- sr-users - kamailio.org

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

2 2

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.

2 1

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.

2 2

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?

2 2

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 >

2 1

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

2 2

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

2 4

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.

2 1

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?

1 0

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

2 4

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

2 4

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!

2 1

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

3 7

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

1 0

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?

1 0

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

2 4

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

1 0

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

2 4

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?

3 4

Initial Install
by Phil Daws 20 Jun '14

20 Jun '14

Hello and good evening all! Am trying to build a Kamailio installation in my lab and have hit the first hurdle; database build. I have a three node MariaDB cluster in place and attempting to get kamdbctl create to set up the tables. What seems apparent is that command requires the database to not already exist. Where can I find the necessary schema to load into MariaDB myself please ? Thank you. Phil

2 2

Setting priority (q) for kamctl ul add
by David Wilson 20 Jun '14

20 Jun '14

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 2. Edit the q value of an existing record by using a kamctl command. Cheers, Dave.

3 2

Starting Kamailio SIP server
by Horo, Priyam 19 Jun '14

19 Jun '14

Hi, I am following the quick-start installation guide at http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x and I am stuck when I start the kamailio, using /etc/init.d/kamailio start it gives me an error of "invalid configuration file". I am using the /usr/local/etc/kamailio/kamailio.cfg file which came along with tar.gz file. I do not want sql and no authentication to begin with. What else needs to be modified or is required in the config file. Thanks!

1 0

Error codes with management commands
by PIERRE Laurent 19 Jun '14

19 Jun '14

Hello, It's unfortunate that error codes are not included with management commands. For example, if you tape " kamctl diaplan rm rule 2 999 " instead of " kamctl diaplan rmrule 2 999 " then all the entries in dialplan table will be deleted. In fat the system perform "kamctl diaplan rm" It's very annoying . -- Laurent PIERRE http://www.linkedin.com/in/lpierre

1 0

Unable To Start MSRP Session
by Megha Patil 19 Jun '14

19 Jun '14

Hi, I am trying to establish a chat session(MSRP) with two sipclients endpoints by registering to kamailio server(4.0.0), but I am getting 500 Internal server error(Reason: SIP ;text="media stream failed to start" ;cause=500). What might be the problem?? Does it mean that kamailio can not handle msrp session? If it can handle, how to enable it? I have attached the captures with this mail. Please help me in this regard.

1 0

Re: [SR-Users] SIP Trunk
by Keith 19 Jun '14

19 Jun '14

Hi, Thanks Daniel, however don't I need to send a register to get the 401 back? If so how do I do this bit? Thanks Keith

2 1

Possible Oversight, but CDRs are not being written
by Joel White 19 Jun '14

19 Jun '14

I made a significant change to the way the system I am building is designed. I removed the Database to a redundant cluster and not on the same server as the kamailio install. I changed all references to the DBURL to reflect the IP of the new cluster. I also changed the SQLOPS statement for CDRs to reflect the new IP modparam("sqlops", "sqlcon", "cb=>mysql://kamailio:kamailiorw@10.11.12.5/kamailio") All ACC transactions are being recorded to the new DB and the configuration is running well with the exception of CDRs. What am I missing that would prevent writing of CDRs to the DB? I did check connectivity from the Kamailio server to the DB server as well as access to the cdrs table

2 3

Cisco 79XX Series Phones and TLS integration with kamailio
by Jonathan Hunter 19 Jun '14

19 Jun '14

Hi All, As you guys might remember I have been doing alot of work with these legacy handsets recently, in particular the Cisco IP phones 7945G and 7965G. They work well now with kamailio, using UDP or TCP as the transport protocol. I am now looking to implement SIP over TLS with them, and wondered if anyone had completed the same, as it appears they were designed very much to support SIP over TLS and SRTP but with Cisco call manager and not other SIP devices. I am trying to understand if its possible to integrate with kamailio, as from Cisco documentation it appears a CTL file (client trust list) is required which seems to be generated by cisco software, and the handset needs it before it will try to initiate a SIP connection over TLS. I have implemented TLS /SRTP with Cisco SPA's, Bria etc on kamailio fine, so its more a question around the handset and if anyone has achieved this. Thanks Jon

3 8

Internal server error
by Deepak Bhagwat 19 Jun '14

19 Jun '14

Hi all, I am trying to establish a chat session(MSRP) with two sipclients endpoints by registering to kamailio server(4.0.0), but I am getting 500 Internal server error. What might be the problem?? Does it mean that kamailio can not handle msrp? If yes how to enable it in the server? Packets captured are attached. Please help me in this regard. -- Thanks Deepak Bhagwat

1 0

dialog module -- timeout parameter
by Måns Nilsson 19 Jun '14

19 Jun '14

Hi, I need to allow arbitrarily long calls while still using the dialog module. The module documentation does not mention a maxiumum value (2^32 sec?) nor a setting to disable the timeout functionality. I'd guess that setting it to 0 is a common convention in such cases, but I'd rather get it verified. Anything above 10 years is probably a safe bet, so a 32-bit unsigned int is quite fine. :-) Is there a max value? Is there a way to disable the check? -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 I think I'll KILL myself by leaping out of this 14th STORY WINDOW while reading ERICA JONG'S poetry!!

2 1

One way Audio with kamailio -Rtpproxy
by master voip 18 Jun '14

18 Jun '14

Hi can some one help me . I am using kamailio 4.1.3 and having one way audio issue. Scenario is Pjsip 2.0 based UA (private IP)------->Router(publicIP)--------> Kamailio (with RTP proxy)-----------> Third Party_Sip Server ------>PSTN UA is registering fine with sip server but when i make a call they can hear but i can not hear the other party . What i found is in SDP first c=IN is private ip of UA and second c=IN is public ip of UA i dont know if this is the cause When I trace the call on sip server , sip server is sending RTP to UA private ip so it is not reaching kamailio back to forward to UA . I am attaching the kamailio configuration file as well as SIP/SDP trace user agent private ip : 192.168.1.4 User agent public ip : 61.61.61.61 Kamailio IP : 81.81.81.81 Sip Server IP: 71.71.71.71 (* ip addresses are not actual ips) Please reply with detailed instructions to fix this isue config file: #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # #!define WITH_NAT # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ' '; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ' '; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=2 log_stderror=no #!else debug=3 log_stderror=no #!endif memdbg=1 memlog=1 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=7878 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" loadmodule "nat_traversal.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 7) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication # Below line commented by test-MATHUR as PC-2-PC calls are not getting through #route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { sl_send_reply("100", "Trying"); setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { xlog("testM : Unable to relay !!!! :{ \n"); sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp )\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$s i:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # 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()) { route(DLGURI); if (is_method("BYE")) { #xlog("testM : Got BYE from some PSTN/UAC.\n"); setflag(FLT_ACC); # do accounting ... #xlog("testM : Did accounting.\n"); setflag(FLT_ACCFAILED); # ... even if the transa ction fails #xlog("testM : Calling route{NATMANAGE} from WI THINDLG.\n"); #t_newtran(); #t_reply("200", "OK"); #xlog("testM : Called route{NATMANAGE} for BYE froom WITHINDLG.\n"); #exit; } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per R FC 6665. record_route(); } 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 route(RELAY); exit; } else { # ACK without matching transaction ... i gnore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $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"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } #if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) #{ # rtpproxy_manage(); # return; #} rtpproxy_manage("z90"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { add_rr_param(";nat=yes"); } } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n" ); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bu g in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_i p) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } ------------------------------------------------------------------------------------------------------------------ SIP TRACE REGISTER sip:71.71.71.71 SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK1b54.98ebaeef387ced8eb54624e6c7a90504.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjI5YxCTUEIkkWEg0VswESOCq7lpvwYUkE Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW To: "57778" <sip:test@71.71.71.71> Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 CSeq: 62147 REGISTER User-Agent: pj_arubaslim-16/r2 Contact: "57778" <sip:test@192.168.1.4:58931;ob> Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 P-hint: outbound SIP/2.0 407 Proxy Authentication Required CSeq: 62147 REGISTER Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK1b54.98ebaeef387ced8eb54624e6c7a90504.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjI5YxCTUEIkkWEg0VswESOCq7lpvwYUkE From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 To: "57778" <sip:test@71.71.71.71>;tag=170604141058 Proxy-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301389817101016806045824224" Content-Length: 0 REGISTER sip:71.71.71.71 SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKfb54.1eaeef967bf46fb647eed0bb87a9406e.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjRPPULjuny1lG1x2Qn0UumpeHPLGxqQHI Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW To: "57778" <sip:test@71.71.71.71> Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 CSeq: 62148 REGISTER User-Agent: pj_arubaslim-16/r2 Contact: "57778" <sip:test@192.168.1.4:58931;ob> Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Proxy-Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301389817101016806045824224", uri="sip:71.71.71.71", response="0ca765f266b6bfc8ea0ef00227272393" Content-Length: 0 P-hint: outbound SIP/2.0 200 OK CSeq: 62148 REGISTER Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKfb54.1eaeef967bf46fb647eed0bb87a9406e.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjRPPULjuny1lG1x2Qn0UumpeHPLGxqQHI From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 To: "57778" <sip:test@71.71.71.71>;tag=170604141059 Contact: "57778" <sip:test@192.168.1.4:58931;ob>;expires=600 Expires: 600 Content-Length: 0 SUBSCRIBE sip:test@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK0f2c.8c652a23fe5f790c4bb9216d48b80199.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPj.syjKlsSGhtf420D3Gs4.gYpLomA2oQq Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC To: "57778" <sip:test@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB CSeq: 10987 SUBSCRIBE Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: pj_arubaslim-16/r2 Content-Length: 0 P-hint: outbound SIP/2.0 401 Unauthorised CSeq: 10987 SUBSCRIBE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK0f2c.8c652a23fe5f790c4bb9216d48b80199.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPj.syjKlsSGhtf420D3Gs4.gYpLomA2oQq From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=170605141000 Content-Length: 0 WWW-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301390017101016806050024224" Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> SUBSCRIBE sip:test@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKef2c.315393ac3a4f54a9fd7fe71cb140b41b.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjIFtOYolYGyRGCBo77jg6j38Co3oSuB6l Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC To: "57778" <sip:test@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB CSeq: 10988 SUBSCRIBE Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: pj_arubaslim-16/r2 Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301390017101016806050024224", uri="sip:test@71.71.71.71", response="15ee3de757c18e084fca607438017093" Content-Length: 0 P-hint: outbound SIP/2.0 200 OK CSeq: 10988 SUBSCRIBE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKef2c.315393ac3a4f54a9fd7fe71cb140b41b.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjIFtOYolYGyRGCBo77jg6j38Co3oSuB6l From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=170605141001 Expires: 3600 Content-Length: 0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> NOTIFY sip:test@71.71.71.71 SIP/2.0 CSeq: 100 NOTIFY Via: SIP/2.0/UDP 71.71.71.71:5060 From: "57778" <sip:test@71.71.71.71>;tag=170605141001 Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Contact: <sip:71.71.71.71:5060;transport=udp> Subscription-State: active Event: message-summary Content-Type: application/simple-message-summary Content-Length: 65 Messages-Waiting: no Message-Account: sip:test@71.71.71.71 SIP/2.0 404 Not here CSeq: 100 NOTIFY Via: SIP/2.0/UDP 71.71.71.71:5060;rport=5060 From: "57778" <sip:test@71.71.71.71>;tag=170605141001 Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Server: kamailio (4.1.3 (i386/linux)) Content-Length: 0 INVITE sip:919000000002@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjcJJWEomCJmbTkvY8iXMdrmQ935aS1h1g Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23382 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: pj_arubaslim-16/r2 Content-Type: application/sdp Content-Length: 356 P-hint: outbound v=0 o=- 3612003234 3612003234 IN IP4 81.81.81.81 s=pjmedia c=IN IP4 192.168.1.4 t=0 0 m=audio 43314 RTP/AVP 3 18 0 8 101 c=IN IP4 81.81.81.81 a=rtcp:43315 a=sendrecv a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes SIP/2.0 407 Proxy Authentication Required CSeq: 23382 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjcJJWEomCJmbTkvY8iXMdrmQ935aS1h1g From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Proxy-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301391017101016806051024224" Content-Length: 0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> ACK sip:919000000002@71.71.71.71 SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0 Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23382 ACK Content-Length: 0 INVITE sip:919000000002@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23383 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: pj_arubaslim-16/r2 Proxy-Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301391017101016806051024224", uri="sip:919000000002@71.71.71.71", response="4edf5e2efd1e8e03bdbe4a693fdfee4c" Content-Type: application/sdp Content-Length: 356 P-hint: outbound v=0 o=- 3612003234 3612003234 IN IP4 81.81.81.81 s=pjmedia c=IN IP4 192.168.1.4 t=0 0 m=audio 43314 RTP/AVP 3 18 0 8 101 c=IN IP4 81.81.81.81 a=rtcp:43315 a=sendrecv a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes SIP/2.0 183 Session Progress CSeq: 23383 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Type: application/sdp Content-Length: 252 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> v=0 o=SipSwitch 6248 7248 IN IP4 71.71.71.71 s=VoipSIP i=Audio Session c=IN IP4 71.71.71.71 t=0 0 m=audio 6248 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 180 Ringing CSeq: 23383 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Type: application/sdp Content-Length: 252 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> v=0 o=SipSwitch 6248 7248 IN IP4 71.71.71.71 s=VoipSIP i=Audio Session c=IN IP4 71.71.71.71 t=0 0 m=audio 6248 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 200 OK CSeq: 23383 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Type: application/sdp Content-Length: 252 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> v=0 o=SipSwitch 6248 7248 IN IP4 71.71.71.71 s=VoipSIP i=Audio Session c=IN IP4 71.71.71.71 t=0 0 m=audio 6248 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ACK sip:71.71.71.71:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.e624c28002d41f48ae76813b26e516f3.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJ8XsuToNj.92GHJLiuK-JZ5T5PpfTbhr Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23383 ACK Content-Length: 0 BYE sip:71.71.71.71:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK976c.6ca2249268e6f9592a6948becb29c474.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJy-pnGAhtqwqV5uRixN-mvDWOTPLupLU Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23384 BYE User-Agent: pj_arubaslim-16/r2 Content-Length: 0 SIP/2.0 200 OK CSeq: 23384 BYE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK976c.6ca2249268e6f9592a6948becb29c474.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJy-pnGAhtqwqV5uRixN-mvDWOTPLupLU From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Length: 0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

1 0

Trigger m_dump() with REGISTER
by Marc M. 18 Jun '14

18 Jun '14

Hi, I have a basic MSILO setup up and running. For the moment every time the client sends a REGISTER, m_dump() is called. I would like to achieve the following: The clients are registering with either 200 or 300 sec expiration time. I would like to dump the messages only when the client is registering for 300 secs. Only a REGISTER with an expiration time of 300 should trigger m_dump(). Can you help me with this? Any help or idea is appreciated. thanks Marc

1 0

Re: [SR-Users] SIP Trunk
by Keith 18 Jun '14

18 Jun '14

Hi, I have a carrier who requires SIP authentication on calls, can anyone point me in the right direction? Normally in Asterisk or Freeswitch you can authenticate a peer, but can't see how to do this in Kamailio. Thanks Keith

2 1

Use of asterisk * in Kamailio routing
by Joel White 18 Jun '14

18 Jun '14

I am having an issue using an asterisk * in the kamailio routing. How would I implement this? Thank you in advance

3 6

JSON RPC Usage in Kamailio
by Jaytirth S Khairatkar 18 Jun '14

18 Jun '14

Hello, Can anyone give me a link or document which explains how to use the JSON and JSON-RPC module in Kamailio SER? I'll greatly appreciate it. Thanks and regards Jaytirth *Disclaimer* ------------------------------------------------------------ Notice: This e-mail and / or attachments (communication) may contain information that is confidential and proprietary to The Tata Power Company Limited, Strategic Engineering Division (Tata Power SED). If you are not the intended recipient of this communication, please do not use or disseminate the information contained therein, notify the sender by e-mail or telephone and delete the communication (permanently) from your system. Any information in this communication that does not relate to the official business of Tata Power SED shall be understood as neither given nor endorsed by it. Please note that e-mails are susceptible to change and Tata Power SED shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt. This communication is not guaranteed to be free from computer viruses and it is recommended that you check for all viruses before download. Print this communication only if absolutely essential. Thank you for your co-operation. ------------------------------------------------------------

1 0

s-cscf - limit of entries for 1 contact
by Ciprus, Daniel 17 Jun '14

17 Jun '14

All, is any of you aware about limitation how many reg entries can be stored in s-cscf for 1 contact ? From what I'm observing here : s-cscf can have up to 12 reg entries for 1 contact and then other entries are silently discarded. Use case: - crash of app happens 12 times - app retries to register - s-cscf sends back 200 OK with 13 contacts - another app registers and tries to send SIP MESSAGE to 1st app Result: message never gets delivered and tcpdump shows 12 messages sent out to p-cscf but not 13th one which is actually live session between client and proxy. thanks Dan ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 2

Install SEMS
by Agiftel 17 Jun '14

17 Jun '14

Hi All, i am trying to figure out how to get SEMS and install it. If i go to iptel website, documentations and last versions give always back NOT FOUND. Is this project still maintaned?/avaible? Regards -- View this message in context: http://sip-router.1086192.n5.nabble.com/Install-SEMS-tp128500.html Sent from the Users mailing list archive at Nabble.com.

2 1

Re: [SR-Users] LCR from URI (CLI)
by Keith 17 Jun '14

17 Jun '14

Hi all, I have an issue where I have two carriers, one only accepts +44 then the CLI and the other only accepts the number with the leading zero dropped. I am using LCR to route to carriers, however I can only switch carriers by manually altering the CLI. Is there a way I can manipulate the CLI based on the destination the call is being sent to? Thanks Keith

2 1

Problems to start kamailio
by Carlos Rangel 17 Jun '14

17 Jun '14

Hello I am having problems to start kamailio 4.1.3 on a fresh centos 6.5 install with Asterisk 11.9 following the tutorial. Everything has gone well except that I cannot start kamailio. Basically I have two problems. The message that I got from the console when I try to start is [root@gt-ev-i08 kamailio-4.1.3]# kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed An when I look at the error messages in var/log/messages I see the following errors. Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 839, column 18: syntax error Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 839, column 18: bad command: missing ';'? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 839, column 18: bad command Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 6: syntax error Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 6: missing '(' or ')' ? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 6: bad command: missing ';'? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 16: '('')' expected (function call) Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 16: bad command: missing ';'? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 18: bad command I did not touch that section of the kamailio.cfg file. I am really lost now. Hopefully someone can help. Thanks Carlos

2 2

Unable to make calls within the extensions
by Chandramouli P 17 Jun '14

17 Jun '14

Hi, I am new to Kamailio. I started RTPProxy using "rtpproxy -A 54.85.12.15 -F -l 10.0.0.122 -s udp:localhost:7722" command and see that my sip phones are registered with Kamailio. I am able to see using "kamctl ul show" command. But, I am unable to establish call between my registered extension through Asterisk using Kamailio. I see that calls are hitting my Asterisk server when I call from extension. I am not getting any audio and the other extension is also not at all ringing. I could not able to figure out where I am doing the mistake? I am not sure whether the mistake is in Kamailio or Asterisk. In my Asterisk server, I am seeing the correct configuration using "odbcinst -q -d" and "isql -v MySQL-asterisk chandra test123456" and "odbc show (At CLI>)". Please find the below environment: Operating System: Ubuntu 14.04 Server (64-Bit) Kamailio: 4.0 Asterisk: 11.10 Database: MySQL (UNIX ODBC) Environment: Amazon EC2 Follwed Links: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb http://pastebin.com/VPpfErYn Daniel's Patched RTPProxy Installation (Please note that My Asterisk and Kamailio servers are behind NAT in Amazon EC2) Below is my configuration: *http://pastebin.com/fr8m9gr7 <http://pastebin.com/fr8m9gr7>* *Note:* I inserted the records in to the respective tables with "cmp" as context. When I call from 100 to 500, I am not getting any sound and another extension is not ringing and getting the below messages at Asterisk CLI: *http://pastebin.com/y4tXXrnF <http://pastebin.com/y4tXXrnF>* Any update would be appreciated. Thanks in advance. Regards, Chandra.

3 5

Problem with CANCEL handling
by Mordechay Kaganer 17 Jun '14

17 Jun '14

B.H. Hello, all! I'm trying to set up kamailio as a simple load balancer proxy between PSTN gateway and a cluster of asterisk servers - rather common use case IMHO. I'm completely new to kamailio. I have started with the default kamailio.cfg that comes with Ubuntu package and added the dispatcher module as described in the docs ( http://www.kamailio.org/docs/modules/4.0.x/modules/dispatcher.html). At the end, i got this up and running, but with one exception: if incoming call is not answered by the asterisk end (i.e. it is in ringing state), and then the gateway sends CANCEL request, kamailio is unable to recognize the transaction and is not forwarding the request to asterisk. The result is: the call is left "dangling" in the PBX and is not properly hang up. When the gateway sends CANCEL to kamailio, i get the following log messages: 4(25442) DEBUG: tm [t_lookup.c:769]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT: searching on hash entry 31287 4(25442) DEBUG: tm [t_lookup.c:492]: matching_3261(): DEBUG: RFC3261 transaction matching failed 4(25442) DEBUG: tm [t_lookup.c:859]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT: no CANCEL matching found! 4(25442) DEBUG: tm [t_lookup.c:861]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT completed 4(25442) DEBUG: tm [tm.c:1039]: w_t_lookup_cancel(): lookup_original: t_lookupOriginalT returned: (nil) 4(25442) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=485 a=25 n=xdbg 4(25442) DEBUG: <script>: is_method(CANCEL): t_check_trans()==false 4(25442) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=487 a=2 n=exit The relevant script snippet is standard from the distro package: # CANCEL processing if (is_method("CANCEL")) { xdbg("is_method(CANCEL): entry"); if (t_check_trans()) { xdbg("is_method(CANCEL): t_check_trans()==true"); route(RELAY); } else { xdbg("is_method(CANCEL): t_check_trans()==false"); } exit; } How can i fix this? Thanks! -- משיח NOW! Moshiach is coming very soon, prepare yourself! יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד!

