hand, in my side the RTPPRoxy response to the source IP address of SIP packet instead of the IP address in "Connection information". Is there a way to instruct the use of the IP address announce in the SDP instead of the last IP source of the last SIP packet? Regards, Igor.

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segfault at 18 error 4 in auth_db.so on Version 4.0.6-1 Release
by Jonathan Hunter 26 Jun '14

26 Jun '14

Hi Guys, We had our kamailio instance crash today, it appeared to be after multiple registration failures, the resulting logs showing; Jun 25 15:47:55 kernel: : kamailio[14993]: segfault at 18 ip 00007f03102f9609 sp 00007fff1b346500 error 4 in auth_db.so[7f03102f1000+b000]Jun 25 15:47:55 /usr/sbin/kamailio[14989]: ALERT: <core> [main.c:788]: handle_sigs(): child process 14993 exited by a signal 11Jun 25 15:47:55 /usr/sbin/kamailio[14989]: ALERT: <core> [main.c:791]: handle_sigs(): core was not generatedJun 25 15:47:55 /usr/sbin/kamailio[14989]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLD Any comments welcome. thanks Jon

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Syntax help in comparison of values of variables
by aawaise 26 Jun '14

26 Jun '14

I need some help regarding syntax. Can we put some comparisons on the values of these variables like if ("$ru"=="$fu") {xlog("both are same")}; If this is not the way then what is the way to do so ?? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Syntax-help-in-comparison-of-values… Sent from the Users mailing list archive at Nabble.com.

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Extract information from invite packet
by aawaise 26 Jun '14

26 Jun '14

I want to log some information of callee from incoming invite packet. how can it be done. I want to get it logged in /var/log/messages. Any help will be appreciated. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Extract-information-from-invite-pac… Sent from the Users mailing list archive at Nabble.com.

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Passing Command-Line Arguments to Kamailio Config Script
by AliReza Khoshgoftar Monfared 25 Jun '14

25 Jun '14

Hi, I had a simple question to which I could not find a clear answer in the documentation. How can I pass a command line argument to Kamailio config script? For example, assume that I run my kamailio daemon using script myconf.cfg: >>kamailio -f myconf.cfg Now I want to pass a value, say an IP address and use it inside the script. Sth like >>kamailio -f myconf.cfg --ip __MY_IP__ and then have an assignment of "$var=__MY_IP__" inside myconf.cfg Is there an easy way to do this?

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Re: [SR-Users] IMS and SIP over WebSocket 500/Error
by DURECU, OLIVIER (OLIVIER) 25 Jun '14

25 Jun '14

Hi Carsten, I had a look Kamailio repo and did not find the configuration you proposed to post. Did I look at the bad place or didn't you find the time to do it ? Thanks, Olivier Hi Olivier, i will update the configuration files for WebSockets at the P-CSCF after KamailioWorld. The week after KamailioWorld, we are at another congress, but probably around easter, you should find a proper configuration for WebSocket on the P-CSCF (including media). Kind regards, Carsten 2014-04-01 12:02 GMT+02:00 DURECU, OLIVIER (OLIVIER) <olivier.durecu at alcatel-lucent.com>: > Hello all, > > I have installed Kamailio 4.1.2 in IMS configuration (3 Kamailio instances > to run in P/I/S-CSCF modes) > > Everything works fine with Native IMS clients (Boghe) > > I want now to introduce Web clients using WebSockets. I have added to the > P-CSCF Kamailio.cfg the websockets parts > Registration is Ok but I can't manage placing a call. > > With SIPoWS, the contact header is sip:olivier at 132.invalid. So when the > P-CSCF receives a terminating INVITE, its destination URI is > sip:olivier at 132.invalid , it tries to make a DNS resolution of 312.invalid > and it doesn't forward to the client saying the domain is unknown and can't > be routed > > I guess that something is missing in my configuration around > aliases/nathelper/outbound but didn't find what... > > Thanks for any help > > > Olivier DURECU > Bell Labs / IP Platform Research > Villarceaux Center > Tel: +33(0) 130772759 > GSM: +33(0) 683776482 > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users at lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