1 2

Re: [SR-Users] sip aliases
by Anna Schuster 17 Jun '14

17 Jun '14

Hi Quick question: If I create different sip aliases under my account, is any way I can see under Missed Called section on serweb.iptel.org or any elsewhere on which alias the call came on? Thank you, Anna On Sat, Jun 7, 2014 at 3:11 PM, <registrar(a)iptel.org> wrote: > > Thank you for registering with iptel.org. > > We are reserving the following SIP address for you: sip:mielul@iptel.org > > To finalize your registration please check the following URL within 24 > hours: > http://serweb.iptel.org/user/reg/confirmation.php?nr=0eb4729fad0f99434d205f… > (If you confirm later you will have to re-register.) > > Here are the details of your account: > --------------------------------------------------------------------- > Username: mielul(a)iptel.org > Password: q1q1q1q1 > Email: annasc74(a)gmail.com > Full Name: Anna M > Sip address: sip:mielul@iptel.org > (You can always retrieve these via the "Forgot Password" link on the > signup page) > > > > -- This message is automatically generated by SerWeb. >

2 1

Re: [SR-Users] exec_avp problem
by Ansuman Manna - Asergis Telecom 17 Jun '14

17 Jun '14

Hi, I am using the exec_avp command as followed: exec_avp("/usr/bin/mwi/test.sh $fU"); $fU taking command line arguments. Here is the shell script content: #!/bin/bash -e echo $1 logger done! $1 logger done mkdir /usr/bin/mwi/$1 || { logger 'touch failed' $? $errorMessage ; exit 1; } logger is working fine but.mkdir and other system commands are not executing. I cant figureout my misses Regards, Ansuman Manna Software Developer Asergis Telecom Services Private Limited Office Tel: +91 11 6606 8400 Office Fax: +91 112554 5156 http://www.asergis.in <http://www.asergis.in>

2 1

How to use unixodbc/freetds for kamailio backend
by Tim Chubb 17 Jun '14

17 Jun '14

Hi Im just getting started with kamailio with a view of evaluating it for production use. Is it possible to do the following: 1) I would like to use ODBC/FreeTDS to connect to an existing ms sql server cluster and use it for all DB related activity relating to kamailio, is this possible as we would be running kamailio in a HA configuration in production and I really don't want to have to support another DB cluster? 2) Looking at the kamdbctl scripts it seems that currently every popular db but sql server is supported which is a bit of a shame, however am I right in saying that if I was to port the sql files to SQL server, and implemented a script similar to kamdbctl.pgsql to wrap around the freetds TSQL executable that would be all that I would need to do add support for SQL server? 3) Is there any particular reason why there isn't an ODBC option for the kamdbctl script? Cheers.

2 1

Kamailio and Freepbx Integration
by Carlos Rangel 16 Jun '14

16 Jun '14

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

1 0

stampeeding server.
by Måns Nilsson 16 Jun '14

16 Jun '14

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

2 3

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

16 Jun '14

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

2 1

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

13 Jun '14

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

1 1

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

13 Jun '14

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

2 6

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

13 Jun '14

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

2 2

MSILO add on deliver event route still no joy
by Roberto Fichera 13 Jun '14

13 Jun '14

Hi All, I'm trying to add an event route to be execute for each successfully delivered message from msilo module, but I'm still not getting running the event route I have defined for it in the script. If someone can have a look at the patch and give me some hint would be really great. Thanks in advance. Roberto Fichera.

2 15

Transport protocol related question - how to bind a dialog to a negotiated transport protocol
by Klaus Feichtinger 12 Jun '14

12 Jun '14

Hello, I wonder if it is allowed using transport protocol UDP for SIP NOTIFY requests (which are generated by Kamailio/presence module), when the SUBSCRIBE dialog was established using TCP as transport protocol. In other words: this is a principal question if it is allowed changing the transport protocol for in-dialog transactions e.g. from TCP to UDP. Initially I rather thought that in-dialog transactions shall use the same transport protocol as the dialog itself (e.g. SIP INFO requests within a standard media session), except the message would be too big for UDP, where a change to TCP is recommended. Can anybody give me a hint, if the current behaviour of Kamailio is correct or not? Or - how can I bind Kamailio to a specific transport protocol (for messages that are created by modules themselves)? Kamailio is sending NOTIFY requests with UDP, even when the subscription was done with TCP (see excerpt below). 09:58:10.360749 IP (tos 0x0, ttl 64, id 35302, offset 0, flags [DF], proto TCP (6), length 444) 10.1.1.14.37883 > 10.1.1.44.5060: P, cksum 0x1cb3 (correct), 1:393(392) ack 1 win 457 <nop,nop,timestamp 624699305 795715664> SUBSCRIBE sip:116006@10.1.1.44;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 10.1.1.14:5070;rport;branch=z9hG4bK1540213071 From: <sip:1@10.1.1.14:5070>;tag=620071678 To: <sip:116006@10.1.1.44;transport=TCP> Call-ID: 449986375 CSeq: 20 SUBSCRIBE Contact: <sip:1@10.1.1.14:37883> Max-Forwards: 70 Expires: 1200 Event: presence Content-Length: 0 09:58:10.361324 IP (tos 0x10, ttl 64, id 65438, offset 0, flags [DF], proto TCP (6), length 431) 10.1.1.44.5060 > 10.1.1.14.37883: P, cksum 0x29fb (correct), 1:380(379) ack 393 win 215 <nop,nop,timestamp 795715792 624699305> SIP/2.0 202 OK Via: SIP/2.0/TCP 10.1.1.14:5070;rport=37883;branch=z9hG4bK1540213071 From: <sip:1@10.1.1.14:5070>;tag=620071678 To: <sip:116006@10.1.1.44;transport=TCP>;tag=4f7a7e54f75c89f5b968c90011d693b5-9eed Call-ID: 449986375 CSeq: 20 SUBSCRIBE Expires: 1200 Contact: <sip:10.1.1.44:5060;transport=tcp> Content-Length: 0 09:58:10.361507 IP (tos 0x10, ttl 64, id 32295, offset 0, flags [none], proto UDP (17), length 484) 10.1.1.44.5060 > 10.1.1.14.37883: SIP, length: 456 NOTIFY sip:1@10.1.1.14:37883 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.44;branch=z9hG4bKc408.509e6347000000000000000000000000.0 To: sip:1@10.1.1.14;tag=620071678 From: sip:116006@10.1.1.44;tag=4f7a7e54f75c89f5b968c90011d693b5-9eed CSeq: 2 NOTIFY Call-ID: 449986375 Content-Length: 0 User-Agent: kamailio (4.0.4 (i386/linux)) Max-Forwards: 70 Event: presence Contact: <sip:10.1.1.44:5060;transport=tcp> Subscription-State: active;expires=1200 Thx Klaus

2 2

Re: [SR-Users] Crash Kamailio 4.1.3
by Igor Potjevlesch 12 Jun '14

12 Jun '14

Hello, Do you any ideas about the reason of this issue? Is it caused by the last call? Do you have enough information with the PAI? Regards, Igor. 2014-05-28 13:47 GMT+02:00 Igor Potjevlesch <igor.potjevlesch(a)gmail.com>: > Hello, > > p *msg->pai : > > $4 = {type = HDR_PAI_T, name = { > s = 0x7fad2cf8af74 "P-Asserted-Identity: > \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: > application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 > 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 > a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 > PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 > telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", > len = 19}, body = { > s = 0x7fad2cf8af89 "\"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: > application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 > 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 > a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 > PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 > telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", > len = 43}, len = 66, > parsed = 0x7fad389a7220, next = 0x7fad2cf8baf0} > > > p *((p_id_body_t*)msg->pai->parsed) : > > $5 = {id = 0x0, num_ids = 0, next = 0x0} > > Regards, > > > > Igor. > > > 2014-05-28 13:29 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > > Hello, >> >> I need now: >> >> p *msg->pai >> p *((p_id_body_t*)msg->pai->parsed) >> >> Cheers, >> Daniel >> >> >> On 28/05/14 12:46, Igor Potjevlesch wrote: >> >> Hello Daniel, >> >> Here is the result of p *msg : >> >> $3 = {id = 128120, pid = 2414, tval = {tv_sec = 1401265832, tv_usec = >> 5951}, fwd_send_flags = {f = 0 '\000', blst_imask = 0 '\000'}, >> rpl_send_flags = {f = 0 '\000', >> blst_imask = 0 '\000'}, first_line = {type = 1, len = 48, u = >> {request = {method = { >> s = 0x7fad2cf8ac30 "INVITE sip:0123456789@domainSIP/2.0\r\nVia: SIP/2.0/UDP >> a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: 69\r\nFrom: >> \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 6}, uri = { >> s = 0x7fad2cf8ac37 "sip:0123456789@domain SIP/2.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 31}, version = { >> s = 0x7fad2cf8ac57 "SIP/2.0\r\nVia: SIP/2.0/UDP >> a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: 69\r\nFrom: >> \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 7}, >> method_value = 1}, reply = {version = { >> s = 0x7fad2cf8ac30 "INVITE sip:0123456789@domainSIP/2.0\r\nVia: SIP/2.0/UDP >> a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: 69\r\nFrom: >> \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 6}, status = { >> s = 0x7fad2cf8ac37 "sip:0123456789@domain SIP/2.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 31}, reason = { >> s = 0x7fad2cf8ac57 "SIP/2.0\r\nVia: SIP/2.0/UDP >> a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: 69\r\nFrom: >> \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 7}, >> statuscode = 1}}}, via1 = 0x7fad2cf8b158, via2 = 0x0, headers = >> 0x7fad2cf8b118, last_header = 0x7fad2cf8bb30, parsed_flag = >> 18446744073709551615, >> h_via1 = 0x7fad2cf8b118, h_via2 = 0x0, callid = 0x7fad2cf8b7a0, to = >> 0x7fad2cf8b548, cseq = 0x7fad2cf8b7e0, from = 0x7fad2cf8b300, contact = >> 0x7fad2cf8b760, >> maxforwards = 0x7fad2cf8b2c0, route = 0x0, record_route = 0x0, >> content_type = 0x7fad2cf8baf0, content_length = 0x7fad2cf8bb30, >> authorization = 0x0, expires = 0x0, >> proxy_auth = 0x7fad2cf8b8d0, supported = 0x7fad2cf8ba70, require = 0x0, >> proxy_require = 0x0, unsupported = 0x0, allow = 0x7fad2cf8ba30, event = >> 0x0, accept = 0x0, >> accept_language = 0x0, organization = 0x0, priority = 0x0, subject = >> 0x0, user_agent = 0x7fad2cf8b850, server = 0x0, content_disposition = 0x0, >> diversion = 0x0, >> rpid = 0x7fad2cf8b890, refer_to = 0x0, session_expires = 0x0, min_se = >> 0x0, sipifmatch = 0x0, subscription_state = 0x0, date = 0x7fad384baa90, >> identity = 0x0, >> identity_info = 0x0, pai = 0x7fad2cf8bab0, ppi = 0x0, path = 0x0, >> privacy = 0x0, body = 0x0, >> eoh = 0x7fad2cf8afea "\r\nv=0\r\no=root 1751857229 1751857230 IN IP4 >> a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 a.b.c.d\r\nt=0 >> 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:3 >> GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> unparsed = 0x7fad2cf8afea "\r\nv=0\r\no=root 1751857229 1751857230 IN >> IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 a.b.c.d\r\nt=0 >> 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:3 >> GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> rcv = {src_ip = {af = 2, len = 4, u = {addrl = {1950469485, >> 140381949486832}, addr32 = {1950469485, 0, 943417072, 32685}, addr16 = >> {52589, >> 29761, 0, 0, 26352, 14395, 32685, 0}, addr = >> "m\315At\000\000\000\000\360f;8\255\177\000"}}, dst_ip = {af = 2, len = 4, >> u = {addrl = {67638457, 0}, >> addr32 = {67638457, 0, 0, 0}, addr16 = {5305, 1032, 0, 0, 0, 0, >> 0, 0}, addr = "\271\024\b\004", '\000' <repeats 11 times>}}, src_port = >> 5060, dst_port = 5060, >> proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = >> 2, sa_data = "\023\304m\315At\000\000\000\000\000\000\000"}, sin = >> {sin_family = 2, >> sin_port = 50195, sin_addr = {s_addr = 1950469485}, sin_zero = >> "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, >> sin6_flowinfo = 1950469485, sin6_addr = {__in6_u = {__u6_addr8 = >> '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, >> __u6_addr32 = {0, 0, 0, >> 0}}}, sin6_scope_id = 0}}, bind_address = 0x7fad383b6588, >> proto = 1 '\001'}, >> buf = 0x7fad2cf8ac30 "INVITE sip:0123456789@domain SIP/2.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 1254, new_uri = { >> s = 0x7fad2ceabdef "sip:0123456789@gw SIP/2.0\r\nRecord-Route: >> <sip:a.b.c.d;lr=on>\r\nVia: SIP/2.0/UDP >> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5060\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 INVITE\ >> sendrecv\r\n", len = 1254, new_uri = { >> s = 0x7fad2ceabdef "sip:0123456789@gw SIP/2.0\r\nRecord-Route: >> <sip:a.b.c.d;lr=on>\r\nVia: SIP/2.0/UDP >> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5060\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 INVITE\---Type >> <return> to continue, or q <return> to quit--- >> r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nDate: Wed, 28 >> May 2014 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY, INFO\r\nSupported: replaces, >> timer\r\nP-Asserted-Identity: \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 19}, dst_uri = {s = 0x0, len = 0}, parsed_uri_ok = 0, parsed_uri = >> {user = { >> s = 0x7fad2ceabdf3 "0123456789@gw SIP/2.0\r\nRecord-Route: >> <sip:a.b.c.d;lr=on>\r\nVia: SIP/2.0/UDP >> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5060\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nDate: Wed, 28 >> May 2014 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY, INFO\r\nSupported: replaces, >> timer\r\nP-Asserted-Identity: \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 10}, passwd = {s = 0x0, len = 0}, host = { >> s = 0x7fad2ceabdfe "gw SIP/2.0\r\nRecord-Route: <sip:a.b.c.d;lr=on>\r\nVia: >> SIP/2.0/UDP >> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5060\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nDate: Wed, 28 >> May 2014 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY, INFO\r\nSupported: replaces, >> timer\r\nP-Asserted-Identity: \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 4}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, >> sip_params = {s = 0x0, len = 0}, headers = { >> s = 0x0, len = 0}, port_no = 0, proto = 0, type = SIP_URI_T, flags >> = 0, transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = >> {s = 0x0, >> len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, >> lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, >> transport_val = { >> s = 0x0, len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = >> {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, >> len = 0}, lr_val = { >> s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, >> len = 0}}, parsed_orig_ruri_ok = 0, parsed_orig_ruri = {user = {s = 0x0, >> len = 0}, passwd = { >> s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len >> = 0}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers >> = {s = 0x0, >> len = 0}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = 0, >> transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = {s = >> 0x0, len = 0}, >> maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = >> 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, >> transport_val = {s = 0x0, >> len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, >> len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, >> lr_val = {s = 0x0, >> len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = >> 0}}, add_rm = 0x7fad2ce523c8, body_lumps = 0x0, reply_lump = 0x0, >> add_to_branch_s = '\000' <repeats 57 times>, add_to_branch_len = 0, >> hash_index = 26854, msg_flags = 145, flags = 2, set_global_address = {s = >> 0x0, len = 0}, >> set_global_port = {s = 0x0, len = 0}, force_send_socket = 0x0, path_vec >> = {s = 0x0, len = 0}, instance = {s = 0x0, len = 0}, reg_id = 0, ruid = {s >> = 0x0, len = 0}, >> location_ua = {s = 0x0, len = 0}, ldv = {flow = {decoded = 0, rcv = >> {src_ip = {af = 0, len = 0, u = {addrl = {0, 0}, addr32 = {0, 0, 0, 0}, >> addr16 = {0, 0, 0, 0, 0, >> 0, 0, 0}, addr = '\000' <repeats 15 times>}}, dst_ip = {af >> = 0, len = 0, u = {addrl = {0, 0}, addr32 = {0, 0, 0, 0}, addr16 = {0, 0, >> 0, 0, 0, 0, 0, 0}, >> addr = '\000' <repeats 15 times>}}, src_port = 0, dst_port = >> 0, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 0, >> sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, >> sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = >> "\000\000\000\000\000\000\000"}, sin6 = { >> sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr >> = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, >> 0, 0, 0, 0, 0, 0}, >> __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, >> bind_address = 0x0, proto = 0 '\000'}}}} >> >> Regards, >> >> >> >> Igor. >> >> >> >> 2014-05-28 12:01 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >> >>> Hello, >>> >>> can you give the output in gdb for: >>> >>> set print elements 2000 >>> p *msg >>> >>> Cheers, >>> Daniel >>> >>> >>> On 28/05/14 11:48, Igor Potjevlesch wrote: >>> >>> hello, >>> >>> Kamailio 4.1.3 crashed today, with an segmentation fault error : >>> kernel: kamailio[2406]: segfault at 18 ip 0000000000495792 sp >>> 00007fff3d9d2d80 error 4 in kamailio[400000+275000] >>> >>> Here is the result of BT : >>> #0 0x0000000000495792 in pv_get_strval (msg=0x7fad2cf8a530, >>> param=0x7fad3858c288, res=0x7fff3d9d2eb0, sval=0x18) at pvapi.c:521 >>> #1 0x00007fad365b0da8 in pv_get_pai (msg=0x7fad2cf8a530, >>> param=0x7fad3858c288, res=0x7fff3d9d2eb0) at pv_core.c:1021 >>> #2 0x0000000000499518 in pv_get_spec_value (msg=0x7fad2cf8a530, >>> sp=0x7fad3858c270, value=0x7fff3d9d2eb0) at pvapi.c:1266 >>> #3 0x00007fad34c6d7d9 in extra2strar (extra=0x7fad3858c260, >>> rq=0x7fad2cf8a530, val_arr=0x7fad34e7ae30, >>> int_arr=0x7fad34e7b37c,type_arr=0x7fad34e7b4e7 "\002\002\002\002") at >>> acc_extra.c:261 >>> #4 0x00007fad34c65393 in acc_db_request (rq=0x7fad2cf8a530) at acc.c:474 >>> #5 0x00007fad34c6fb43 in acc_onreply (t=0x7fad2ce0b138, >>> req=0x7fad2cf8a530, reply=0x7fad3859faa8, code=200) at acc_logic.c:471 >>> #6 0x00007fad34c7008d in tmcb_func (t=0x7fad2ce0b138, type=512, >>> ps=0x7fff3d9d3120) at acc_logic.c:559 >>> #7 0x00007fad36e6a478 in run_trans_callbacks_internal >>> (cb_lst=0x7fad2ce0b1a8, type=512, trans=0x7fad2ce0b138, >>> params=0x7fff3d9d3120) at t_hooks.c:290 >>> #8 0x00007fad36e6a68a in run_trans_callbacks_with_buf (type=512, >>> rbuf=0x7fad2ce0b1f8, req=0x7fad2cf8a530, repl=0x7fad3859faa8, flags=183) at >>> t_hooks.c:336 >>> #9 0x00007fad36e9cc06 in relay_reply (t=0x7fad2ce0b138, >>> p_msg=0x7fad3859faa8, branch=0, msg_status=183, cancel_data=0x7fff3d9d3480, >>> do_put_on_wait=1) at t_reply.c:2001 >>> #10 0x00007fad36e9f0b7 in reply_received (p_msg=0x7fad3859faa8) at >>> t_reply.c:2499 >>> #11 0x000000000045d837 in do_forward_reply (msg=0x7fad3859faa8, mode=0) >>> at forward.c:777 >>> #12 0x000000000045e0f8 in forward_reply (msg=0x7fad3859faa8) at >>> forward.c:860 >>> #13 0x00000000004a5887 in receive_msg ( >>> buf=0x924600 "SIP/2.0 183 Session Progress\r\nVia: SIP/2.0/UDP >>> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0;received=a.b.c.d\r\nVia: >>> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5"..., len=872, >>> rcv_info=0x7fff3d9d3800) at receive.c:273 >>> #14 0x000000000053c838 in udp_rcv_loop () at udp_server.c:536 >>> #14 0x000000000053c838 in udp_rcv_loop () at udp_server.c:536 >>> #15 0x000000000046d42b in main_loop () at main.c:1617 >>> #16 0x00000000004704d3 in main (argc=7, argv=0x7fff3d9d3b38) at >>> main.c:2533 >>> >>> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> >> >

2 19

Kamailio v4.1.4 Released
by Daniel-Constantin Mierla 12 Jun '14

12 Jun '14

Hello, Kamailio SIP Server v4.1.4 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.3. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.4. For more details about version 4.1.4 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/06/kamailio-v4-1-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Problem with MSRP AUTH
by Muhammad Shahzad 12 Jun '14

12 Jun '14

Hi, I am trying to authentication MSRP connection using the example code of msrp event route in module documentation here, http://kamailio.org/docs/modules/4.1.x/modules/msrp.html#idp119248 -- ... } else if ($msrp(method)=="AUTH") { ... if (!pv_www_authenticate("WEBRTC_SIP_REALM", "$var(passwd)", "0", "$msrp(method)")) { if (auth_get_www_authenticate("WEBRTC_SIP_REALM", "0", "$var(wauth)")) { xlog("L_INFO", "Generated www authenticate header for MSRP is [$var(wauth)] \n"); msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Internal Server Error"); }; exit; }; ... -- However i see in logs following error message and authentication fails, -- ERROR: auth [auth_mod.c:690]: pv_www_authenticate2(): failed to get method value -- Which is indicates that value of $msrp(method) is null. However, as you can see in example code in URL provided above we have an IF condition which explicitly checks $msrp(method) == "AUTH". For the sake of testing i even replaced the variable with actual string value, and still get the same error. So i am guessing it is a bug. What do you guys suggest? Kamailio: v4.1.3 (i386/linux) 236326 MSRP Lib: Crocodile MSRP - v1.0.0 Thank you.