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Setting priority (q) for kamctl ul add
by David Wilson 25 Jun '14

25 Jun '14

Thanks Daniel, Rather than changing the kamctl script, I can just run a command like: kamctl mi ul_add location <user> <uri> <expires> <q> 0 0 0 0 When I run this and set expires to 0 (as per the comment in kamctl, "expires 0 means persistent contact"), kamctl ul show shows expires=-1403662437 which siremis shows as 1970-01-01 08:00:00. Within 20 minutes, the entry is gone and I'm not sure why. Is the 8:00 (my timezone is UTC+8) causing a check for zero to fail or is there something else I'm missing? For reference, I'm using db_mode=2 for usrloc. Regards, Dave. On 20 Jun 2014, at 6:00 pm, sr-users-request(a)lists.sip-router.org wrote: > Date: Fri, 20 Jun 2014 07:38:50 +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] Setting priority (q) for kamctl ul add > > Hello, > > On 20/06/14 06:16, David Wilson wrote: >> Hi All, >> >> I'm trying to add a permanent usrloc entry via kamctl ul add. >> >> This works, but the created entry has a q value of 1.0 which is higher than I need. >> >> Is there a way to either: >> >> 1. Specify a q value when using kamctl ul add, or > apparently the Q is hardcoded, a way to fix it is to edit kamctl (which > is a shell script), search for 'ul_add' and update the 1.0 value in that > command. > >> 2. Edit the q value of an existing record by using a kamctl command. >> > Maybe removing and add it it with same other attributes... > > Cheers, > Daniel

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Adding/manipulating headers in CANCEL
by Örn Arnarson 24 Jun '14

24 Jun '14

Hello, I'm running Kamailio 4.0.4, and I'm running into problems when trying to insert or update headers when the method is CANCEL. It just seems that nothing at all happens to the message. Is this a bug or a design choice? Regards, Örn

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How to route on whether INVITE is for MSRP
by Alex Villacís Lasso 24 Jun '14

24 Jun '14

So far, I have a kamailio.cfg that routes INVITEs between Asterisk and the external networks. Now I want to start adding MSRP support. However, with my current configuration, the test client (Blink) sends an INVITE with a SDP payload that specifies msrps media. This gets routed to Asterisk, and it gets promptly rejected with "Not Acceptable Here". I need to make Kamailio route the request by itself if the INVITE is for MSRP, and delegate to Asterisk for other INVITEs. What is the most elegant way to test for the INVITE packet to see if it is MSRP? This testing should be at least mentioned in the documentation for the msrp.so module.

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Setting up two Kamailio Proxies in Tandem
by AliReza Khoshgoftar Monfared 23 Jun '14

23 Jun '14

Hi, I am trying to build a simple architecture like this: UAC <--> Proxy1 <--> Proxy2 <--> UAS (each on a different VM) Is it possible to have this architecture built without any "registrar" module? I have simulated my UAC and UAS using SIPp and the routing in Proxy1 and Proxy2 is to forward to the next hop. for example in Proxy1, I have: *route{ record_route(); if (loose_route()) { route(FORWARD); break; } if (uri == myself){ setdsturi("sip:__ADRESS_OF_PROXY2__"); route(FORWARD);* * break;* * } route(FORWARD); }route[FORWARD]{ if (!t_relay()) { sl_reply_error(); }}* I simulate packets in UAC in the following manner: *sipp -sn uac -rsa __ADDRESS_OF_PROXY1__ __ADDRESS_OF_PROXY1__* The problem is that my first proxy (Proxy1) is always returning "404 Not Found" on any INVITE and never forwards to Proxy2, even though I have asked it to do so in "if(uri==myself)". Am I missing something in connecting the proxies in such a simple architecture?

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kamailio not sending 202 after 408 is sent
by Allen Zhang 23 Jun '14

23 Jun '14

Hi all, I have an edge proxy (E) and a registrar (R) behind it. In R, if a MESSAGE failed to deliver for any reason, R stores the MESSAGE in msilo. If the MESSAGE timed out, R sends a 408 time out first and then send a 202 Accepted after the MESSAGE is stored. The problem is, E happily forwards the 408 to the sender of the MESSAGE, but decides not to forward the 202. The log: DEBUG: t_reply_matching: hash 16662 label 0 branch 0 DEBUG: tm [t_lookup.c:1032]: DEBUG: t_reply_matching: reply matched (T=0x7f99eebb2ae0)! DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=3 global id=3 T end=0x7f99eebb2ae0 DEBUG: tm [t_reply.c:2085]: DEBUG: reply_received: org. status uas=408, uac[0]=408 local=0 is_invite=0) DEBUG: tm [t_reply.c:1207]: ->>>>>>>>> T_code=408, new_code=202 DEBUG: tm [t_reply.c:1706]: DEBUG: relay_reply: branch=0, save=0, relay=-1 Can I change this behaviour to also forward the 202 withouting hacking in the source code? Regards, Allen

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Specifying Upstream and Downstream Servers for a simple SIP proxy
by AliReza Khoshgoftar Monfared 23 Jun '14

23 Jun '14

Dear Kamailio users, I am trying to set up a simple scenario as follows: UAC --> Proxy_1 --> Proxy_2 --> Proxy_3 --> UAC I am new to Kamailio and had the following basic questions that came to my mind after reading the documentation an the default kamilio.cfg config script: 1) Is there any good example of the above scenario in which the UAC and UAS (caller and callee) are modeled using SIPp and proxies are simple kamailio instances run on different machines that simply forward the SIP packets? 2) In specific, I want the proxies to simply forward the messages to their downstream (I do not care about registration or other operations and look for a minimally working simple SIP scenario). Is there any example of such configuration? 3) Let's imagine Proxy_2 in the above example, in the default config, it looks like that with modifying line 164, "listen=udp:10.0.0.10:5060" I can specify the address of its upstream, Proxy_1, but how and where shall I set the specifications of the downstream, Proxy_3? I suspect it is somewhere in the Routing Logic block (line 449) but not sure how it is exactly done. I see a "route(SIPOUT)" call for example but I am not sure how and where the value of SIPOUT is modified. Is it a representation of the downstream servers in the config file? Thanks, Alireza

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Is Kamailio PacketCable 2.0 compatible
by Horo, Priyam 23 Jun '14

23 Jun '14

I was looking at installing Kamailio and was wondering it is Packetcable 2.0 compatible? Thanks!

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Disabling rtpengine abrt handler
by Alex Balashov 22 Jun '14

22 Jun '14

Hello, Despite this fix https://github.com/sipwise/rtpengine/commit/cbe1f805363b3d6a117e9e5425d7994… I am continuing to experience periodic crash problems under high loads with rtpengine. This is after upgrading mediaproxy-ng -> rtpengine. Unfortunately, after this upgrade, it is impossible to tell the cause of the crash because the [seemingly] newer rtpengine SIGABRT and/or crash handler is not running properly and is getting in my way: Jun 20 15:47:55 rtphost abrtd: Directory 'ccpp-2014-06-20-15:47:54-1123' creation detected Jun 20 15:47:55 rtphost abrt[2159]: Saved core dump of pid 1123 (/usr/sbin/rtpengine) to /var/spool/abrt/ccpp-2014-06-20-15:47:54-1123 (97013760 bytes) Jun 20 15:47:55 rtphost abrtd: Executable '/usr/sbin/rtpengine' doesn't belong to any package and ProcessUnpackaged is set to 'no' Jun 20 15:47:55 rtphost abrtd: 'post-create' on '/var/spool/abrt/ccpp-2014-06-20-15:47:54-1123' exited with 1 Jun 20 15:47:55 rtphost abrtd: Deleting problem directory '/var/spool/abrt/ccpp-2014-06-20-15:47:54-1123' I am not sure why this script is exiting, but the result is that I have no /var/spool/abrt/ccpp-2014-06-20-15:47:54-1123 and no core dump to work with. This is on a standard minimal CentOS 6.5.x install. Is there any way to either disable this signal handler and just let it dump core normally, or, otherwise, to definitively fix the crash handler? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