2 17

Download SER SIP server
by Horo, Priyam 12 Jun '14

12 Jun '14

Hi, I cannot find the link to download the SER SIP server. I went to http://www.iptel.org/download, but do not see how to download 0.9.6 version. Could you please point me to the link. Thanks, Priyam.

3 2

check for the number of open branches in Reply
by Sebastian Damm 12 Jun '14

12 Jun '14

Hi, I have a scenario where I want to send a custom SIP package whenever a call is successfully established or definitely missed. This works pretty well, except when a user has multiple devices online. I need to send exactly one packet. When initiating the call I set a flag, in the reply route I check for that flag, and on the first 4xx or 2xx response the packet gets sent out. After that I reset the flag, so no other packet is generated for another response coming in. Now when a user has two devices and rejects the call on the first one but accepts it on the second, I see this call as missed, not answered. So I need to use the first 2xx response or the last 4xx response. For that I need to find out how many branches are open at the moment. But I'm missing something. I registered two devices on the same AOR. I tried accessing $branch(count) in the onreply route, but it is always 0. $br and $bR are <null>. Does anyone have an idea, how to get that information in the reply_route? Best Regards, Sebastian

3 5

planning the release of v4.1.4
by Daniel-Constantin Mierla 12 Jun '14

12 Jun '14

Hello, I am considering to release v4.1.4 next week, in one of the days within the interval June 10-12, depending on work that has to be done. If anyone is aware of issue not reported on tracker or patches that have not been backported, add to the tracker or write a message to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 4

Kamailio accepts and stores XCAP XML document from jitsi but rejects it on retrieval
by Alex Villacís Lasso 11 Jun '14

11 Jun '14

I am trying to set up XCAP support for SIP SIMPLE, and using jitsi as the sample client. The kamailio database was previously configured using ODBC and kamailio-4.1.3 and was apparently working correctly. I was following the guide at http://nil.uniza.sk/instant-messaging/simple/configuring-xcap-support-im-an… as well as the standard documentation at http://kamailio.org/docs/modules/4.1.x/modules/xcap_server.html . However, the rls support is currently disabled. When I use jitsi to connect, and add one contact to the empty contact list, I see the database being populated (mysql backend): mysql> select * from xcap; +----+-------------+-----------------+--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+----------+-----------------------+--------+-----------------------------------------------------------------------+------+ | id | username | domain | doc | doc_type | etag | source | doc_uri | port | +----+-------------+-----------------+--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+----------+-----------------------+--------+-----------------------------------------------------------------------+------+ | 3 | avillacisIM | pbx.elastix.com | <?xml version="1.0" encoding="UTF-8" standalone="no"?><resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"><list name="RootGroup"><entry uri="sip:bmacias@pbx.elastix.com"><display-name>bmacias</display-name></entry></list></resource-lists> | 4 | sr-1401316328-31413-1 | 0 | /xcap-root/resource-lists/users/sip:avillacisIM@pbx.elastix.com/index | 0 | | 4 | avillacisIM | pbx.elastix.com | <?xml version="1.0" encoding="UTF-8" standalone="no"?><ruleset xmlns="urn:ietf:params:xml:ns:common-policy"><rule id="presence_allow"><conditions><identity><one id="sip:bmacias@pbx.elastix.com"/></identity></conditions><actions><sub-handling xmlns="urn:ietf:params:xml:ns:pres-rules">allow</sub-handling></actions><transformations><provide-devices xmlns="urn:ietf:params:xml:ns:pres-rules"><all-devices/></provide-devices><provide-persons xmlns="urn:ietf:params:xml:ns:pres-rules"><all-persons/></provide-persons><provide-services xmlns="urn:ietf:params:xml:ns:pres-rules"><all-services/></provide-services></transformations></rule></ruleset> | 2 | sr-1401316328-31416-1 | 0 | /xcap-root/pres-rules/users/sip:avillacisIM@pbx.elastix.com/presrules | 0 | +----+-------------+-----------------+--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+----------+-----------------------+--------+-----------------------------------------------------------------------+------+ 2 rows in set (0.00 sec) As far as I can see, both XML documents are well formed. However, if I shutdown and then restart jitsi, I see (with wireshark) that the server returned a 500 Server Error when trying to fetch resource-lists . At the same time, the following appears on the log: May 28 17:39:12 elx /usr/sbin/kamailio[31421]: ALERT: <script>: ===== xhttp: request [HTTP/1.1] GET => /xcap-root/xcap-caps/global/index May 28 17:39:12 elx /usr/sbin/kamailio[31421]: ALERT: <script>: ===== xhttp: request [HTTP/1.1] GET => /xcap-root/xcap-caps/global/index May 28 17:39:12 elx /usr/sbin/kamailio[31421]: ALERT: <script>: ===== xhttp: xcap-caps : <null> May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ALERT: <script>: ===== xhttp: request [HTTP/1.1] GET => /xcap-root/resource-lists/users/sip:avillacisIM@pbx.elastix.com/index May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ALERT: <script>: ===== xhttp: request [HTTP/1.1] GET => /xcap-root/resource-lists/users/sip:avillacisIM@pbx.elastix.com/index May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ALERT: <script>: ===== xhttp: resource-lists : sip:avillacisIM@pbx.elastix.com May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ERROR: xcap_server [xcap_server.c:758]: xcaps_get_db_doc(): invalid xml doc retrieved from database May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ERROR: xcap_server [xcap_server.c:1171]: w_xcaps_get(): could not fetch xcap document What can I do to solve this issue. This is preventing contacts from being retrieved or added once the first contact is stored. My event-route looks like this (WITH_XCAPSRV is defined elsewhere in the configuration): #!ifdef WITH_XHTTP event_route[xhttp:request] { #!ifdef WITH_XHTTP_RPC $var(xhttp_rpc_root) = $(hu{s.substr,0,9}); if ($var(xhttp_rpc_root) == "/http_rpc") { dispatch_xhttp_rpc(); } #!endif #!ifdef WITH_XHTTP_PI $var(xhttp_rpc_root) = $(hu{s.substr,0,8}); if ($var(xhttp_rpc_root) == "/http_pi") { dispatch_xhttp_pi(); } #!endif #!ifdef WITH_XCAPSRV if ($hu =~ "^/xcap-root/") { route(XCAPSRV); } #!endif set_reply_close(); set_reply_no_connect(); # if ($Rp != 80 ##!ifdef WITH_TLS # && $Rp != 443 ##!endif # ) { # # xlog("L_WARN", "HTTP request received on $Rp\n"); # xhttp_reply("403", "Forbidden", "text/html", "Forbidden"); # exit; # } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } #!endif #!ifdef WITH_XCAPSRV route[XCAPSRV] { xlog("L_ALERT", "===== xhttp: request [$rv] $rm => $hu\n"); # Remove port specification from Host header and use as authentication domain $var(http_domain) = $(hdr(Host){s.select,0,:}); if (!www_authorize("$var(http_domain)", "subscriber")) { www_challenge("$var(http_domain)", "0"); exit; } set_reply_close(); set_reply_no_connect(); if ($hu=~"^/xcap-root/oma_status-icon/users/.*/.*$") $var(doc_uri) = $(hu{re.subst,/(^\/xcap-root\/)oma_status-icon(\/users\/.*\/).*$/\1org.openmobilealliance.pres-content\2oma_status-icon\/preseImage/}); else $var(doc_uri) = $hu; # xcap ops $xcapuri(u=>data) = $var(doc_uri); if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else if($xcapuri(u=>xuid)=~".+@.+") $var(uri) = "sip:" + $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; xlog("L_ALERT", "===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); if ($xcapuri(u=>auid) == "xcap-caps") { if ($rm == "GET") { $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> <auid>org.openmobilealliance.user-profile</auid> <auid>org.openmobilealliance.pres-content</auid> <auid>org.openmobilealliance.search</auid> <auid>org.openmobilealliance.xcap-directory</auid> </auids> <extensions></extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> <namespace>urn:oma:xml:xdm:user-profile</namespace> <namespace>urn:oma:xml:prs:pres-content</namespace> <namespace>urn:oma:xml:xdm:search</namespace> <namespace>urn:oma:xml:xdm:xcap-directory</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "OK", "application/xcap-caps+xml", "$var(xbody)"); } else { append_to_reply("Allow: GET\r\n"); xhttp_reply("405", "Method Not Allowed", "", ""); } exit; } # be sure auth user access only its documents if ($au != $(var(uri){uri.user})) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>$si:$sp</body></html>"); exit; } switch($rm) { case "PUT": xcaps_put("$var(uri)", "$var(doc_uri)", "$rb"); if ($xcapuri(u=>auid) =~ "pres-rules") { pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); # } else if ($xcapuri(u=>auid) =~ "rls-services" || $xcapuri(u=>auid) =~ "resource-lists") { # rls_update_subs("$var(uri)", "presence"); } else if ($xcapuri(u=>auid) =~ "pidf-manipulation") { pres_refresh_watchers("$var(uri)", "presence", 2, "$xcapuri(u=>uri_adoc)", "$xcapuri(u=>file)"); } exit; break; case "GET": xcaps_get("$var(uri)", "$var(doc_uri)"); exit; break; case "DELETE": xcaps_del("$var(uri)", "$var(doc_uri)"); if ($xcapuri(u=>auid) =~ "pres-rules") { pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); # } else if ($xcapuri(u=>auid) =~ "rls-services" || $xcapuri(u=>auid) =~ "resource-lists") { # rls_update_subs("$var(uri)", "presence"); } else if ($xcapuri(u=>auid) =~ "pidf-manipulation") { pres_refresh_watchers("$var(uri)", "presence", 2, "$xcapuri(u=>uri_adoc)", "$xcapuri(u=>file)"); } exit; break; case "POST": if ($xcapuri(u=>auid) =~ "search") { xhttp_reply("501", "Not Implemented", "", ""); } else { if ($xcapuri(u=>auid) =~ "xcap-directory") { append_to_reply("Allow: GET\r\n"); } else { append_to_reply("Allow: DELETE, GET, PUT\r\n"); } xhttp_reply("405", "Method Not Allowed", "", ""); } exit; break; } # other http requests xhttp_reply("404", "Not Found", "", ""); exit; } #!endif

2 11

Re: [SR-Users] Kamailio and Asterisk
by Carlos Rangel 11 Jun '14

11 Jun '14

Found the problem per a previous post. I had uncommented in kamctlrc by mistake. #STANDARD_MODULES="standard acc lcr domain group permissions registrar usrloc msilo # alias_db uri_db speeddial avpops auth_db pdt dialog dispatcher # dialplan" Thanks CR De: Carlos Rangel [mailto:crangel@globaltelesourcing.com] Enviado el: miércoles, 11 de junio de 2014 02:19 p.m. Para: sr-users(a)lists.sip-router.org Asunto: Kamailio and Asterisk Hello Trying to do my first Kamailio and Asterisk integration following the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I am using Centos though. Everything goes well except when trying to create the kamailio DB. The core tables cannot be created. Any ideas how to solve this problem??. Thanks CR [root@gt-ev-i08 kamailio-4.1.3]# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset INFO: creating database kamailio ... INFO: granting privileges to database kamailio ... INFO: creating standard tables into kamailio ... /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql: line 264: /usr/local/share/kamailio//mysql/#-create.sql: No such file or directory ERROR: Creating core tables failed at #!

1 0

Kamailio and Asterisk
by Carlos Rangel 11 Jun '14

11 Jun '14

Hello Trying to do my first Kamailio and Asterisk integration following the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I am using Centos though. Everything goes well except when trying to create the kamailio DB. The core tables cannot be created. Any ideas how to solve this problem??. Thanks CR [root@gt-ev-i08 kamailio-4.1.3]# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset INFO: creating database kamailio ... INFO: granting privileges to database kamailio ... INFO: creating standard tables into kamailio ... /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql: line 264: /usr/local/share/kamailio//mysql/#-create.sql: No such file or directory ERROR: Creating core tables failed at #!

1 0

kamailio and app_python
by Victor Seva 11 Jun '14

11 Jun '14

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

4 6

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

11 Jun '14

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

2 1

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

11 Jun '14

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

2 1

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

11 Jun '14

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

2 4

Location issue
by Igor Potjevlesch 11 Jun '14

11 Jun '14

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

2 6

Kamailio and Freepbx Integration
by Carlos Rangel 10 Jun '14

10 Jun '14

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

1 0

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

10 Jun '14

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

2 8

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

10 Jun '14

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

2 1

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

09 Jun '14

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

2 1

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

09 Jun '14

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

2 1

Kamailio on PPC
by ashwin 09 Jun '14

09 Jun '14

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

2 1

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

09 Jun '14

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

2 1

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

09 Jun '14

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

2 1

Branch feedback behaviour
by Alex Balashov 08 Jun '14

08 Jun '14

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

2 3

SDP and NAT
by Marc Soda 06 Jun '14

06 Jun '14

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

2 1

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

06 Jun '14

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

1 1

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

06 Jun '14

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

1 0

CANCEL requests
by Andrea Meroni 06 Jun '14

06 Jun '14

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

2 2

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

06 Jun '14

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

2 1

FS#439 - CRASH: segmentation fault after INVITE
by Nuno Reis 05 Jun '14

05 Jun '14

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

1 0

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

05 Jun '14

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

3 8

MySQL query result pagination
by Allen Zhang 05 Jun '14

05 Jun '14

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

2 4

Re: [SR-Users] MySQL query result pagination
by Allen Zhang 04 Jun '14

04 Jun '14

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

1 0

CNXCC and multiple servers
by Kelvin Chua 04 Jun '14

04 Jun '14

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

2 1

mediaproxy
by Slava Bendersky 03 Jun '14

03 Jun '14

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

3 3

cxdx_forced_peer - more info
by Daniel Ciprus 03 Jun '14

03 Jun '14

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

2 2

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

03 Jun '14

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

1 1

ODP: cnxcc
by Andrzej Ciupek 03 Jun '14

03 Jun '14

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

3 10

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

03 Jun '14

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

1 0

sqlops
by Kelvin Chua 03 Jun '14

03 Jun '14

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

2 2

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

02 Jun '14

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

3 3

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

02 Jun '14

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

2 5

drouting failure loop
by Pete Ashdown 02 Jun '14

02 Jun '14

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

2 1

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

01 Jun '14

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

2 1

Reason header body to acc_sip_reason_column
by Juha Heinanen 30 May '14

30 May '14

if response has reason header, i would like to gets its body to acc_sip_reason_column field rather than the reason on the response status line which does not bring any added value. any comments on how to achieve that? -- juha Session Initiation Protocol Status-Line: SIP/2.0 404 Not Found ... Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"

2 4

Use of received=X.X.X.X param in Contact header of REGISTER message
by Luis F Urrea 30 May '14

30 May '14

Hi all, I have an existing Kamailio installation that is handling registrations and NAT traversal for clients. There is one particular build of softphone that sends us a received=X.X.X.X param in its REGISTER message and it seems that the current Kamailio's config in place always uses that address to store the AOR. It happens to be that the client appears to be misbehaving and sending an outdated IP address on that field. My question is where in the NAT traversal functions do I have control over using this param or not. TIA

2 2

Problems migrating from rtpproxy/rtoproxy.so to rtpproxy-ng/rtpengine
by LAA 30 May '14

30 May '14

On 05/28/14 13:31, LAA wrote: >>* Hi all, *>> >>* I'm currently running a pilot with kamailio 4.1.3 stable, and I would *>>* like to test WebRTC Capabilities. Websockets Support is runnig OK, and *>>* now I'm trying to deal with calls between WebTRC and legacy softphones. *>>* I have installed rtpengine (as it a replacement for old mediaproxy-ng), *>>* and it is running: *>> >>* rtpengine --table=0 --ip=192.168.0.197 --listen-udp=127.0.0.1:12222 <http://127.0.0.1:12222> * >You need to use the --listen-ng=... option instead of listen-udp for >this to work. >cheers Thanks Richard, of course it works. Regards. Luis. > > > > > >

1 0

Kamailio on P2020RDB-PCA
by ashwin 30 May '14

30 May '14

Hi, I tried to compile Kamailio on Freescale P2020RDB-PCA.And I'm getting the following error bison -d -b cfg cfg.y make: *** [cfg.tab.h] Broken Pipe Please give recommendations to solve the error. Thanks -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-on-P2020RDB-PCA-tp128073.h… Sent from the Users mailing list archive at Nabble.com.

2 2

Problems migrating from rtpproxy/rtoproxy.so to rtpproxy-ng/rtpengine
by LAA 28 May '14

28 May '14

Hi all, I'm currently running a pilot with kamailio 4.1.3 stable, and I would like to test WebRTC Capabilities. Websockets Support is runnig OK, and now I'm trying to deal with calls between WebTRC and legacy softphones. I have installed rtpengine (as it a replacement for old mediaproxy-ng), and it is running: rtpengine --table=0 --ip=192.168.0.197 --listen-udp=127.0.0.1:12222 As a first step, I'm trying to migrate the standard NAT handling configuration to the new module. (...) #loadmodule "rtpproxy.so" loadmodule "rtpproxy-ng.so" (...) #modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7624") modparam("rtpproxy-ng", "rtpproxy_sock", "udp:127.0.0.1:12222") NAT hanling is not working, and I catched this message from /var/log/syslog May 28 18:55:35 kamailio rtpengine[2474]: Failed to properly parse UDP command line '4547_3 d7:command4:pinge' from 127.0.0.1:60874, using fallback RE May 28 18:55:35 kamailio rtpengine[2474]: Failed to properly parse UDP command line '4545_4 d7:command4:pinge' from 127.0.0.1:57236, using fallback RE And from Kamailio's log: May 28 19:23:47 kamailio /usr/local/sbin/kamailio[4868]: ERROR: rtpproxy-ng [rtpproxy.c:1425]: rtpp_test(): proxy responded with invalid response May 28 19:23:47 kamailio /usr/local/sbin/kamailio[4869]: ERROR: rtpproxy-ng [rtpproxy.c:1425]: rtpp_test(): proxy responded with invalid response Is rtpproxy-ng module compatible with rtpengine? Best regards Luis.

2 1

Interpreting messages in log
by Gilbert T. Gutierrez, Jr. 28 May '14

28 May '14

Can someone help me interpret what I a seeing in my log? Gilbert May 28 09:53:09 faxsvr /usr/sbin/kamailio[1811]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=521 a=5 n=route May 28 09:53:09 faxsvr /usr/sbin/kamailio[1811]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=665 a=16 n=if May 28 09:53:09 faxsvr /usr/sbin/kamailio[1811]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=625 a=24 n=has_totag May 28 09:53:09 faxsvr /usr/sbin/kamailio[1811]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=525 a=24 n=t_check_trans May 28 09:53:09 faxsvr /usr/sbin/kamailio[1811]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=528 a=5 n=route May 28 09:53:09 faxsvr /usr/sbin/kamailio[1811]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=797 a=16 n=if May 28 09:53:09 faxsvr /usr/sbin/kamailio[1811]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=784 a=25 n=is_method May 28 09:53:09 faxsvr /usr/sbin/kamailio[1811]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=792 a=16 n=if May 28 09:53:09 faxsvr /usr/sbin/kamailio[1811]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=787 a=27 n=auth_check May 28 09:53:09 faxsvr /usr/sbin/kamailio[1811]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=788 a=26 n=auth_challenge May 28 09:53:09 faxsvr /usr/sbin/kamailio[1811]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=789 a=2 n=exit

2 2

Has Kamailio 4.1.3 ability like Asterisk permit/deny
by Yasin Caner 28 May '14

28 May '14

Hello all; After wasting my time to search asterisk ability permit/deny ip for Kamailio , i decided to consult you. Is there a module or a way to do permit/deny ip for users. As you know asterisk can do that easily . I found permisson module and permisson.conf but it doesn't work with db and online , so do i write my code with my rules or is there a module to do it? Thanks for helps.