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MSILO storing wrong content-type.
by Prasun Bheri 22 Jun '14

22 Jun '14

Hi All, I have Kamailio 4.0.0 configured with MSILO in centos 6. When a Message destined to offline recipient is received by kamailio, MSILO stores the message with wrong content type. In my scenario Incoming message has content type "message/cpim" where as in database its is stored as "text/plain". and obviously when recipient registers back in the stored message is sent with content type as "text/plain". I have attached a wireshark capture where in you can see packet #23 (recived Message) and #88 (dispatched Message). Could you guys let me know how to fix this. Thanks & Regards, Prasun

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UAC module refresh registration, could not reuse callid/fromtag !!!
by nevious 21 Jun '14

21 Jun '14

Hello all! I make kamailio1 remotely register to kamailio2. Here I set K1 modparam("uac", "reg_timer_interval", 15), and expires in uacreg as 30s. Then K1 should refresh registration after 15s. However the callid and from tag of every registration are different, and the Cseq just 10 or 11,never increased.In RFC3261, when UAC want refresh bindings, A UA SHOULD use the same Call-ID for all registrations during a single boot cycle. Is there any way to reuse the callid and fromtag during refresh registration ? Many thanks! And how to make the Cseq become 12 or more?

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Kamailio clean unconfirmed dialog
by Arun Kumar 21 Jun '14

21 Jun '14

Hi Server : kamailio 4.0 Configuration :Outbound with engage_media_proxy (See the attachment ) Shared Mem : 1024 MB Dialog TimeOut : 600 (Seconds) Problem : Dialog's are not cleared sometimes , So after sometime server is not responding (out of memory error), After investigating long time , the problem is non confirmed dialog's are not cleared even dialog timeout Thanks Arunkumar

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Re: [SR-Users] SIP Trunk
by Keith 20 Jun '14

20 Jun '14

Hi, So when I use the uac_reg_request_to, for some reason it sends an invite and not a register?? This means I'm not getting a 407 back to the use sac_auth in the failure route. Any ideas? Thanks Keith

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mediaproxy-ng query
by Alex Balashov 20 Jun '14

20 Jun '14

Hello, I'm invoking mediaproxy-ng with rtpproxy_offer/answer("ow") and am getting a scenario where: INVITE is forwarded to signalling gateway xxx.xxx.xxx.xxx, which returns an SDP answer of yyy.yyy.yyy.yyy:51964. rtpproxy is invoked in both directions. However, media forwarded from media endpoint A is sent to xxx.xxx.xxx.xxx:51964 (the correct port, but to the signalling gateway), rather than yyy.yyy.yyy.yyy:51964, which is what's specified in the SDP answer in the 200 OK from xxx.xxx.xxx.xxx. No media is coming back to mediaproxy-ng from yyy.yyy.yyy.yyy:51964. The cause of this is a separate and logically unrelated issue. Nevertheless, I am curious as to why mediaproxy-ng doesn't send media from the near end to yyy.yyy.yyy.yyy:51964--in other words, to the specified SDP endpoint on the far end, rather than to the signalling gateway. I initially imagined that this might be due to the "w" option to rtpproxy_offer(), which the documentation says forces symmetric RTP. However, this behaviour is not elaborated upon or defined. I tried removing the 'w' option but seem to be getting the same result. Is this a default behaviour of COMEDIA-style source port latching? In other words, does mediaproxy-ng need to latch onto incoming RTP from yyy.yyy.yyy.yyy:51964 before it will send media there, as opposed to the signalling gateway? If so, is there any way this behaviour can be disabled, so that mediaproxy-ng simply follows SDP in all cases? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

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Kamailio b2bua
by Grant Bagdasarian 20 Jun '14

20 Jun '14

Hello, OpenSIPS has a module for B2BUA. Does Kamailio have a similar one or one that comes close to it? Perhaps by combining modules to make Kamailio act as a B2BUA?

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