2 1

Help with load balancing Kamalio based on DNS
by Veerabhara Gundu 28 May '14

28 May '14

Hi All, This question might have addressed, but not able to find correct answer in archives and I have spent enough time, so posting here. I want to load balance between 2-3 kamailio servers based on DNS, I don't need to worry about media just trying IM with SIP messages. Configured two servers with same domain alias(example.com) and able to send messages between users registered to one Node. But when i tried to send messages between users registered to two different servers, messages are not getting routed properly. When Server A tried to reach user registered to Server:B directly, firewall is not allowing. Tried to route message to correct proxy based on the socket details in Loctaion table, but not able to read socket details from location table, I am getting always NULL. I am sure,I am doing something wrong with configuration, but not sure and also not able to find example for this on Kamailio website. I am attaching my configuration, appreciate any help. Thanks, VG

2 1

Crash Kamailio 4.1.3
by Igor Potjevlesch 28 May '14

28 May '14

hello, Kamailio 4.1.3 crashed today, with an segmentation fault error : kernel: kamailio[2406]: segfault at 18 ip 0000000000495792 sp 00007fff3d9d2d80 error 4 in kamailio[400000+275000] Here is the result of BT : #0 0x0000000000495792 in pv_get_strval (msg=0x7fad2cf8a530, param=0x7fad3858c288, res=0x7fff3d9d2eb0, sval=0x18) at pvapi.c:521 #1 0x00007fad365b0da8 in pv_get_pai (msg=0x7fad2cf8a530, param=0x7fad3858c288, res=0x7fff3d9d2eb0) at pv_core.c:1021 #2 0x0000000000499518 in pv_get_spec_value (msg=0x7fad2cf8a530, sp=0x7fad3858c270, value=0x7fff3d9d2eb0) at pvapi.c:1266 #3 0x00007fad34c6d7d9 in extra2strar (extra=0x7fad3858c260, rq=0x7fad2cf8a530, val_arr=0x7fad34e7ae30, int_arr=0x7fad34e7b37c,type_arr=0x7fad34e7b4e7 "\002\002\002\002") at acc_extra.c:261 #4 0x00007fad34c65393 in acc_db_request (rq=0x7fad2cf8a530) at acc.c:474 #5 0x00007fad34c6fb43 in acc_onreply (t=0x7fad2ce0b138, req=0x7fad2cf8a530, reply=0x7fad3859faa8, code=200) at acc_logic.c:471 #6 0x00007fad34c7008d in tmcb_func (t=0x7fad2ce0b138, type=512, ps=0x7fff3d9d3120) at acc_logic.c:559 #7 0x00007fad36e6a478 in run_trans_callbacks_internal (cb_lst=0x7fad2ce0b1a8, type=512, trans=0x7fad2ce0b138, params=0x7fff3d9d3120) at t_hooks.c:290 #8 0x00007fad36e6a68a in run_trans_callbacks_with_buf (type=512, rbuf=0x7fad2ce0b1f8, req=0x7fad2cf8a530, repl=0x7fad3859faa8, flags=183) at t_hooks.c:336 #9 0x00007fad36e9cc06 in relay_reply (t=0x7fad2ce0b138, p_msg=0x7fad3859faa8, branch=0, msg_status=183, cancel_data=0x7fff3d9d3480, do_put_on_wait=1) at t_reply.c:2001 #10 0x00007fad36e9f0b7 in reply_received (p_msg=0x7fad3859faa8) at t_reply.c:2499 #11 0x000000000045d837 in do_forward_reply (msg=0x7fad3859faa8, mode=0) at forward.c:777 #12 0x000000000045e0f8 in forward_reply (msg=0x7fad3859faa8) at forward.c:860 #13 0x00000000004a5887 in receive_msg ( buf=0x924600 "SIP/2.0 183 Session Progress\r\nVia: SIP/2.0/UDP a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0;received=a.b.c.d\r\nVia: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5"..., len=872, rcv_info=0x7fff3d9d3800) at receive.c:273 #14 0x000000000053c838 in udp_rcv_loop () at udp_server.c:536 #14 0x000000000053c838 in udp_rcv_loop () at udp_server.c:536 #15 0x000000000046d42b in main_loop () at main.c:1617 #16 0x00000000004704d3 in main (argc=7, argv=0x7fff3d9d3b38) at main.c:2533 Many thanks. Regards, Igor.

2 3

Possible causes of calls being terminated (ACK packet never received)
by Alex Villacís Lasso 27 May '14

27 May '14

I am trying to diagnose a SIP issue between our carrier and our network. The carrier has a CARRIER_IP and a different CARRIER_MEDIA_IP, and it submits an INVITE packet to MY_PUBLIC_IP using MY_DID. The firewall at our public IP (where the attached traffic sample was taken) redirects it to a particular Kamailio server at 192.168.10.10 inside the LAN, and it in turn routes it to the Asterisk instance in localhost. The issue is that the INVITE is received, then the SIP/2.0 100 Trying and then SIP/2.0 200 OK are routed back (or so I think), and then the expected ACK from is never received, even though the caller already hears the media from the Asterisk IVR. After a timeout, our Asterisk closes the call, in the middle of the conversation. I am trying to explain the situation to our carrier, but I want to rule out possible misconfigurations on our side. Are there common misconfigurations that produce the symptoms described here? Are there any issues evident from the attached traffic?

3 7

Kamailio and MSILO with TLS AORs
by Roberto Fichera 27 May '14

27 May '14

Hi All, I've started to play with MSILO module on kamailio v4.1.2, but it seems that all the stored messages aren't delivered to the give user even if, looking at the debug log, the given messages are found and queued for shipping. Another info I can add is that all the users are connected in TLS. The relevant conf is below. Doe anyone can suggest a working setup for TLS users? Thanks in advance, Roberto Fichera. # ----- msilo params ----- modparam("msilo", "db_url", DBURL) modparam("msilo", "from_address", "sip:$rU@test.net") modparam("msilo", "contact_hdr", "Contact: msilo@test.net:5060;msilo=yes\r\n") modparam("msilo", "content_type_hdr", "Content-Type: text/plain\r\n") # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } # MSILO - dumping user's offline messages if (m_dump()) { xlog("MSILO: offline messages dumped - if they were for user $fu\n"); } else { xlog("MSILO: no offline messages dumped\n"); } exit; } } and in the route[LOCATION] I've this snippet xlog( "Checking the destination user $rU exists\n"); if(avp_db_query("select username from subscriber where username='$rU'")) { xlog("MESSAGE received $ru -> storing using MSILO user=$rU\n"); if(m_store("$ru")) { xlog("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); } } # if the downstream UA doesn't support MESSAGE requests # go to failure_route[1] t_on_failure("1"); exit; } else { # The destination recipiend doen't exist xlog( "Unable to store message for $ru because the user doesn't exist" ); if (!t_reply("404", "Not Found")) { sl_reply_error(); } exit; } }

3 20

Authentication header onfailure_route can not read - Remote Registartion
by Yasin Caner 27 May '14

27 May '14

Hey all; I am trying to remote registration between Kamailio and Asterisk. I tried remote Registration with UAC module. But this module doesn't work online. So that i wrote my remote registration code, is below. But there is a little problem that i cant take Authentication header to onfailure_route so that cannot forward Auth to client. it is working well with uac_auth but gets this error (" run_failure_handlers(): ERROR: run_failure_handlers: Error in run_top_route ") so cannot trust this function. How can i get Auth header to add Thanks for helps. My Code: route[REMOTEREG]{ xlog("L_INFO","Remote Reg: $fd : $rm : $fU : $si : "); sql_xquery("ca","CALL kamailio.route_3xx( '$fd', '$rm'); ","ra"); xlog("L_INFO","Remote Reg:ip_addr:$xavp(ra=>ip_addr): \n"); if($xavp(ra=>ip_addr)!=$null){ $avp(ipmiz)=$xavp(ra=>ip_addr); xlog("L_INFO","pump it:$rU:$avp(ipmiz):$ru:$rd"); $ru="sip:"+$fU+"@"+$avp(ipmiz); xlog("L_INFO","pump it :new::$avp(ipmiz)"); t_on_failure("REMOTE_AUTH"); t_on_reply("REGISTER_SUC"); if(!t_relay()){ t_reply("408", "have a problem"); exit; } exit; } return; } failure_route[REMOTE_AUTH]{ xlog("L_INFO","REMOTE_AUTH:$fU:$si$rm:$ci:$adu:ar:$ar"); if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_NOTICE", "Remote asked for authentication "); #uac_auth(); ## need to add Authentication header. append_to_reply("Allow:$hdr(Allow) \r\n"); t_reply("$T_reply_code","Unauthorized"); } } onreply_route[REGISTER_SUC]{ xlog("L_INFO","REGISTER_SUC"); if(t_check_status("200")){ if (!save("location")) { xlog("L_NOTICE", "Register yapilamadi"); } exit; } } Request-Line: REGISTER sip:K1;transport=UDP SIP/2.0 Message Header Via: SIP/2.0/UDP U1:61185;branch=z9hG4bK-d8754z-597229c1c2842862-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:At1@U1:61185;rinstance=0e0e0d00e8aa79f1;transport=UDP> To: "At1"<sip:At1@K1;transport=UDP> From: "At1"<sip:At1@K1;transport=UDP>;tag=cc6f1214 Call-ID: MjYwOGQ0MTRmZTEwMzcxNDc2YjBhM2FlYTU3MjU5OWE. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.0.19632 r19608 Allow-Events: presence, kpml Content-Length: 0 Request-Line: REGISTER sip:At1@UAS1 SIP/2.0 Message Header Via: SIP/2.0/UDP K1;branch=z9hG4bKc19c.d20e7767bea000b39f25e178c4831a2b.0 Via: SIP/2.0/UDP U1:61185;branch=z9hG4bK-d8754z-597229c1c2842862-1---d8754z-;rport=61185 Max-Forwards: 70 Contact: <sip:At1@U1:61185;rinstance=0e0e0d00e8aa79f1;transport=UDP> To: "At1"<sip:At1@K1;transport=UDP> From: "At1"<sip:At1@K1;transport=UDP>;tag=cc6f1214 Call-ID: MjYwOGQ0MTRmZTEwMzcxNDc2YjBhM2FlYTU3MjU5OWE. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.0.19632 r19608 Allow-Events: presence, kpml Content-Length: 0 Message Header Via: SIP/2.0/UDP K1;branch=z9hG4bKc19c.d20e7767bea000b39f25e178c4831a2b.0;received=K1;rport=5060 Via: SIP/2.0/UDP U1:61185;branch=z9hG4bK-d8754z-597229c1c2842862-1---d8754z-;rport=61185 From: "At1"<sip:At1@K1;transport=UDP>;tag=cc6f1214 To: "At1"<sip:At1@K1;transport=UDP>;tag=as63346477 Call-ID: MjYwOGQ0MTRmZTEwMzcxNDc2YjBhM2FlYTU3MjU5OWE. CSeq: 1 REGISTER Server: Thirdlane2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="aaaa", nonce="1b23af8f" Content-Length: 0 Status-Line: SIP/2.0 401 Unauthorized Message Header Via: SIP/2.0/UDP U1:61185;branch=z9hG4bK-d8754z-597229c1c2842862-1---d8754z-;rport=61185 To: "At1"<sip:At1@K1;transport=UDP>;tag=a98c7436060b0c2b493a17bc0f200923-ed9e From: "At1"<sip:At1@K1;transport=UDP>;tag=cc6f1214 Call-ID: MjYwOGQ0MTRmZTEwMzcxNDc2YjBhM2FlYTU3MjU5OWE. CSeq: 1 REGISTER Allow:INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Server: NetGSM Content-Length: 0

2 1

Authentication header onfailure_route can not read - Remote Registartion
by Yasin Caner 26 May '14

26 May '14

Hey all; I am trying to remote registration between Kamailio and Asterisk. I tried remote Registration with UAC module. But this module doesn't work online. So that i wrote my remote registration code, is below. But there is a little problem that i cant take Authentication header to onfailure_route so that cannot forward Auth to client. it is working well with uac_auth but gets this error (" run_failure_handlers(): ERROR: run_failure_handlers: Error in run_top_route ") so cannot trust this function. How can i get Auth header to add Thanks for helps. My Code: route[REMOTEREG]{ xlog("L_INFO","Remote Reg: $fd : $rm : $fU : $si : "); sql_xquery("ca","CALL kamailio.route_3xx( '$fd', '$rm'); ","ra"); xlog("L_INFO","Remote Reg:ip_addr:$xavp(ra=>ip_addr): \n"); if($xavp(ra=>ip_addr)!=$null){ $avp(ipmiz)=$xavp(ra=>ip_addr); xlog("L_INFO","pump it:$rU:$avp(ipmiz):$ru:$rd"); $ru="sip:"+$fU+"@"+$avp(ipmiz); xlog("L_INFO","pump it :new::$avp(ipmiz)"); t_on_failure("REMOTE_AUTH"); t_on_reply("REGISTER_SUC"); if(!t_relay()){ t_reply("408", "have a problem"); exit; } exit; } return; } failure_route[REMOTE_AUTH]{ xlog("L_INFO","REMOTE_AUTH:$fU:$si$rm:$ci:$adu:ar:$ar"); if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_NOTICE", "Remote asked for authentication "); #uac_auth(); ## need to add Authentication header. append_to_reply("Allow:$hdr(Allow) \r\n"); t_reply("$T_reply_code","Unauthorized"); } } onreply_route[REGISTER_SUC]{ xlog("L_INFO","REGISTER_SUC"); if(t_check_status("200")){ if (!save("location")) { xlog("L_NOTICE", "Register yapilamadi"); } exit; } } Request-Line: REGISTER sip:K1;transport=UDP SIP/2.0 Message Header Via: SIP/2.0/UDP U1:61185;branch=z9hG4bK-d8754z-597229c1c2842862-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:At1@U1:61185;rinstance=0e0e0d00e8aa79f1;transport=UDP> To: "At1"<sip:At1@K1;transport=UDP> From: "At1"<sip:At1@K1;transport=UDP>;tag=cc6f1214 Call-ID: MjYwOGQ0MTRmZTEwMzcxNDc2YjBhM2FlYTU3MjU5OWE. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.0.19632 r19608 Allow-Events: presence, kpml Content-Length: 0 Request-Line: REGISTER sip:At1@UAS1 SIP/2.0 Message Header Via: SIP/2.0/UDP K1;branch=z9hG4bKc19c.d20e7767bea000b39f25e178c4831a2b.0 Via: SIP/2.0/UDP U1:61185;branch=z9hG4bK-d8754z-597229c1c2842862-1---d8754z-;rport=61185 Max-Forwards: 70 Contact: <sip:At1@U1:61185;rinstance=0e0e0d00e8aa79f1;transport=UDP> To: "At1"<sip:At1@K1;transport=UDP> From: "At1"<sip:At1@K1;transport=UDP>;tag=cc6f1214 Call-ID: MjYwOGQ0MTRmZTEwMzcxNDc2YjBhM2FlYTU3MjU5OWE. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.0.19632 r19608 Allow-Events: presence, kpml Content-Length: 0 Message Header Via: SIP/2.0/UDP K1;branch=z9hG4bKc19c.d20e7767bea000b39f25e178c4831a2b.0;received=K1;rport=5060 Via: SIP/2.0/UDP U1:61185;branch=z9hG4bK-d8754z-597229c1c2842862-1---d8754z-;rport=61185 From: "At1"<sip:At1@K1;transport=UDP>;tag=cc6f1214 To: "At1"<sip:At1@K1;transport=UDP>;tag=as63346477 Call-ID: MjYwOGQ0MTRmZTEwMzcxNDc2YjBhM2FlYTU3MjU5OWE. CSeq: 1 REGISTER Server: Thirdlane2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="aaaa", nonce="1b23af8f" Content-Length: 0 Status-Line: SIP/2.0 401 Unauthorized Message Header Via: SIP/2.0/UDP U1:61185;branch=z9hG4bK-d8754z-597229c1c2842862-1---d8754z-;rport=61185 To: "At1"<sip:At1@K1;transport=UDP>;tag=a98c7436060b0c2b493a17bc0f200923-ed9e From: "At1"<sip:At1@K1;transport=UDP>;tag=cc6f1214 Call-ID: MjYwOGQ0MTRmZTEwMzcxNDc2YjBhM2FlYTU3MjU5OWE. CSeq: 1 REGISTER Allow:INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Server: NetGSM Content-Length: 0

1 0

Kamailio Replies in setup with LB
by Andrea Meroni 26 May '14

26 May '14

Hi all, i have the following setup: C -> LB-P -> K-P -> LB-S -> K-S The problem is the following: when K-S (Kamailio Server) replies it relays the responses to the K-P (Kamailio Proxy) instead of the LB-P (load-balancer of the proxies). I tried almost every solution: changing aliases, adding Path headers, changing destination-URI, changing Via: headers ... But the result is always the same: using transactions it seems that, even if requests are correctly load-balanced, the replies are not relayed back following the same path (in particular, referring to the precedent scheme, a reply for the previous request will follow the path C <- K-P <- K-S). Additional information: - K-P uses as alias the HOSTNAME of LB-P; - K-S uses as alias the HOSTNAME of LB-S; Everyone has some hints? Andrea

1 0

RTPProxy timeout
by Igor Potjevlesch 26 May '14

26 May '14

Hello, I can't figure out how the "timeout_socket" should be configured. I try with the default parameter in the doc: xmlrpc:http://127.0.0.1:8000/RPC2 I launch RTPProxy like that: -n tcp:127.0.0.1:8000 RTPProxy socket is setup like: "rtpproxy_sock", "udp:127.0.0.1:7722". It looks like there is no communication on the port 8000. The goal is to setup a timeout in case of asymmetric RTP. Many help would be appreciate. Thank you! Regards, Igor.

2 5

Kamailio LCR issue
by José Seabra 26 May '14

26 May '14

Hello there, I'm configuring the LCR module, with more than one lcr instance, for example each instance id will have atleast 1 diferent gateway, well the issue here is that if i configure another gateways with diferent LCR_ID and then run kamcmd lcr.reload then kamcmd lcr.dump_gws i only see the gateways that belongs to the instance 1, should it show me the all gateways for each lcr instance correct? I'm using kamailio-4.0.4 please have a look at my lcr configuration example and the kamailio logs: Mysql: mysql> select * from lcr_gw; +----+--------+---------+-------------+-------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | prefix | tag | flags | defunct | +----+--------+---------+-------------+-------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | 1 | 1 | teste1 | 192.168.1.1 | 192.168.1.1 | 5060 | NULL | 1 | 1 | NULL | NULL | NULL | 0 | NULL | | 2 | 2 | teste2 | 192.168.1.2 | 192.168.1.2 | 5060 | NULL | 1 | 1 | NULL | NULL | NULL | 0 | NULL | +----+--------+---------+-------------+-------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ 2 rows in set (0.01 sec) mysql> select * from lcr_rule; +----+--------+--------+----------+-------------+---------+---------+ | id | lcr_id | prefix | from_uri | request_uri | stopper | enabled | +----+--------+--------+----------+-------------+---------+---------+ | 1 | 1 | NULL | NULL | 256* | 0 | 1 | | 2 | 2 | NULL | NULL | 236* | 0 | 1 | +----+--------+--------+----------+-------------+---------+---------+ 2 rows in set (0.00 sec) mysql> select * from lcr_rule_target; +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 1 | 1 | 1 | 1 | 1 | 10 | | 2 | 2 | 2 | 2 | 1 | 10 | +----+--------+---------+-------+----------+--------+ 2 rows in set (0.00 sec) kamcd lcr.dump_gws kamcmd lcr.dump_gws 12(3267) DEBUG: ctl [../../io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x7ffb90811c00, 9, 3, 0x1ca0880), fd_no=1 12(3267) DEBUG: ctl [io_listener.c:453]: handle_new_connect(): handle_stream read: new connection (1) on /tmp/kamailio_ctl { lcr_id: 1 gw_id: 1 gw_index: 1 gw_name: teste1 scheme: sip ip_addr: 192.168.1.1 hostname: 192.168.1.1 port: 5060 params: transport: UDP strip: 0 prefix: tag: flags: 0 defunct_until: 0 } 12(3267) DEBUG: ctl [io_listener.c:506]: handle_stream_read(): handle_stream read: eof on /tmp/kamailio_ctl 12(3267) DEBUG: ctl [../../io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x7ffb90811c00, 9, -1, 0x10) fd_no=2 called kamcmd lcr.reload logs: kamcmd lcr.reload 12(3267) DEBUG: ctl [../../io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x7ffb90811c00, 9, 3, 0x1ca0880), fd_no=1 12(3267) DEBUG: ctl [io_listener.c:453]: handle_new_connect(): handle_stream read: new connection (1) on /tmp/kamailio_ctl 12(3267) DEBUG: <core> [db.c:318]: db_do_init2(): connection 0x7ffb95b50ff0 not found in pool 12(3267) DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening connection: mysql://xxxx:xxxx@localhost/kamailio 12(3267) DEBUG: db_mysql [km_my_con.c:121]: db_mysql_new_connection(): connection type is Localhost via UNIX socket 12(3267) DEBUG: db_mysql [km_my_con.c:122]: db_mysql_new_connection(): protocol version is 10 12(3267) DEBUG: db_mysql [km_my_con.c:123]: db_mysql_new_connection(): server version is 5.1.69 12(3267) DEBUG: <core> [db_res.c:118]: db_new_result(): allocate 56 bytes for result set at 0x7ffb95b51160 12(3267) DEBUG: db_mysql [km_res.c:68]: db_mysql_get_columns(): 6 columns returned from the query 12(3267) DEBUG: <core> [db_res.c:155]: db_allocate_columns(): allocate 48 bytes for result names at 0x7ffb95b4e3b0 12(3267) DEBUG: <core> [db_res.c:165]: db_allocate_columns(): allocate 24 bytes for result types at 0x7ffb95b51240 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[0] at 0x7ffb95b51280 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51280)[0]=[id] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[1] at 0x7ffb95b512b0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b512b0)[1]=[prefix] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[2] at 0x7ffb95b512e0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b512e0)[2]=[from_uri] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[3] at 0x7ffb95b51310 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51310)[3]=[stopper] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[4] at 0x7ffb95b511c0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b511c0)[4]=[enabled] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[5] at 0x7ffb95b51340 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51340)[5]=[request_uri] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_dbase.c:358]: db_mysql_fetch_result(): converting row 0 of 1 count 1 12(3267) DEBUG: <core> [db_row.c:119]: db_allocate_row(): allocate 192 bytes for row values at 0x7ffb95b49280 12(3267) DEBUG: <core> [db_val.c:73]: db_str2val(): converting INT [1] 12(3267) DEBUG: <core> [db_val.c:56]: db_str2val(): converting NULL value12(3267) DEBUG: <core> [db_val.c:56]: db_str2val(): converting NULL value12(3267) DEBUG: <core> [db_val.c:73]: db_str2val(): converting INT [0] 12(3267) DEBUG: <core> [db_val.c:73]: db_str2val(): converting INT [1] 12(3267) DEBUG: <core> [db_val.c:117]: db_str2val(): converting STRING [256*] 12(3267) DEBUG: lcr [lcr_mod.c:1238]: reload_tables(): loading, cycle 0 with <1> rows 12(3267) DEBUG: lcr [hash.c:89]: rule_hash_table_insert(): inserted rule_id <1>, prefix <>, from_uri <>, request_uri <256*>, stopper <0>, into index <0> 12(3267) DEBUG: lcr [hash.c:104]: rule_hash_table_insert(): inserted rule_id <1> addr <0x7ffb8dbd5480> into rule_id hash table index <1> 12(3267) DEBUG: <core> [db_res.c:54]: db_free_rows(): freeing 1 rows 12(3267) DEBUG: <core> [db_row.c:97]: db_free_row(): freeing row values at 0x7ffb95b49280 12(3267) DEBUG: <core> [db_res.c:62]: db_free_rows(): freeing rows at 0x7ffb95b513a0 12(3267) DEBUG: <core> [db_res.c:81]: db_free_columns(): freeing 6 columns 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[0] at 0x7ffb95b51280 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[1] at 0x7ffb95b512b0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[2] at 0x7ffb95b512e0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[3] at 0x7ffb95b51310 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[4] at 0x7ffb95b511c0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[5] at 0x7ffb95b51340 12(3267) DEBUG: <core> [db_res.c:94]: db_free_columns(): freeing result names at 0x7ffb95b4e3b0 12(3267) DEBUG: <core> [db_res.c:99]: db_free_columns(): freeing result types at 0x7ffb95b51240 12(3267) DEBUG: <core> [db_res.c:136]: db_free_result(): freeing result set at 0x7ffb95b51160 12(3267) DEBUG: <core> [db_res.c:118]: db_new_result(): allocate 56 bytes for result set at 0x7ffb95b51160 12(3267) DEBUG: db_mysql [km_res.c:68]: db_mysql_get_columns(): 13 columns returned from the query 12(3267) DEBUG: <core> [db_res.c:155]: db_allocate_columns(): allocate 104 bytes for result names at 0x7ffb95b49280 12(3267) DEBUG: <core> [db_res.c:165]: db_allocate_columns(): allocate 52 bytes for result types at 0x7ffb95b49360 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[0] at 0x7ffb95b511c0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b511c0)[0]=[gw_name] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[1] at 0x7ffb95b51310 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51310)[1]=[ip_addr] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[2] at 0x7ffb95b512e0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b512e0)[2]=[port] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[3] at 0x7ffb95b512b0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b512b0)[3]=[uri_scheme] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[4] at 0x7ffb95b51280 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51280)[4]=[transport] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[5] at 0x7ffb95b51370 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51370)[5]=[params] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[6] at 0x7ffb95b513a0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b513a0)[6]=[hostname] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[7] at 0x7ffb95b493c0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b493c0)[7]=[strip] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[8] at 0x7ffb95b493f0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b493f0)[8]=[prefix] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[9] at 0x7ffb95b49420 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b49420)[9]=[tag] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[10] at 0x7ffb95b49450 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b49450)[10]=[flags] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[11] at 0x7ffb95b49480 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b49480)[11]=[defunct] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[12] at 0x7ffb95b51240 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51240)[12]=[id] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: <core> [db_res.c:184]: db_allocate_rows(): allocate 16 bytes for rows at 0x7ffb95b4e3b0 12(3267) DEBUG: <core> [db_row.c:119]: db_allocate_row(): allocate 416 bytes for row values at 0x7ffb95b494b0 12(3267) DEBUG: <core> [db_val.c:117]: db_str2val(): converting STRING [teste1] 12(3267) DEBUG: <core> [db_val.c:117]: db_str2val(): converting STRING [192.168.1.1] 12(3267) DEBUG: <core> [db_val.c:73]: db_str2val(): converting INT [5060] 12(3267) DEBUG: <core> [db_val.c:73]: db_str2val(): converting INT [1] 12(3267) DEBUG: <core> [db_val.c:73]: db_str2val(): converting INT [1] 12(3267) DEBUG: <core> [db_val.c:56]: db_str2val(): converting NULL value12(3267) DEBUG: <core> [db_val.c:117]: db_str2val(): converting STRING [192.168.1.1] 12(3267) DEBUG: <core> [db_val.c:56]: db_str2val(): converting NULL value12(3267) DEBUG: <core> [db_val.c:56]: db_str2val(): converting NULL value12(3267) DEBUG: <core> [db_val.c:56]: db_str2val(): converting NULL value12(3267) DEBUG: <core> [db_val.c:73]: db_str2val(): converting INT [0] 12(3267) DEBUG: <core> [db_val.c:56]: db_str2val(): converting NULL value12(3267) DEBUG: <core> [db_val.c:73]: db_str2val(): converting INT [1] 12(3267) DEBUG: lcr [lcr_mod.c:836]: insert_gw(): inserted gw <1, teste1, 192.168.1.1, 5060, 192.168.1.1> at index 1 12(3267) DEBUG: <core> [db_res.c:81]: db_free_columns(): freeing 13 columns 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[0] at 0x7ffb95b511c0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[1] at 0x7ffb95b51310 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[2] at 0x7ffb95b512e0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[3] at 0x7ffb95b512b0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[4] at 0x7ffb95b51280 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[5] at 0x7ffb95b51370 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[6] at 0x7ffb95b513a0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[7] at 0x7ffb95b493c0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[8] at 0x7ffb95b493f0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[9] at 0x7ffb95b49420 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[10] at 0x7ffb95b49450 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[11] at 0x7ffb95b49480 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[12] at 0x7ffb95b51240 12(3267) DEBUG: <core> [db_res.c:94]: db_free_columns(): freeing result names at 0x7ffb95b49280 12(3267) DEBUG: <core> [db_res.c:99]: db_free_columns(): freeing result types at 0x7ffb95b49360 12(3267) DEBUG: <core> [db_res.c:54]: db_free_rows(): freeing 1 rows 12(3267) DEBUG: <core> [db_row.c:97]: db_free_row(): freeing row values at 0x7ffb95b494b0 12(3267) DEBUG: <core> [db_res.c:62]: db_free_rows(): freeing rows at 0x7ffb95b4e3b0 12(3267) DEBUG: <core> [db_res.c:136]: db_free_result(): freeing result set at 0x7ffb95b51160 12(3267) DEBUG: <core> [db_res.c:118]: db_new_result(): allocate 56 bytes for result set at 0x7ffb95b51160 12(3267) DEBUG: db_mysql [km_res.c:68]: db_mysql_get_columns(): 13 columns returned from the query 12(3267) DEBUG: <core> [db_res.c:155]: db_allocate_columns(): allocate 104 bytes for result names at 0x7ffb95b49280 12(3267) DEBUG: <core> [db_res.c:165]: db_allocate_columns(): allocate 52 bytes for result types at 0x7ffb95b49360 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[0] at 0x7ffb95b49450 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b49450)[0]=[gw_name] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[1] at 0x7ffb95b49420 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b49420)[1]=[ip_addr] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[2] at 0x7ffb95b493f0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b493f0)[2]=[port] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[3] at 0x7ffb95b493c0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b493c0)[3]=[uri_scheme] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[4] at 0x7ffb95b513a0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b513a0)[4]=[transport] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[5] at 0x7ffb95b51370 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51370)[5]=[params] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[6] at 0x7ffb95b51280 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51280)[6]=[hostname] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[7] at 0x7ffb95b512b0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b512b0)[7]=[strip] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[8] at 0x7ffb95b512e0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b512e0)[8]=[prefix] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[9] at 0x7ffb95b51310 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51310)[9]=[tag] 12(3267) DEBUG: db_mysql [km_res.c:136]: db_mysql_get_columns(): use DB1_STRING result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[10] at 0x7ffb95b511c0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b511c0)[10]=[flags] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[11] at 0x7ffb95b51340 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51340)[11]=[defunct] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[12] at 0x7ffb95b51240 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51240)[12]=[id] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:169]: db_mysql_convert_rows(): no rows returned from the query 12(3267) DEBUG: <core> [db_res.c:81]: db_free_columns(): freeing 13 columns 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[0] at 0x7ffb95b49450 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[1] at 0x7ffb95b49420 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[2] at 0x7ffb95b493f0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[3] at 0x7ffb95b493c0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[4] at 0x7ffb95b513a0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[5] at 0x7ffb95b51370 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[6] at 0x7ffb95b51280 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[7] at 0x7ffb95b512b0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[8] at 0x7ffb95b512e0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[9] at 0x7ffb95b51310 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[10] at 0x7ffb95b511c0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[11] at 0x7ffb95b51340 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[12] at 0x7ffb95b51240 12(3267) DEBUG: <core> [db_res.c:94]: db_free_columns(): freeing result names at 0x7ffb95b49280 12(3267) DEBUG: <core> [db_res.c:99]: db_free_columns(): freeing result types at 0x7ffb95b49360 12(3267) DEBUG: <core> [db_res.c:136]: db_free_result(): freeing result set at 0x7ffb95b51160 12(3267) DEBUG: <core> [db_res.c:118]: db_new_result(): allocate 56 bytes for result set at 0x7ffb95b51160 12(3267) DEBUG: db_mysql [km_res.c:68]: db_mysql_get_columns(): 4 columns returned from the query 12(3267) DEBUG: <core> [db_res.c:155]: db_allocate_columns(): allocate 32 bytes for result names at 0x7ffb95b51240 12(3267) DEBUG: <core> [db_res.c:165]: db_allocate_columns(): allocate 16 bytes for result types at 0x7ffb95b511c0 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[0] at 0x7ffb95b51310 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51310)[0]=[rule_id] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[1] at 0x7ffb95b512e0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b512e0)[1]=[gw_id] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[2] at 0x7ffb95b512b0 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b512b0)[2]=[priority] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_res.c:85]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[3] at 0x7ffb95b51280 12(3267) DEBUG: db_mysql [km_res.c:92]: db_mysql_get_columns(): RES_NAMES(0x7ffb95b51280)[3]=[weight] 12(3267) DEBUG: db_mysql [km_res.c:100]: db_mysql_get_columns(): use DB1_INT result type 12(3267) DEBUG: db_mysql [km_dbase.c:358]: db_mysql_fetch_result(): converting row 0 of 1 count 1 12(3267) DEBUG: <core> [db_row.c:119]: db_allocate_row(): allocate 128 bytes for row values at 0x7ffb95b49280 12(3267) DEBUG: <core> [db_val.c:73]: db_str2val(): converting INT [1] 12(3267) DEBUG: <core> [db_val.c:73]: db_str2val(): converting INT [1] 12(3267) DEBUG: <core> [db_val.c:73]: db_str2val(): converting INT [1] 12(3267) DEBUG: <core> [db_val.c:73]: db_str2val(): converting INT [10] 12(3267) DEBUG: lcr [lcr_mod.c:1450]: reload_tables(): loading, cycle 0 with <1> rows 12(3267) DEBUG: lcr [hash.c:160]: rule_hash_table_insert_target(): found rule with id <1> and addr <0x7ffb8dbd5480> 12(3267) DEBUG: <core> [db_res.c:54]: db_free_rows(): freeing 1 rows 12(3267) DEBUG: <core> [db_row.c:97]: db_free_row(): freeing row values at 0x7ffb95b49280 12(3267) DEBUG: <core> [db_res.c:62]: db_free_rows(): freeing rows at 0x7ffb95b51370 12(3267) DEBUG: <core> [db_res.c:81]: db_free_columns(): freeing 4 columns 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[0] at 0x7ffb95b51310 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[1] at 0x7ffb95b512e0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[2] at 0x7ffb95b512b0 12(3267) DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[3] at 0x7ffb95b51280 12(3267) DEBUG: <core> [db_res.c:94]: db_free_columns(): freeing result names at 0x7ffb95b51240 12(3267) DEBUG: <core> [db_res.c:99]: db_free_columns(): freeing result types at 0x7ffb95b511c0 12(3267) DEBUG: <core> [db_res.c:136]: db_free_result(): freeing result set at 0x7ffb95b51160 12(3267) DEBUG: <core> [db_pool.c:102]: pool_remove(): removing connection from the pool [root@seabra kamailio]# 12(3267) DEBUG: ctl [io_listener.c:506]: handle_stream_read(): handle_stream read: eof on /tmp/kamailio_ctl 12(3267) DEBUG: ctl [../../io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x7ffb90811c00, 9, -1, 0x10) fd_no=2 called Thank you Best Regards -- Cumprimentos José Seabra

2 2

Missing AVPs in onreply_route (default reply route)
by Seudin Kasumovic 26 May '14

26 May '14

Hi, In next example $avp(test)=<null> in onreply_route. ... loadmodule "tm" ... request_route { ... $avp(test)='test'; ... t_relay(); ... } onreply_route { log("L_DEBUG","onreply_route: $$avp(test)=$avp(test)\n"; } But, if add t_on_reply: ... loadmodule "tm" ... request_route { ... $avp(test)='test'; ... t_on_reply("MANAGE_REPLY"); ... t_relay(); ... } onreply_route { log("L_DEBUG","onreply_route: $$avp(test)=$avp(test)\n"; } onreply_route["MANAGE_REPLY"] { log("L_DEBUG","MANAGE_REPLY:$$avp(test)=$avp(test)\n"; } log output is: onreply_route: $avp(test)=<null> MANAGE_REPLY: $avp(test)=test Why there is no AVPs in onreply_route? version: kamailio 4.1.3 (i386/linux) flags: STATS: Off, EXTRA_DEBUG, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: compiled on 17:23:27 Apr 25 2014 with gcc 4.5.4 Thanks, Seudin Kasumovic

2 1

Will the dispatcher server and redirect server What is the difference?
by vast 24 May '14

24 May '14

Hi, I have some questions . Will the dispatcher server and redirect server What is the difference? There is, if I use the dispatcher module, you must use freeswitch or asterisk backend Why? If I want to also use the kamailio as backend or receive the dispatcher module can?

2 1

TLS and SIP
by Fabian Borot 23 May '14

23 May '14

Hi I am using Kamailio 4.0.1 in front of an asterisk servers farm to handle TLS with our clients and providers. The idea is to have kamailio "talking" SIP/UDP/5060 and TLS/TCP/5061 with the customers and providers and regular SIP/UDP/5060 with our internal asterisk servers. So far at least for the customers it looks like it can work. But I have a problem, when the call is established and the called person hangs up, the BYE from the called person to the calling person is ignored. Only when the calling person hangs up first the call is terminated properly. This is what I have been able to see: 1- Customer starts the TLS handshake/connection. 2- Kamailio authenticate it, then routes the call to the asterisk server using regular SIP/UDP/5060 but I see that it is inserting 2 Record Routes in the INVITE: Record-Route: <sip:192.168.1.58;r2=on;lr=on> Record-Route: <sip:192.168.1.58:5061;transport=tls;r2=on;lr=on> 3- The Contact on that INVITE to the asterisk also comes like this: Contact: <sip:94167032@172.31.196.21:53325;transport=tls> 4- The ACK sent to the asterisk once it accepts the call (200 OK) also has those 2 Record-Routes: Record-Route: <sip:192.168.1.58;r2=on;lr=on> Record-Route: <sip:192.168.1.58:5061;transport=tls;r2=on;lr=on> 5- The call is established, once the called person decides to hang up the BYE looks like this: BYE sip:94167032@172.31.196.21:53325;transport=tls SIP/2.0 Via: SIP/2.0/UDP 192.168.1.59:5060;branch=z9hG4bK40fa1c23;rport Route: <sip:192.168.1.58;r2=on;lr=on>,<sip:192.168.1.58:5061;transport=tls;r2=on;lr=on> Max-Forwards: 70 From: <sip:3030500@1.2.3.4>;tag=as37953869 To: "kamailio" <sip:kamailio@1.2.3.4>;tag=788cd7c892df40f3b1967112395e2ca4 Call-ID: f9fe65daf1074219be26cb0c224339f1 CSeq: 102 BYE User-Agent: Asterisk PBX 11.3.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 My kamailio TLS config is shown below: enable_tls=yes loadmodule "tls.so" # ----- tls params ----- modparam("tls", "config", "/usr/local/kamailio-4.1//etc/kamailio/tls.cfg") modparam("tls", "private_key", "./privkey.pem") modparam("tls", "certificate", "./kamailio1_cert.pem") modparam("tls", "ca_list", "./calist.pem") modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) The TLS client that I am using is called Blink.At this point I don't know whether kamailio is sending the BYE using TLS to the customer and waiting for the 200 OK from the customer or whether kamailio does not like something in the BYE and that is why is ignoring it. I see some encrypted packets from kamailio to the client but I don't know what is inside. Any help would be very appreciated. thank you fabian

7 14

Issues after upgrade
by Daniel W. Graham 23 May '14

23 May '14

I recently upgraded to 4.0.6 from an older 4.0.x version using GIT, since then I have had several functions stop working. Here is one example: I have a UA that is not in the location table, when calling I get the following output as a result of my xlog statements - (xlog statements added for troubleshooting) INFO: xlog: 506:5d4d339e5a0fcb2d1ce245cb1c07bb71: Location not found for user-x INFO: xlog: 524:5d4d339e5a0fcb2d1ce245cb1c07bb71: no match if (!lookup("location")) { xlogl("L_INFO", "$ci : Location not found for $rU\n"); $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: xlogl("L_INFO", "$ci : case -1\n"); send_reply("480", "Temporarily Unavailable"); exit; case -2: xlogl("L_INFO", "$ci : case -2\n"); send_reply("405", "Method Not Allowed"); exit; case -3: xlogl("L_INFO", "$ci : case -3\n"); sl_send_reply("500", "Server Internal Error"); exit; } } xlogl("L_INFO", "$ci : no match\n"); This worked previous to the upgrade, it should match case -1 but doesn't. Any direction is appreciated. Thanks, -dan

2 6

kamailio 4.1.3 remote registration
by betul.cerit＠netgsm.com.tr 23 May '14

23 May '14

Hi to all, We use uac module to use remote registration but when I add a new user to uacreg table the change doesnt take effect until I restart kamailio service which is not convenient for us.Is there another way to achieve this without restarting service? My configuration is below.Am I doing sth wrong? modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ------ uac params ------------------ modparam("uac", "reg_db_url",DBURL) modparam("uac", "reg_timer_interval", 30) modparam("uac", "reg_retry_interval", 60) modparam("uac", "reg_db_table", "uacreg") modparam("uac","restore_mode","auto") modparam("uac", "reg_contact_addr","my ip") modparam("uac","auth_realm_avp","$avp(i:10)") modparam("uac","auth_username_avp","$avp(i:11)") modparam("uac","auth_password_avp","$avp(i:12)") modparam("uac","credential","username:domain:password") modparam("uac","restore_to_avp","$avp(original_uri_to)") modparam("uac","restore_from_avp","$avp(original_uri_from)") failure_route[REMOTE_AUTH]{ xlog("L_INFO","REMOTE_AUTH:$fU:$si$rm:REGISTER"); if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_NOTICE", "Remote asked for authentication"); uac_auth(); } } if(is_method("REGISTER")){ xlog("L_INFO","AUTH:$fU:$si$rm:REGISTER:Ru:$rU"); if(uac_reg_request_to("$fU","0")){ xlog("L_NOTICE", "Found remote user [$rU] on [$rd] via [$du]"); t_on_failure("REMOTE_AUTH"); t_relay(); exit; }else{ xlog("L_INFO","AUTH:$fU:$si$rm:REGISTER:else"); if (!auth_check("$fd", "subscriber", "0")) { auth_challenge("$fd", "0"); exit; } } } Thanks in advance.

2 1

Kamailio replies
by Andrea Meroni 23 May '14

23 May '14

Hi all, maybe this is a trivial question but ... how Kamailio manages SIP replies? I mean, how does it know to what URI reply in onreply_route? Does it use the SIP Via header or something else, maybe in memory? Regards, Andrea

2 2

Modifying SDP in Kamailio
by MrIhaveAnOpinionOnEverything 23 May '14

23 May '14

Hi guys: I am a R&D engineer trying to learn kamailio. After following some tutorials and reading the thread in this mailing list I was able to setup a voip backend with this configuration XLITE/LINPHONE ---> KAMAILIO ----> FREESWITCH I am using Freeswitch as a media server. After configuring RTP Proxy and kamailio to use bridged mode. I was able to successfully setup a voip backend like the one above. I encountered a problem when the UAC I am using is a webclient like sipml5. I noticed that when SIP INVITES from KAMAILIO to FREESWITCH are being passed when a INVITE transaction is initiated from a sipml5 client FREESWITCH is trying to use the public ip of webrtc server of the sipml5 backend. Unfortunately, I am using private ip/LAN IP between kamailio and freeswitch. As a result calls are established but there is no audio that is happening. I am attaching a snapshot of the ngrep that on kamailio and freeswitch server for your reference. I would like to know if there is a setting in kamailio that would allow me to modify the IP in the "o" and "c" sdp parameter when forwarding an invite to Freeswitch. I did another test. XLITE ---> KAMAILIO ---> FREESWITCH ----> KAMAILIO ----> sipml5 And the call works. It has audio. I think it must have something to do with the SDP header that is being generated by sip5ml UAC that is conflicting with my setup. Any help or advice will be greatly appreciated. Thanks. -- "When I look at you I see two people, ther person you are and the person you are supposed to be. Someday these two people will meet. And when they do, they will achieve great things" - Gene Hackman, The Replacements Catch up on me and check my BLOG at http://mrihaveanopiniononeverything.blogspot.com

3 2

insufficient ports returned from mediaproxy
by Morten Isaksen 23 May '14

23 May '14

We have a periodic problem with the communication between mediaproxy (kamailio module) <-> media-dispatcher <-> media-relay The problems seems to be that media-dispatcher mixes the requests and answers, so kamailio get a wrong answer to its request. Kamailio 3.1.1 Mediaproxy 2.4.4 I know they are old versions, but I have checket the new releases, and it does not look like anything related to this has been changed. This log snippet illustrates the problem well. We have added a few extra debug information: PID 3433 sends a update request to dispatcher dispatcher sends request to relay 10.253.253.32 relay 10.253.253.32 answers dispatcher PID 5214 sends a remove request to dispatcher dispatcher sends request to relay 10.253.253.32 relay 10.253.253.32 answers dispatcher dispatcher answer PID 3433 with the answer for PID 5214 The answer for PID 3433 is newer received by kamailio. May 15 08:02:33 sip-3-1 /usr/local/sbin/kamailio[5211]: WARNING: <script>: new request M=INVITE RURI=sip:40804XXX@76752XXX.example.com F= sip:40804XXX@76752XXX.example.com T=sip:40804XXX@172.16.9.20 IP=172.16.9.20 ID=738ce06b2a507610599b767b6b164657(a)hpbxout.one-connect.dk May 15 08:02:33 sip-3-1 media-dispatcher[3433]: debug: Command update Headers ['type: request', 'call_id: 738ce06b2a507610599b767b6b164657(a)hpbxout.one-connect.dk', 'cseq: 102', 'from_uri: 40804XXX(a)76752XXX.example.com', 'to_uri: 40804XXX(a)172.16.9.20', 'from_tag: as35017b2b', 'user_agent: one-connect', 'signaling_ip: 172.16.9.20', 'media: audio:172.16.9.18:50934:sendrecv:ice=no'] May 15 08:02:33 sip-3-1 media-dispatcher[3433]: debug: Issuing "update" command to relay at 10.253.253.32 May 15 08:02:33 sip-3-1 media-dispatcher[3433]: debug: To relay update 7529;type: request;call_id: 738ce06b2a507610599b767b6b164657(a)hpbxout.one-connect.dk;cseq: 102;from_uri: 40804XXX(a)76752XXX.example.com;to_uri: 40804XXX(a)172.16.9.20;from_tag: as35017b2b;user_agent: one-connect;signaling_ip: 172.16.9.20;media: audio:172.16.9.18:50934:sendrecv:ice=no;; May 15 08:02:33 sip-3-1 media-dispatcher[3433]: debug: line received 7529 172.16.8.32 63730 May 15 08:02:33 sip-3-1 /usr/local/sbin/kamailio[5214]: WARNING: <script>: new request M=BYE RURI=sip:65981002@172.16.9.21:5060 F= sip:65982380@172.16.8.37 T=sip:65981002@172.16.8.8 IP=172.16.8.8 ID= 24b6995108fc59946c0222084a79e1a7@172.16.8.37:5060 May 15 08:02:33 sip-3-1 media-dispatcher[3433]: debug: Command remove Headers ['call_id: 24b6995108fc59946c0222084a79e1a7@172.16.8.37:5060', 'from_tag: as792f4baa', 'to_tag: as798301b8'] May 15 08:02:33 sip-3-1 media-dispatcher[3433]: debug: Issuing "remove" command to relay at 10.253.253.32 May 15 08:02:33 sip-3-1 media-dispatcher[3433]: debug: To relay remove 7530;call_id: 24b6995108fc59946c0222084a79e1a7@172.16.8.37:5060;from_tag: as792f4baa;to_tag: as798301b8;; May 15 08:02:33 sip-3-1 media-dispatcher[3433]: debug: line received 7530 {"from_tag": "as792f4baa", "start_time": 1400133626.74, "call_id": " 24b6995108fc59946c0222084a79e1a7@172.16.8.37:5060", "duration": 126, "streams": [{"status": "closed", "caller_codec": "G711a", "post_dial_delay": 0.175052881241, "callee_codec": "G711a", "caller_bytes": 1269200, "start_time": 0, "callee_packets": 6088, "callee_bytes": 1217600, "caller_packets": 6346, "callee_remote": "172.16.9.18:58756", "end_time": 126, "caller_remote": "172.16.8.37:13014", "media_type": "audio", "callee_local": "172.16.8.32:61298", "timeout_wait": 0, "caller_local": " 172.16.8.32:61296"}], "to_tag": "as798301b8", "to_uri": "65981002(a)172.16.8.8", "caller_ua": "Asterisk PBX 1.8.12.0", "callee_ua": "one-connect", "from_uri": "65982380(a)172.16.8.37"} May 15 08:02:33 sip-3-1 media-dispatcher[3433]: debug: Got statistics: {'from_tag': 'as792f4baa', 'dialog_id': None, 'start_time': 1400133626.74, 'timed_out': False, 'call_id': ' 24b6995108fc59946c0222084a79e1a7@172.16.8.37:5060', 'to_tag': 'as798301b8', 'streams': [{'status': 'closed', 'caller_codec': 'G711a', 'post_dial_delay': 0.17505288124099999, 'callee_codec': 'G711a', 'start_time': 0, 'caller_bytes': 1269200, 'callee_bytes': 1217600, 'caller_packets': 6346, 'end_time': 126, 'callee_remote': '172.16.9.18:58756', 'caller_remote': '172.16.8.37:13014', 'media_type': 'audio', 'callee_local': '172.16.8.32:61298', 'timeout_wait': 0, 'caller_local': ' 172.16.8.32:61296', 'callee_packets': 6088}], 'duration': 126, 'to_uri': ' 65981002(a)172.16.8.8', 'from_uri': '65982380(a)172.16.8.37', 'callee_ua': 'one-connect', 'caller_ua': 'Asterisk PBX 1.8.12.0'} May 15 08:02:33 sip-3-1 /usr/local/sbin/kamailio[5211]: ERROR: mediaproxy [mediaproxy.c:1689]: mediaproxy response removed^M May 15 08:02:33 sip-3-1 /usr/local/sbin/kamailio[5211]: ERROR: mediaproxy [mediaproxy.c:1702]: Len of tokens: 1 May 15 08:02:33 sip-3-1 /usr/local/sbin/kamailio[5211]: ERROR: mediaproxy [mediaproxy.c:1713]: insufficient ports returned from mediaproxy: got 0, expected 1 callid: 738ce06b2a507610599b767b6b164657(a)hpbxout.one-connect.dk May 15 08:02:33 sip-3-1 /usr/local/sbin/kamailio[5211]: ERROR: <script>: RTPPROXY: use_media_proxy failed bm: 4326 Code=1 M=INVITE RURI=sip:40804XXX@sip-core.uni-tel.local F=sip:40804XXX@76752600.example.comT= sip:40804XXX@172.16.9.20 IP=172.16.9.20 ID= 738ce06b2a507610599b767b6b164657(a)hpbxout.one-connect.dk May 15 08:02:33 sip-3-1 /usr/local/sbin/kamailio[5214]: NOTICE: acc [acc.c:275]: ACC: transaction answered: timestamp=1400133753;method=BYE;from_tag=as04fb28c3;to_tag=367167352;call_id= 51654c57361b47d0751b3d1c33d19a4d(a)172.16.8.34;code=200;reason=OK -- Morten Isaksen

1 1

Need a commercial SIP Service
by Karl Grimm 22 May '14

22 May '14

Dear iptel, Thankyou for your open source website and excellent information. I hope you are able to help me with the following. We are an Australian organisation. We are trying to find a good quality SIP provider, commercial grade, with high connectivity success rates. We have used Skype, and it works incredibly well. It gets through your firewall and succeeds with login and successful voice and video call connection just about every time. But every public or commercial SIP Service I have used has failed to perform as well. Both the open source and the commercial SIP servers all suffer from the same problems: a.. they don't login unless you open up port 5060 on your firewall, regardless of whether STUN or TURN or ICE is in use b.. they don't reliably connect each call, sometimes the calling end just keeps ringing, and the receiving end doesn't ring c.. even if a call gets connected, they often have one of the 4 A/V streams missing, eg. we get video and outgoing audio, but no incoming audio d.. when one party hangs up, sometimes the other end sometimes doesn't know about it and just sits there thinking the call is still underway e.. when the Internet is busy, eg. at 4pm when children get home from school and jump on the Internet, their media relays struggle to keep up and UDP streams are badly degraded Note that we have not yet tried iptel.org, but I note that video is not talked about much on your website, and also you are an open source organisation, who cannot be expected to provide a commercial grade service. (If I am wrong about this, please tell me and I will certainly give iptel.org a try!) Using Jitsi, Linphone, and many other softphones, we have tried .... a.. ippi.fr b.. jumblo.com c.. pennytel.com d.. opensips.org e.. sip.linphone.org f.. an Australian commercial VoIP service by MyNetFone (works very well, but voice only) We are a business, and we are prepared to pay for a SIP Service. We cannot use ooVoo, Tango, Goober, ippi Messenger and other closed applications. We have to use an open source softphone such as Jitsi or Linphone, because they have an API. So my questions are .... a.. Are you aware of any commercial grade SIP providers that work as well as Skype: a.. they have STUN / TURN / ICE support in place b.. they support both video and audio calls c.. login occurs reliably: a.. regardless of where I am located (at home, at work, on my smartphone, using WiFi at a hotel, etc.) b.. regardless of what firewall I am behind d.. call connection occurs reliably, regardless of where I am located and regardless of firewall a.. every call always connects b.. if codecs at each end match, the call successfully connects audio and video streams both ways c.. when one party hangs up, the other end knows about it and doesn't just sit there thinking the call is still underway e.. their servers and infrastructure performs well even at peak periods, for example at 4pm when all the children get home from school and jump on the Internet f.. they have high performance media relays in place, so that audio and/or video can be relayed if required b.. If no company or organisation offers a SIP Service that performs as well as Skype, can you recommend any company or individual that we can engage (pay) to establish our own SIP Service. They must have the skills to setup SIP Server, STUN Server, TURN Server, and to adjust ports and settings so that anybody can login and make a call from anywhere, behind any firewall. Your recommendations would be very much appreciated. Thankyou, Karl Grimm. -------------------------------------------------------------------------------- Karl Grimm, Managing Director Mobile: 0418 310 989 | karl(a)konnekt.net.au Konnekt feel connected, feel safe

2 1

avpops issue
by Gilbert T. Gutierrez, Jr. 22 May '14

22 May '14

I am trying to setup call forwarding but I am getting a failure when I attempt to implement it. Kamailio Version 4.1.1 Centos 6 x64 Using precompiled RPMs from telephony.repo I am following guidance from the following urls... http://www.kamailio.org/wiki/tutorials/mini-howto-admin/call_forwarding http://www.kamailio.org/dokuwiki/doku.php/tutorials:avpops http://www.kamailio.org/dokuwiki/doku.php/examples:set-blind-call-forwarding 0(16018) ERROR: <core> [cfg.y:3272]: yyparse(): cfg. parser: failed to find command is_user_in 0(16018) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 1046, column 41: unknown command, missing loadmodule? 0(16018) ERROR: <core> [cfg.y:3272]: yyparse(): cfg. parser: failed to find command avp_write 0(16018) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 1048, column 39: unknown command, missing loadmodule? My understanding is that when I load module avpops that it should include those 2 procedures (avp_write and is_user_in). I am loading the module avpops.so Can someone please set me straight. Thank you. Gilbert T. Gutierrez, Jr. Phoenix Internet

2 2

My kamailio.cfg has ofen not registred , please help me check my registrar process is right or wrong
by vast 22 May '14

22 May '14

Hi, My kamailio has ofen not registred ,please help me check my registrar process is right or wrong (I think the MSILO modle maybe tedious or incorrect).this is my kamailio.cfg file.

2 1

Kamailio with Sqlite
by Ashwin Kumar R 22 May '14

22 May '14

Hi, I want to configure kamailio to use SQLITE instead of MYSQL.I made DB_ENGINE=SQLITE in kamctlrc file and loadmodule "db_sqlite.so" in kamctl.conf file. But when I changed these parameters modparam("auth_db", "db_url", "sqlite:///etc/kamailio/kamailio.db") #!define DBURL "sqlite:///etc/kamailio/kamailio.db" Im getting the following error ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed I checked the log file ,following is the comment in syslog May 20 19:16:30 r13pc80-desktop /usr/local/sbin/kamailio[10920]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'etc/kamailio/kamailio.db' What kamailio.cfg and kamctlrc parameters do I need to change to do this? Regards Ashwin

3 4

Accounting based on location/contact
by John Murray 22 May '14

22 May '14

Hello, I have subscribers that have 2 devices in the location table. These can make and receive call correctly and I can account for the calls. However I need to account which device has made or received the call as there are charges for one device type but not the other. I can modify the contact string from the device to add a parameter for example but I can't see how to add that into the accounting records using tm and acc modules. Thanks for your help. Regards John

3 3

i use the csipsimple registrar kamailio get request time out
by vast 22 May '14

22 May '14

Hi, I registrar kamailio ofen can registrar .but at other wifi some time i use the csipsimple registrar kamailio get request time out . when i look up the log I have found ERROR: registrar [save.c:692]: update_contacts(): invalid cseq for aor <1008> I search on google but can`t find a good answer. some one tell me it's not a same callid and low cseq . what would i do for this error ?

2 3

Asterisk Integration w Kamailio and Zoiper phone problems
by Brian Watson 22 May '14

22 May '14

Hi, I have a virtual environment setup and am trying to get the integration of Kamailio and Asterisk to a least be able to make a phone call based off the Asterisk Hello World example. I do the following: 1. I'm running Ubuntu 14.04 and Oracle Virtual Box 2. I Integrate Kamailio and Asterisk based off of the asipto kb: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb with both of them running on the same VM. 3. I've modified sip.conf and extensions.conf to have entries that tie together phones 101 and 102 3. I start asterisk with "asterisk" and "asterisk -rvvvvvv" 4. I install and setup the Zoiper Softphone as indicated in the tutorial. https://wiki.asterisk.org/wiki/display/AST/Hello+World 5. In Zoiper I hardcode the Domain: xxx.xxx.xxx.xxx:5080 and also enter the username and password. 6. I'm doing all of this without access to an internet connection so that's why I hardcode the IP address. 7. After setting of the Zoiper softphone I try to get it to communicate with Asterisk, but I get a 408 request timeout. Any ideas as to what might be wrong? Thanks, Brian

3 5

Kamailio and RTPProxy
by Thyrus Gorges 21 May '14

21 May '14

Hello all, I am currently trying to configure Kamailio and rtpproxy to play back a short message to the user-agent. I’ve started rtpproxy using: rtpproxy -F -s udp:127.0.0.1:7722 -l 10.1.10.102 -d WARN:LOG_LOCAL0 -u root I created the necessary sound file using makeann from the source of rtpproxy. Below are the relevant pieces of kamailio.cfg # ----- rtpproxy parameters ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") Kamailio.cfg ————— # Validate that the User Agent via IP Authorization via a SQL table if (!allow_address("1", "$si", "$sp")) { sl_send_reply("403", "Forbidden"); xlog("**** $si is UNAUTHORIZIED ****"); exit; } xlog("Enter route 1"); if (has_body("application/sdp")) rtpproxy_answer(); if (is_method("INVITE")) { xlog("New Call for route [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]"); if (rtpproxy_offer()) { t_on_reply("1"); } else { t_on_reply("2"); } #t_on_branch("2"); #t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[1] { if (has_body("application/sdp")) rtpproxy_manage(); rtpproxy_stream2uas("/var/rtpproxy/prompts/connect", "-1"); exit; } onreply_route[2] { if (has_body("application/sdp")) rtpproxy_offer(); exit; } When starting Kamailio, it informs me that it found the proxy and it enabled support for it. May 21 20:44:50 localhost /usr/local/sbin/kamailio[23816]: INFO: rtpproxy [rtpproxy.c:1607]: rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled Now, once I start up a call, rtpproxy reports the following information: May 21 20:14:35 localhost rtpproxy[23379]: INFO:handle_command: new session MTFiNWU1N2NiZGQwNWMwYzVkN2MzZjUyNzhkNTU3YWE., tag 88fc4d50;1 requested, type strong May 21 20:14:35 localhost rtpproxy[23379]: INFO:handle_command: new session on a port 41236 created, tag 88fc4d50;1 May 21 20:14:35 localhost rtpproxy[23379]: INFO:handle_command: pre-filling caller's address with 10.8.0.10:8000 May 21 20:14:35 localhost /usr/local/sbin/kamailio[23528]: ERROR: <script>: rtpproxy_offer returned true May 21 20:14:35 localhost /usr/local/sbin/kamailio[23522]: ERROR: rtpproxy [rtpproxy.c:1771]: select_rtpp_node(): script error -no valid set selected May 21 20:14:35 localhost /usr/local/sbin/kamailio[23522]: ERROR: rtpproxy [rtpproxy_stream.c:113]: rtpproxy_stream(): no available proxies Any assistance is greatly appreciated ! — Thyrus Gorges

1 0

Presence
by Joel White 21 May '14

21 May '14

I am working on enabling presence and bla in a Kamailio + FreeSWITCH environment All handsets are Polycom with a handful of Grandstream ATA's I enabled presence and presence_xml I have not got it to work as I keep seeing this meesage in the logs NOTICE: presence [subscribe.c:1030]: handle_subscribe(): Unsupported presence event call-info When I searched online for this error, the answers I got were saying this was a Linksys/Snom message.... How do I get presence to work properly? Thank you in advance

4 5

uri parameter value case sensitiveness
by Daniel-Constantin Mierla 21 May '14

21 May '14

Hello, hopefully someone here with the RFC fresh in mind and its grammar for sip uri parameter can jump on to clarify if the value of the parameter can be changed to lower/upper case. For a specific example, is it allowed that the value of a parameter in contact header uri is changed to upper/lower case when building the sequential request (so this appears in request uri)? Same would be for Recor-Route uri parameter that would appear later in Route headers... Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 8

Realtime integration: Unregistered clients showing as registered?
by Olli Heiskanen 20 May '14

20 May '14

Hello, One of the tests I've been working with is Asterisk realtime integration according to Daniel's guide here: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb Weird thing is the client looks registered but I'm not sure if it really is registered. If I'm not mistaken I should see the peers when I issue 'sip show peers' on asterisk cli. Instead I get this: *CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] Also, calling between clients will fail; in Asterisk cli I get: *CLI> == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [661@default:1] NoOp("SIP/660-00000000", "Testing: Dialed 661") in new stack -- Executing [661@default:2] Dial("SIP/660-00000000", "SIP/661,3600,rt") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/661 == Everyone is busy/congested at this time (1:0/0/1) -- Executing [661@default:3] Hangup("SIP/660-00000000", "") in new stack == Spawn extension (default, 661, 3) exited non-zero on 'SIP/660-00000000' In this setup I have SIP peers in an asterisk table added like this: INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser, fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com '); I have Kamailio and Asterisk on the same machine where Kamailio listens port 5060 and Asterisk listens 5070. Things that differ from the guide are Kamailio and Asterisk versions, which in my case are newer. Also, for another testing case I have MULTIDOMAIN enabled in Kamailio, does this interfere with the realtime integration? I'm using only one domain though. Please let me know if any configs or traces I can provide will help figure out what's going on. cheers, Olli

4 10

Upgrade to version 4.1
by VOIP Tests 20 May '14

20 May '14

Hello, I understand that there are some bugs in ver 4.03. Can someone let me know how I can update my running version of Kamailio ( 4.03 ) to version 4.1? Will I need to do a fresh install of the new version or is there a way I can do an upgrade/update to ver 4.03? Thank you, Arun

2 2

[rtpengine] No media from WebRTC UA
by Alexey Rybalko 20 May '14

20 May '14

Hello! During a call from classical SIP softphone to WebRTC there's no media from the browser (Mozilla, the same result is for Chrome). In case of a call from the browser to the softphone there's media flow from both sides. The snippets from kamailio.cfg related to the problem case (SIP-->WebRTC) are below. OFFER: $var(rtpp_flags) = "trust-address symmetric replace-origin replace-session-connection"; $var(rtpp_flags) = $var(rtpp_flags) + " ICE=force"; $var(rtpp_flags) = $var(rtpp_flags) + " RTP/SAVPF"; rtpengine_offer($var(rtpp_flags)); ANSWER: $var(rtpp_flags) = "trust-address symmetric replace-origin replace-session-connection"; $var(rtpp_flags) = $var(rtpp_flags) + " ICE=remove"; $var(rtpp_flags) = $var(rtpp_flags) + " RTP/AVP"; rtp.log is attached. Any help on this issue would be very appreciated. with best regards, Alexey

4 6

code review tools
by Daniel-Constantin Mierla 20 May '14

20 May '14

Hello, some time ago it was proposed to use a code review tool to analyze patches, the topic was also discussed during the last IRC devel meeting. I cc-ed users mailing list, because people there might have experiences with some of them. I found on wikipedia a short list of available applications for this purpose: - http://en.wikipedia.org/wiki/List_of_tools_for_code_review Personally I haven't used any so far. Iirc, gerrit was proposed before by Victor Seva. From the wikipedia list, Differential would be straightforward to deploy from dependencies point of view, as we have php on our servers. But I have no clue which is the right/best to use, so here we have to decide if worth to use one and which one. Besides the above one, perhaps github has an integrated review tool. I can be considered as an alternative as well. My opinion is that such tool is useful, specially for big patches, to allow easier more people to have a look at them. Hoping to get feedback on benefits of such tools and which would be the right/best to use. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

failed to find command redis_cmd
by Ryan Brindley 20 May '14

20 May '14

Hey Community, I'm trying to play around with ndb_redis, but seem to be missing something probably obvious as I'm getting a "failed to find command redis_cmd" error on load. loadmodule "ndb_redis.so" modparam("ndb_redis", "server", "name=test;addr=127.0.0.1;port=7000") request_route { redis_cmd("test", "SET foo bar", "r"); } I've verified that mpath = /usr/local/lib64/kamailio/modules/ and that ndb_redis.so is in that directory. Any help is greatly appreciated. Ryan Brindley

3 9

cfg values read from Database possible?
by Daniel Vukicevic 19 May '14

19 May '14

kamailio -V version: kamailio 3.2.2 (x86_64/linux) I am setting different cfg Parameters via XMLRPC. Which is good for changing parameters per runtime and being independent from database lookups when kamailio is running. Unfortunately cfg settings will be lost and empty after Restarting Kamailio if cfg values aren’t set before the request_route. So the idea was to look if cfg value is set, if not look into database and take the last cfg setting which was set. (Last cfg values will updated via SQL Query in the XMLRPC branch). # check if abc.test is defined if (strempty($sel(cfg_get.abc.test))) { sql_xquery("ca", "SELECT * from cfg where abc=‚test'", "ra“); The problem is that it seems, that there is no way to read out the values from database and to set the cfg variable again when kamailio is starting, because there is no way to inject cfg values into kamailio, when kamailio is already started. Here i need something like this: abc.test = "$xavp(ra=>LB01)" desc „TEST“; <<which doesn’t work in this form. (just to make it clear what i want) I found http://kamailio.org/docs/modules/stable/modules/cfg_db.html , but documentation is quite bad. Are there any other possibilities for me? The only way it might work for me (which is in my opinion quite dirty), to exec_dset an external command and to inject the new cfg via XMLRPC. The new cfg value is updated into database via XMLRPC again (which works, just for the sake of completeness): #!ifdef WITH_XMLRPC route[XMLRPC] { … … # Keeping values for Kamailio Restart sql_query("ca", "UPDATE cfg set LB01='$sel(cfg_get.abc.test)' where abc=‚test‘"); Mabye of one of you guys have a hint for me to solve this problem? Cheers Daniel

2 4

kamailio-OpenIms Presence
by Ashwin Kumar R 19 May '14

19 May '14

Hi, At the moment I'm working in a project, in which I need to use OpenIMSCore with a presence server, so I decided to use Kamailio.What should be added in the kamctlrc and kamailio.config files to make it meet the above requirement. Regards Ashwin

1 0

can you give me the complete kamailio.cfg that can use rtpproxy and uac_redirect?
by vast 19 May '14

19 May '14

Hello, lists. I was kamailio beginners, it seems demo on uac_redirect modules, but does not know how to use the kamailio.cfg file, can you give me an example of a complete hair. Thank you

1 0

Rtpproxy ptime in ringing inject
by Grygoriy Dobrovolskyy 19 May '14

19 May '14

Hello list, a long time ago i have build and application which injects ringing from rtpproxy to the client, while i am trying to reach his destination. The commande looks like this (log) May 19 10:17:44 rtpproxy[2529]: DBUG:handle_command: received command "29304_278705 P-1 B2B.256.470174 /usr/local/src/rtpproxy/sounds/ring session d20fbd749deed7e1f54460af276dee6d-0ead;1 " May 19 10:17:44 rtpproxy[2529]: INFO:handle_play: -1 times playing prompt /usr/local/src/rtpproxy/sounds/ring codec 8 May 19 10:17:44 rtpproxy[2529]: DBUG:doreply: sending reply "29304_278705 0#012" Everything works fine, a long time ago i have modded the rtpproxy a bit so when he sends the audio, he sends it with 20ms ptime instead of default 10ms, and i have totally forgot where is that part, i am digging into sources and not able to find it. Dear list if somebody remembers where is this damn setting in sources, i would be glad to hear from you. Thank you

1 0

mtree reload commands don't remove items from memory
by Juha Heinanen 18 May '14

18 May '14

i started to wonder why routing was sometimes wrong in my proxy and after studying the issue, i came to the conclusion that mtree reload commands (mi or rpc) don't remove entries from memory when they are removed from database. i started my proxy with empty mtrees db table (and empty memory table): # sip-proxy_ctl mi mt_summary MT:: TNAME=lp_out TTYPE=0 MEMSIZE=0 NRNODES=0 NRITEMS=0 RELOADCOUNT=0 RELOADTIME=1400419785 then i added one entry to db table and gave reload command. after that one entry correctly showed up in memory table: # sip-proxy_ctl mi mt_summary MT:: TNAME=lp_out TTYPE=0 MEMSIZE=1636 NRNODES=6 NRITEMS=1 RELOADCOUNT=1 RELOADTIME=1400419817 then i deleted the only entry from db, verified with phpmyadmin that the table really was empty, and gave again reload command. that command didn't have any effect in memory table: # sip-proxy_ctl mi mt_summary MT:: TNAME=lp_out TTYPE=0 MEMSIZE=1636 NRNODES=6 NRITEMS=1 RELOADCOUNT=1 RELOADTIME=1400419817 not even reload count got incremented. after that i went and added some debug to rpc_mtree_reload function and got this: May 18 16:30:17 siika /usr/sbin/sip-proxy[26536]: INFO: mtree [mtree_mod.c:1169]: rpc_mtree_reload(): reloading from db 'lp_out' the info is here: LM_INFO("reloading from db '%.*s'\n", pt->tname.len, pt->tname.s); if(mt_load_db(pt)!=0) { LM_ERR("cannot re-load mtree from database\n"); goto error; } my conclusion is that mt_load_db() function is somehow broken. -- juha

2 6

Siremis how to
by Joli Martinez 18 May '14

18 May '14

Hello, I am looking for some documentation or a how to on setting up some carrier routes using Siremis. thanks,

1 0

gateway connection
by Joli Martinez 17 May '14

17 May '14

Hello, I have setup Kamailio with Freeswitch and it is working. Right now I have my SIP providers connected to Freeswitch. Would it make sense to connect my SIP providers to kamailio instead. My setup will be intended to resell minutes. If so what advantages will I gain by connecting my SIP providers to kamailio

2 2

fetch bindings- timeout
by Daniel Ciprus 16 May '14

16 May '14

All, According to RFC3261: 10.2.3<http://tools.ietf.org/html/rfc3261#section-10.2.3> Fetching Bindings A success response to any REGISTER request contains the complete list of existing bindings, regardless of whether the request contained a Contact header field. If no Contact header field is present in a REGISTER request, the list of bindings is left unchanged. We should be able to fetch bindings from registrar. With my very limited knowledge of kamailio code (dialog-ng/dlg-handlers.c) : if ((!msg->cseq && (parse_headers(msg, HDR_CSEQ_F, 0) < 0 || !msg->cseq)) || !msg->cseq->parsed) { LM_ERR("bad sip message or missing CSeq hdr :-/\n"); goto error0; } This condition should also have something like: msg->method != "REGISTER" otherwise it's throwing errors and I'm not getting any response back from kamailio. Request just times out. Can somebody confirm that this feature is/is not implemented ? thank you Dan. Here is snippet of sipp xml scenario which I'm using to fetch bindings: <scenario name="FetchBindings"> <label id="1"/> <send> <![CDATA[ REGISTER sip:[field2] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch] From: <sip:[field0]@[field2]>;tag=aabbcc To: <sip:[field0]@[field2]> Call-ID: [call_id] CSeq: [cseq] REGISTER Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER User-Agent: IM-client/OMA1.0 sipp-script P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000 Privacy: none P-Visited-Network-ID: [field2] Supported: path Content-Length: 0 ]]> </send> <recv response="200" timeout="10000"/> </scenario> -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 1

Just 80 call/second
by Andrea Meroni 16 May '14

16 May '14

Hi all, i have some strange performance issues (with Kamailio config file i guess): using the default configuration file i can reach a 500 call/s rate, while using the default configuration file in the dispatcher folder i can reach just a 80 call/s rate (using the dispatching failover algorithm ...). The dispatcher is in front on a very basic Kamailio-based IMS core. Something has some hint about this issues? Andrea

2 4

Dispatcher poor performance
by Andrea Meroni 16 May '14

16 May '14

Hi all, i have a strange performance problem: i placed a dispatcher in front of a bundle of Kamailio proxies and, even with a really simple dispatching logic route { #ds_select_dstl("1", "2"); $du="sip:192.168.100.51:5060"; forward(); } that actually doesn't even use the dispatcher module i get that when the server's (192.168.100.51) CPU is 100% used the dispatcher's CPU is ~50% used. Using the dispatching algorithm based on the hash of the From-URI or the failover one (that should be the lightest dispatching algorithm, computationally speaking ...) the performance are almost the same. So, since i saw that in the documentation that the dispatcher module is designed to be "suitable for handling heavy SIP traffic", my question is: what i misconfigured? In my opinion it seems weird that the dispatcher module is able to manage just (as my tests suggest) just 2 Kamailio SIP proxies ... Andrea

1 0

Re: [SR-Users] sr-users Digest, Vol 108, Issue 15
by varun pratapsingh 16 May '14

16 May '14

HI Alex, Thanks for the suggestion. I tried using the function sl_send_reply(msg, 403, "Forbidden") in the receive.c file after including the header files. But the following error is coming when compiling the kamailio. CC (gcc) [kamailio] receive.o receive.c: In function âsend_denialâ: receive.c:375: warning: implicit declaration of function âsl_send_replyâ LD (gcc) [kamailio] kamailio receive.o: In function `send_denial': /home/varun/EP1.0/kamailio-4.1.3_src/kamailio-4.1.3/receive.c:375: undefined reference to `sl_send_reply' collect2: ld returned 1 exit status make: *** [kamailio] Error 1 Please suggest any solution to this problem. So I have make some changes in the Makefile as it seems to be linking problem. Thanks in advance. Regards Varun Date: Wed, 14 May 2014 13:09:44 +0100 > From: varun pratapsingh <varunps2003(a)gmail.com> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] How to reply an INVITE with 403 > Message-ID: > <CAMSCRPnCHq+v3hbirAAGWTP2kNSFvmxM=8g= > uWHAk8evaUEr7A(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi All, > > I want to handle a special case in the Kamailio 4.1.3. When I receive > INVITE from some UAC I wanto reply it with 403 message. I am using the > Kamailio in Proxy mode when the INVITE comes I want to reply it with 403 > and do not want to send this INVITE to SIP Server. > > I expored the Kamailio source code and found that fucntion receive_msg() in > file receive.c receives all the SIP messages. I am able to parse the SIP > INVITE from such UACs but how to reply them with 403 SIP Code. I am unable > to use the t_reply("403","Forbidden") here in this function/file. Also in > this function finally the forward_reply(msg); is called which sends the > INVITE to the Sip Server. But I dont want to send anything to SIP Server in > this case but want to reply the UAC with 403. Can you please guide me how > to do and where to do this. > > I am new to kamailio. Please help. > > Regards > Varun >

2 1

Issue in using the sl_send_reply in receive.c
by varun pratapsingh 16 May '14

16 May '14

Hi All, I am using the sl_send_reply function in the receive.c file. In included the header file sl_funcs.h correctly in the receive.c and receive. compiles fine. But when in gcc LD phase it gives the error CC (gcc) [kamailio] cfg.tab.o LD (gcc) [kamailio] kamailio receive.o: In function `send_denial': /home/varun/EP1.0/kamailio-4.1.3_src/kamailio-4.1.3/receive.c:365: undefined reference to `sl_send_reply' collect2: ld returned 1 exit status make: *** [kamailio] Error 1 Please can anybody let me know why this error is coming. If more info is needed I can provide that. Thanks

1 0

Duplicate values in SDP
by VOIP Tests 15 May '14

15 May '14

Hello, Can some one let me know the reason that there are these duplicate values in the SDP ( o, c, m, nortpproxy)? Our server was working with kamailio and asterisk on the same machine and had no problem. When we separated kamailio and asterisk on different servers and added the dispatcher module I see this error because of which there is not audio from an incoming PSTN call. 2014/05/15 18:55:19.972332 172.10.30.8:5060 -> 66.136.17.30:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 66.136.17.30:5060 ;rport=5060;branch=z9hG4bK1sansay1225318698rdb6244. Record-Route: <sip:54.108.18.75;lr=on;ftag=sansay1225318698rdb6244>. Record-Route: <sip:sansay1225318698rdb6244@66.136.17.30:5060 ;lr;transport=udp>. From: <sip:xxxxxx85342@66.136.17.30>;tag=sansay1225318698rdb6244. To: <sip:12142349395@54.108.18.75>;tag=as3414811d. Call-ID: 537507353-0-1838374200(a)64.136.174.226. CSeq: 1 INVITE. Server: Asterisk PBX 1.8.17.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Session-Expires: 1800;refresher=uac. Contact: <sip:xxxxxxx9395@172.10.30.5:5080>. Content-Type: application/sdp. Content-Length: 385. . v=0. o=root 1739301191 1739301191 IN IP4 54.108.18.7554.108.18.7554.108.18.75. s=Asterisk PBX 1.8.17.0. c=IN IP4 54.108.18.7554.108.18.7554.108.18.75. t=0 0. m=audio 388503885038850 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=nortpproxy:yes. a=nortpproxy:yes. a=nortpproxy:yes. Thank you for the help. Arun

1 0

DB select htable:mod-init doesn't work
by Sebastian Damm 15 May '14

15 May '14

Hi, I'm trying to fill a hash table on startup in the htable:mod-init event route. But it doesn't work. Kamailio doesn't start at all, but doesn't print out an error in the log. This is what the route looks like: event_route[htable:mod-init] { xlog("L_NOTICE", "Loading Carrier domains into Hash table.\n"); sql_query("ser", "select domain_name from carrier_domains", "domains"); xlog("L_NOTICE", "Loaded from database...\n"); while($var(i)<$dbr(domains=>cols)) { xlog("L_NOTICE", "Entry: $dbr(domains=>[$var(i),0])\n"); $sht(carrierips=>$var(i)) = $dbr(domains=>[$var(i),0]); $var(i) = $var(i) + 1; } } Of course, the necessary parameters are set. modparam("htable", "htable", "carrierips=>size=6") modparam("sqlops","sqlcon","ser=>mysql://user:password@127.0.0.1/ser") In the log I see the following: May 15 11:25:33 linux /usr/sbin/kamailio[20872]: NOTICE: <script>: Loading Carrier domains into Hash table. As you can see, there should be a lot more output, but nothing happens anymore after that. In ngrep I see other queries going to the same database from other modules, but my query from the event route is never executed. Am I missing something? Best Regards, Sebastian

4 8

Play sound during call waiting with kamailio
by Igor Potjevlesch 15 May '14

15 May '14

Hello, Is it possible to use kamailio 4.1 to play sound for the caller during call waiting? I saw the module "mohqueue" but i m not sure that the called have notification when a call is in a queue. Regards,

1 0

q_malloc crash
by Juha Heinanen 15 May '14

15 May '14

i just noticed that my proxy had crashed on invite request from attacker: May 14 22:03:06 sars /usr/sbin/sip-proxy[10932]: INFO: INVITE <tel:004412127921\ 94> by untrusted <sip:210.125.64.233> from <210.125.64.233> May 14 22:03:06 sars /usr/sbin/sip-proxy[10932]: : <core> [mem/q_malloc.c:159]:\ qm_debug_frag(): BUG: qm_*: prev. fragm. tail overwritten(c0c0c000, abcdefed)[\ 0xb70f6a64:0xb70f6a7c]! May 14 22:03:08 sars /usr/sbin/sip-proxy[11014]: : <core> [pass_fd.c:293]: rece\ ive_fd(): ERROR: receive_fd: EOF on 24 May 14 22:03:08 sars /usr/sbin/sip-proxy[10913]: ALERT: <core> [main.c:775]: ha\ ndle_sigs(): child process 10932 exited by a signal 6 May 14 22:03:08 sars /usr/sbin/sip-proxy[10913]: ALERT: <core> [main.c:778]: ha\ ndle_sigs(): core was generated Program terminated with signal 6, Aborted. #0 0xb7782424 in __kernel_vsyscall () (gdb) where #0 0xb7782424 in __kernel_vsyscall () #1 0xb7616941 in raise () from /lib/i386-linux-gnu/i686/cmov/libc.so.6 #2 0xb7619d72 in abort () from /lib/i386-linux-gnu/i686/cmov/libc.so.6 #3 0x08179f86 in qm_debug_frag (qm=0xb6dea008, f=0xb70f6a64) at mem/q_malloc.c:161 #4 0x0817ac3a in qm_malloc (qm=0xb6dea008, size=48, file=0x81f3169 "<core>: action.c", func=0x81f42f0 "do_action", line=780) at mem/q_malloc.c:386 #5 0x0805e798 in do_action (h=0xbffc1ca0, a=0xbffc1d48, msg=0xb72c6928) at action.c:780 #6 0xb1c5ac1d in pv_set_ruri (msg=0xb72c6928, param=0xb6f75630, op=254, val=0xbffc1e0c) at pv_core.c:2019 #7 0xb1b5df59 in tel2sip (_msg=0xb72c6928, _uri=0xb6f75c70 "H\207\367\266\004", _hostpart=0xb6f75530 "\254y\367\266\004", _res=0xb6f75624 "\006") at checks.c:405 #8 0x0805fdf7 in do_action (h=0xbffc21e0, a=0xb6f77858, msg=0xb72c6928) at action.c:1117 #9 0x08067293 in run_actions (h=0xbffc21e0, a=0xb6f77858, msg=0xb72c6928) at action.c:1599 #10 0x080678e2 in run_actions_safe (h=0xbffc39ac, a=0xb6f77858, msg=0xb72c6928) at action.c:1664 #11 0x081015fe in rval_get_int (h=0xbffc39ac, msg=0xb72c6928, i=0xbffc2528, rv=0xb6f779fc, cache=0x0) at rvalue.c:924 #12 0x08103f83 in rval_expr_eval_int (h=0xbffc39ac, msg=0xb72c6928, res=0xbffc2528, rve=0xb6f779f8) at rvalue.c:1918 #13 0x0810416e in rval_expr_eval_int (h=0xbffc39ac, msg=0xb72c6928, res=0xbffc27c4, rve=0xb6f78360) at rvalue.c:1926 #14 0x0805fa26 in do_action (h=0xbffc39ac, a=0xb6f78820, msg=0xb72c6928) at action.c:1075 #15 0x08067293 in run_actions (h=0xbffc39ac, a=0xb6f78820, msg=0xb72c6928) at action.c:1599 #16 0x0805fca0 in do_action (h=0xbffc39ac, a=0xb6f788c4, msg=0xb72c6928) at action.c:1094 ---Type <return> to continue, or q <return> to quit--- #17 0x08067293 in run_actions (h=0xbffc39ac, a=0xb6f788c4, msg=0xb72c6928) at action.c:1599 #18 0x0805fc5f in do_action (h=0xbffc39ac, a=0xb6f78968, msg=0xb72c6928) at action.c:1090 #19 0x08067293 in run_actions (h=0xbffc39ac, a=0xb6f78968, msg=0xb72c6928) at action.c:1599 #20 0x0805e00d in do_action (h=0xbffc39ac, a=0xb6e7720c, msg=0xb72c6928) at action.c:715 #21 0x08067293 in run_actions (h=0xbffc39ac, a=0xb6e5161c, msg=0xb72c6928) at action.c:1599 #22 0x0805e00d in do_action (h=0xbffc39ac, a=0xb6e50238, msg=0xb72c6928) at action.c:715 #23 0x08067293 in run_actions (h=0xbffc39ac, a=0xb6e50238, msg=0xb72c6928) at action.c:1599 #24 0x0805fc5f in do_action (h=0xbffc39ac, a=0xb6e50bfc, msg=0xb72c6928) at action.c:1090 #25 0x08067293 in run_actions (h=0xbffc39ac, a=0xb6e4891c, msg=0xb72c6928) at action.c:1599 #26 0x0806797a in run_top_route (a=0xb6e4891c, msg=0xb72c6928, c=0x0) at action.c:1685 #27 0x080e2bcf in receive_msg ( buf=0x82f99e0 "INVITE tel:00441212792194 SIP/2.0\r\nVia: SIP/2.0/UDP 210.125.64.233;branch=z9hG4bK4KmbLm4c\r\nMax-Forwards: 69\r\nFrom: <sip:210.125.64.233>;tag=qua2A5c8s9VJZ\r\nTo: <tel:00441212792194>\r\nContact: <sip:210.1"..., len=1115, rcv_info=0xbffc3bb0) at receive.c:211 #28 0x081702cd in udp_rcv_loop () at udp_server.c:536 #29 0x080ad9a0 in main_loop () at main.c:1617 #30 0x080b098f in main (argc=17, argv=0xbffc3e64) at main.c:2533 perhaps due to a bug in tel2sip function. -- juha

2 6

Prevent missing BYE request
by Igor Potjevlesch 15 May '14

15 May '14

Hello, Here is the context: I use to terminate outgoing calls to a PSTN gateway. The gateway is able to detect asymmetric RTP on IP side and then ended the call after a configurable timer if asymmetric RTP is detected. I'm working on the integration of a VoIP Trunk with a wholesaler and I'm wondering how this behaviour can be transposed on full IP interconnection. The first step will be to use Kamailio as an SBC and start the RTPProxy for each session terminated on this wholesaler. Is there a similar mechanism on RTPProxy sides that can detect this particular call-flow and hang up the call? Thanks for your suggestions. Regards, Igor.

2 2

Wrong Via and RR header added in UDP2TCP scenario
by Sebastian Damm 15 May '14

15 May '14

Hi, we have a setup, where we have a TCP loadbalancer, which converts the requests from attached devices using TCP into UDP for the rest of our setup. This loadbalancer has a service IP, and additionally the host IP, it listens to different addresses. Example: Host IP is: 1.1.1.1 Service IP is: 2.2.2.2 Device is at: 5.5.5.5 Proxy: 9.9.9.9 Gateway: 8.8.8.8 Kamailio listen IPs: tcp:1.1.1.1:15060 (used just for XMLRPC querys) tcp:2.2.2.2:5060 udp:1.1.1.1:5062 udp:2.2.2.2:5060 The Host IP should only be used for some management functions, not for routing. We've had the problem at first, that when converting from TCP to UDP, Kamailio did not use the same IP but instead the first configured one. I got rid of that by using force_send_socket. Now I'm calling a registered device. Since the registered device is saved with a path, the request comes in via UDP with the following parameters: U 2014/05/13 16:15:31.225132 9.9.9.9:5060 -> 2.2.2.2:5060 INVITE sip:user@5.5.5.5:5060;uniq=F373C9746E87D6271688292199F83;transport=tcp SIP/2.0 Record-Route: <sip:9.9.9.9;lr> Record-Route: <sip:2.2.2.2;lr;ftag=as458ff869> Via: SIP/2.0/UDP 9.9.9.9;branch=z9hG4bK17ec.ea7c017baba4072b27c5b0b9daa23346.0 Route: <sip:2.2.2.2;lr;received=sip:5.5.5.5:4956%3Btransport%3Dtcp> Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK17ec.7db07192e57ae94739d5c3e1444d7ef3.0 Via: SIP/2.0/UDP 8.8.8.8:5060;branch=z9hG4bK5ba05442 Up until now everything looks good. But this is what the loadbalancer sends out: T 2014/05/13 16:15:31.226987 2.2.2.2:5060 -> 5.5.5.5:4956 [A] INVITE sip:user@5.5.5.5:5060;uniq=F373C9746E87D6271688292199F83;transport=tcp SIP/2.0 Record-Route: <sip:1.1.1.1:15060;transport=tcp;r2=on;lr;ftag=as458ff869> Record-Route: <sip:2.2.2.2;r2=on;lr;ftag=as458ff869> Record-Route: <sip:9.9.9.9;lr> Record-Route: <sip:2.2.2.2;lr;ftag=as458ff869> Via: SIP/2.0/TCP 1.1.1.1:15060 ;branch=z9hG4bK17ec.99e285f5a9e03594d3f7ae6048d1868d.0 Via: SIP/2.0/UDP 9.9.9.9;branch=z9hG4bK17ec.ea7c017baba4072b27c5b0b9daa23346.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK17ec.7db07192e57ae94739d5c3e1444d7ef3.0 Via: SIP/2.0/UDP 8.8.8.8:5060;branch=z9hG4bK5ba05442 As you can see, the the packet is sent out through the correct socket. But in RR and Via header it advertises the other TCP socket. This breaks BYEs later on. Is there some way, I can influence, which IP and port Kamailio writes into those headers? Or is this a bug, that Kamailio doesn't check the sending socket correctly? Best Regards, Sebastian

3 6

Usrloc module issue
by aawaise 15 May '14

15 May '14

I went through usrloc page on kamailio site. Link given below. http://www.kamailio.org/docs/modules/3.1.x/modules_s/usrloc.html It states that the module keeps a user location table and provides access to the table to other modules. The module exports no functions that could be used directly from scripts. How can one extract value of domain from Location List ? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Usrloc-module-issue-tp127715.html Sent from the Users mailing list archive at Nabble.com.

1 0

Kamailio does not start with #!define WITH_AUTH
by Arya Farzan 15 May '14

15 May '14

When the following lines are on top of my kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_NAT #!define WITH_TLS #!define WITH_AUTH Kamailio does not start and prints out root@voip:/etc/kamailio# /etc/init.d/kamailio start [....] Starting Kamailio SIP server: kamailio:loading modules under /usr/lib/x86_64-linux-gnu/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: 107.170.93.193:5060 tcp: 127.0.0.1:5060 tcp: 107.170.93.193:5060 tls: 127.0.0.1:5061 tls: 107.170.93.193:5061 Aliases: tls: voip:5061 tls: localhost:5061 tcp: voip:5060 tcp: localhost:5060 udp: voip:5060 udp: localhost:5060 [FAIL] already running ... failed! . ok But when I remove "#!define WITH_AUTH" it starts up. How can I find out what the problem is?

4 4

Domain Name of an online user from Location List
by aawaise 15 May '14

15 May '14

How can I pick domain name of an online user from Location List ? Which pseudo variable is to be used ?? I want to get the value of domain name and get it printed in logs. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Domain-Name-of-an-online-user-from-… Sent from the Users mailing list archive at Nabble.com.

1 0

How to remove attribute from SDP ?
by Vitaliy Aleksandrov 14 May '14

14 May '14

Hello list, Can anybody suggest the way to remove a line/attribute from SDP that matches some pattern (regexp), some kind of "grep -v" for SDP ? I'm trying to remove unnecessary ICE candidates from the SDP body and only thing I know is the IP address of the candidate I need to remove. I was trying to use avp_subst() + set_body() or subst_body() but didn't get valid SDP. kamailio-3.2.4.

4 4

reg_fetch_contacts for multiple devices registered
by Jonathan Hunter 14 May '14

14 May '14

Hi Guys, I was wondering if anyone could help? I am currently using reg_fetch_contacts to manipulate signaling based on the user agent device being used, therefore I use the result of; if(reg_fetch_contacts("location", "$ru", "callee")){ xlog("callee=>user_agent $ulc(callee=>user_agent)\n");} I then use $ulc(callee=>user_agent) to make routing decisions. This seems to only return the user-agent device being used for the first registration at that AOR, as I have two devices registered, a Cisco, and a jitsi client, and it only returns the user_agent of the jitsi client I registered first. Is it possible to return the user agents of all the devices registered against an AOR so I can manage decisions accordingly? Many thanks Jon

2 9

Kamailio IRC Devel Meeting 2014
by Daniel-Constantin Mierla 14 May '14

14 May '14

Hello, shall we have our next irc devel meeting to discuss near future plans? I am proposing Thursday, May 15, 2014, at 14:00UTC on #kamailio channel at irc.freenode.net. If anyone wants a different date, please propose alternatives and we will select the one suiting most of the participants. I created a wiki page for it, adding some basic content for the moment, feel free to add new topics and details there: - http://www.kamailio.org/wiki/devel/irc-meetings/2014a Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 5

Presence dialoginfo - initial empty NOTIFY
by Ewgeny 14 May '14

14 May '14

Hello list, we're using kamailio presence for BLF functionality, that means lamps on the phone are green when phone is IDLE, blinking RED when - ringing and RED when busy. The problem is - when the phone send SUBSCRIBE first time - presence answers with empty NOTIFY (with no XML info), and lamps on the phone (for some extension) are OFF. The laps will became green only after first call to that extension. How to make non empty first NOTIFY message ? My config: #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode", 2) modparam("presence", "server_address", SBC_IPADDR) modparam("presence", "max_expires", 60) modparam("presence", "db_update_period", 200) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) modparam("presence_xml", "integrated_xcap_server", 0) modparam("presence_dialoginfo", "force_single_dialog", 1) #dialog need as dependence modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 0) #!endif the kamailio 3.3.4 example of initial empty NOTIFY: NOTIFY sip:user_A@192.168.4.2:5064 SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK693a.4fc432a84fe66cca11790dc1db4d520b.0 Via: SIP/2.0/UDP 10.10.10.10;rport=5060;branch=z9hG4bK693a.e25d1976.0 To: sip:user_A@xx.xx.xx.xx;tag=1632547193 From: sip:user_A@xx.xx.xx.xx;tag=40b780c8a91f5771f2f8fb63063e8dc6-f960 CSeq: 57 NOTIFY Call-ID: 997810293-5064-30(a)BJC.BGI.E.C Route: <sip:10.10.11.11;r2=on;lr=on;ftag=1632547193>, <sip:xx.xx.xx.xx;r2=on;lr=on;ftag=1632547193> Content-Length: 0 User-Agent: kamailio (3.3.4 (x86_64/linux)) Max-Forwards: 69 Event: dialog Contact: <sip:10.10.10.10:5060> Subscription-State: active;expires=60 example of NOTIFY with XML info: NOTIFY sip:user_A@12.12.12.12:1049 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.10;branch=z9hG4bKb7e2.4fd3e262.0 To: sip:user_A@xx.xx.xx.xx;tag=2032642966 From: sip:user_B@xx.xx.xx.xx;tag=40b780c8a91f5771f2f8fb63063e8dc6-a5e9 CSeq: 195 NOTIFY Call-ID: 734484242-5064-35(a)BJC.BGI.E.C Route: <sip:10.10.11.11;r2=on;lr=on;ftag=2032642966>, <sip:xx.xx.xx.xx;r2=on;lr=on;ftag=2032642966> Content-Length: 327 User-Agent: kamailio (3.3.4 (x86_64/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:xx.xx.xx.xx:5060> Subscription-State: active;expires=60 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="00000000000" state="full" entity="user_B(a)10.10.10.10"> <dialog id="fd72b08c-5603-1232-f192-005056954c8a" call-id="fd72b08c-5603-1232-f192-005056954c8a" direction="recipient"> <state>terminated</state> </dialog> </dialog-info>

1 0

How to reply an INVITE with 403
by varun pratapsingh 14 May '14

14 May '14

Hi All, I want to handle a special case in the Kamailio 4.1.3. When I receive INVITE from some UAC I wanto reply it with 403 message. I am using the Kamailio in Proxy mode when the INVITE comes I want to reply it with 403 and do not want to send this INVITE to SIP Server. I expored the Kamailio source code and found that fucntion receive_msg() in file receive.c receives all the SIP messages. I am able to parse the SIP INVITE from such UACs but how to reply them with 403 SIP Code. I am unable to use the t_reply("403","Forbidden") here in this function/file. Also in this function finally the forward_reply(msg); is called which sends the INVITE to the Sip Server. But I dont want to send anything to SIP Server in this case but want to reply the UAC with 403. Can you please guide me how to do and where to do this. I am new to kamailio. Please help. Regards Varun

2 1

Create dialog for SIP PRESENCE
by Ewgeny 14 May '14

14 May '14

Hi, i'm trying to create SIP dialog for SUBSCRIBE messages, but it's still unsuccessful (. if (is_method("SUBSCRIBE")) { setflag(4); dlg_manage(); xlog("L_INFO", "DLG_count=$DLG_count, DLG_status=$DLG_status, DLG_lifetime=$DLG_lifetime - (Method=$rm RURI=$ru From=$fu To=$tu IP=$si DST=$du -- $ci)\n"); } In kamailio.log: May 10 18:15:11 /usr/sbin/kamailio[18355]: DEBUG: dialog [dlg_hash.c:652]: no dialog callid='265619152(a)192.168.4.18' found May 10 18:15:11 /usr/sbin/kamailio[18355]: DEBUG: dialog [dlg_hash.c:683]: no dialog callid='265619152(a)192.168.4.18' found May 10 18:15:11 /usr/sbin/kamailio[18355]: DEBUG: dialog [dlg_handlers.c:1170]: Callid '265619152(a)192.168.4.18' not found Does kamailio 3.3.4 support Dialog for PRESENCE (Subscribe, Publish, Notify) ? Best Regards, Ewgeny

3 6

Issue in Multi Domain Environment
by aawaise 14 May '14

14 May '14

I have deployed two kamailio servers with independent domain names, BOTH sharing the same database. When an "INVITE" packet approaches one server from a user. As per my understanding it checks "Location Table" (containing online users) to find about status (online or not) of called user. If user is present in LOCATION TABLE, kamailio server sends INVITE to callee else it disconnects the call. In LOCATION TABLE, DOMAIN NAME is also mentioned along with user name, I want to extract that domain name of callee user from LOCATION TABLE with every INVITE packet. What variable stores this information ?? How can it be extracted ?? I have tried using "$du". It consumes RURI and gives domain name, it picks domain name from invite packet, which is not required here. Requirement is to pick domain name from LOCATION TABLE, by which user is logged into the system. (In multidomain setup, they both can be different). I have also tried "$td", they also pick domain name from INVITE packet header. I want to print (i.e xlog) value of domain from the LOCATION TABLE. And it is bothering for quite a few days now. Any help will be much appreciated. Thanks. Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Issue-in-Multi-Domain-Environment-t… Sent from the Users mailing list archive at Nabble.com.

2 2

SHM error after few days...
by Krishna Kurapati 13 May '14

13 May '14

Hi, After few days of running kamailio (I am using TLS for client), suddenly I see that the connections are being dropped. When I restart kamailio, I see following errors: Stopping kamailio: [FAILED] Not starting : invalid configuration file! 0(20563) : <core> [mem/shm_mem.c:168]: shm_getmem(): ERROR: shm_mem_init: could not attach shared memory segment: Cannot allocate memory 0(20563) : <core> [mem/mem.c:125]: init_shm_mallocs(): could not initialize shared memory pool, exiting... Too much shared memory demanded: 4294967296 0(20563) ERROR: <core> [sr_module.c:627]: load_module(): ERROR: load_module: /usr/lib64/kamailio/modules/tls.so: mod_register failed 0(20563) : <core> [cfg.y:3408]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 328, column 12-19: failed to load module 0(20563) : <core> [mem/shm_mem.c:168]: shm_getmem(): ERROR: shm_mem_init: could not attach shared memory segment: Cannot allocate memory 0(20563) : <core> [mem/mem.c:125]: init_shm_mallocs(): could not initialize shared memory pool, exiting... Too much shared memory demanded: 4294967296 0(20563) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 357, column 71: Can't initialize shared memory ERROR: bad config file (2 errors) 0(20563) ERROR: mi_fifo [mi_fifo.c:271]: mi_destroy(): FIFO stat failed: Bad address My shared memory size for the kamailio is 4096M. It started fine initially. However, after I noticed connection drops, I see the above issue. THen I reduced shared memory to 2048M and it started fine. I am under the impression that shared memory is just a filed I/O mapped which can be several hundred gigs. I looked at my filesystem. It looks fine df Filesystem 1K-blocks Used Available Use% Mounted on /dev/sda3 958788080 200734800 709349640 23% / tmpfs 3975996 0 3975996 0% /dev/shm /dev/sda1 495844 37498 432746 8% /boot What could be wrong here? Any help greatly appreciated. Krish Kurapati

1 0

Memory issue
by Igor Potjevlesch 13 May '14

13 May '14

Hello, I had a memory issue this morning. First, I could see the following logs in /var/log/message : May 12 11:08:21 localhost /usr/local/sbin/kamailio[30268]: ERROR: <core> [rvalue.c:2370]: rval_expr_eval(): rv eval int expression: out of memory May 12 11:08:25 localhost /usr/local/sbin/kamailio[30268]: ERROR: <core> [msg_translator.c:2164]: build_res_buf_from_sip_req(): ERROR: build_res_buf_from_sip_req: out of memory ; needs 482 May 12 11:08:26 localhost /usr/local/sbin/kamailio[30268]: ERROR: <core> [msg_translator.c:2012]: build_res_buf_from_sip_res(): ERROR: build_res_buf_from_sip_res: out of mem May 12 11:08:27 localhost /usr/local/sbin/kamailio[30268]: ERROR: <core> [msg_translator.c:1910]: build_req_buf_from_sip_req(): ERROR: build_req_buf_from_sip_req: out of memory May 12 11:08:28 localhost /usr/local/sbin/kamailio[30268]: ERROR: <core> [rvalue.c:2370]: rval_expr_eval(): rv eval int expression: out of memory May 12 11:08:31 localhost /usr/local/sbin/kamailio[30268]: ERROR: <core> [msg_translator.c:1910]: build_req_buf_from_sip_req(): ERROR: build_req_buf_from_sip_req: out of memory May 12 11:08:32 localhost /usr/local/sbin/kamailio[30268]: ERROR: <core> [db_res.c:181]: db_allocate_rows(): no private memory left May 12 11:08:32 localhost /usr/local/sbin/kamailio[30268]: ERROR: db_mysql [km_res.c:175]: db_mysql_convert_rows(): could not allocate rows May 12 11:08:32 localhost /usr/local/sbin/kamailio[30268]: ERROR: db_mysql [km_res.c:217]: db_mysql_convert_result(): error while converting rows May 12 11:08:32 localhost /usr/local/sbin/kamailio[30268]: ERROR: db_mysql [km_dbase.c:191]: db_mysql_store_result(): error while converting result May 12 11:08:33 localhost kernel: kamailio[30268]: segfault at 30 ip 00007ff44dd93f93 sp 00007fffcb00a200 error 4 in db_mysql.so[7ff44dd8f000+19000] May 12 11:08:37 localhost /usr/local/sbin/kamailio[30235]: ALERT: <core> [main.c:788]: handle_sigs(): child process 30268 exited by a signal 11 May 12 11:08:37 localhost /usr/local/sbin/kamailio[30235]: ALERT: <core> [main.c:791]: handle_sigs(): core was generated May 12 11:08:37 localhost /usr/local/sbin/kamailio[30235]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLD This problem have already occurred and I have set -M 64 to avoid it. It looks that it's not enough anymore. What is the next increment recommended after 64M? On the contrary of the first occurrence, Kamailio crash with coredump just after these errors. I'm not sure that the memory message errors are linked to the crash. I have tried a "bt full": (gdb) bt full #0 db_mysql_store_result (_h=0x7ff44e0e70e0, _r=0x7fffcb00a408) at km_dbase.c:198 code = <value optimized out> __FUNCTION__ = "db_mysql_store_result" #1 0x00007ff44d971e72 in db_do_query_internal (_h=0x7ff44e0e70e0, _k=0x7fffcb00a3e0, _op=0x0, _v=0x7fffcb00a3a0, _c=<value optimized out>, _n=<value optimized out>, _nc=2, _o=0x0, _r=0x7fffcb00a408, val2str=0x7ff44dd97fa0 <db_mysql_val2str>, submit_query=0x7ff44dd92ed0 <db_mysql_submit_query>, store_result=0x7ff44dd93d80 <db_mysql_store_result>, _l=0) at db_query.c:137 tmp = <value optimized out> off = <value optimized out> ret = <value optimized out> __FUNCTION__ = "db_do_query_internal" #2 0x00007ff44d97269a in db_do_query (_h=<value optimized out>, _k=<value optimized out>, _op=<value optimized out>, _v=<value optimized out>, _c=<value optimized out>, _n=<value optimized out>, _nc=2, _o=0x0, _r=0x7fffcb00a408, val2str=0x7ff44dd97fa0 <db_mysql_val2str>, submit_query=0x7ff44dd92ed0 <db_mysql_submit_query>, store_result=0x7ff44dd93d80 <db_mysql_store_result>) at db_query.c:156 Thanks in advance for your help. Regards, Igor.

2 6

RTP ports
by Neville D'Souza 13 May '14

13 May '14

HI , How do I find out which udp ports my kamailio server is using. In Kamailio.cfg I am using socket like below. #!ifdef WITH_NAT # ----- rtpproxy params ----- #modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock") I need to know the exact port range to open ports on my firewall if necessary. Our Kamailio server works just fine outside our network, but when I try to make calls through linphone client using a android device inside our network, as soon as I pick up the call it gets disconnected. The same thing happens when someone from outside tries to call in to our network. Funny part is when I use a desktop client like jitsi the calls between 2 users in the network or with someone outside the network work fine. I am guessing the rtp traffic is being dropped by the firewall when using linphone android client. It was working fine previously but after a firewall OS update it stopped working. Best Regards, Neville D'Souza Network Administrator r_gogopay_96x73 Gogopay Online Pvt Ltd. India +91 832 222 0119 ncd(a)gogopay.com www.gogopay.com

2 1

Why Kamailio reply with null compression method when TLS handshake?
by 刘日新 13 May '14

13 May '14

Hi, all. I has set up a Kamailio server with TLS & compression enabled, I thought I set most things suitable , I has set modparam("tls","tls_disable_compression",0) I can find such log records as below when Kamailio boost: 0(10905) INFO: tls [tls_init.c:549]: init_tls_h(): tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.0-fips 29 Mar 2010" (0x10000003), kerberos support: on, compression: on 0(10905) INFO: tls [tls_init.c:557]: init_tls_h(): tls: init_tls_h: installed openssl library version "OpenSSL 1.0.0-fips 29 Mar 2010" (0x10000003), kerberos support: on, zlib compression: on compiler: gcc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT -m64 -DL_ENDIAN -DTERMIO -Wall -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -Wa,--noexecstack -DMD32_REG_T=int -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM -DWHIRLPOOL_ASM 0(10905) WARNING: tls [tls_init.c:611]: init_tls_h(): tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 11534336 and 5767168 bytes 0(10905) INFO: <core> [cfg/cfg_ctx.c:613]: cfg_set_now(): INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 11534336 0(10905) INFO: <core> [cfg/cfg_ctx.c:613]: cfg_set_now(): INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 5767168 And I run kamcmd tls.options, I got: { force_run: 0 method: TLSv1 verify_certificate: 0 verify_depth: 9 require_certificate: 0 private_key: /ca/cert.pem ca_list: certificate: /ca/cert.pem cipher_list: session_cache: 1 session_id: vic22 config: /etc/kamailio/tls.cfg log: 3 debug: 3 connection_timeout: 600 disable_compression: 0 ssl_release_buffers: -1 ssl_freelist_max: -1 ssl_max_send_fragment: -1 ssl_read_ahead: 0 send_close_notify: 0 low_mem_threshold1: 11534336 low_mem_threshold2: 5767168 ct_wq_max: 10485760 con_ct_wq_max: 65536 ct_wq_blk_size: 4096 } But when My UA connect to this server, when TLS handshake, I can find that the clienthello with two compression method :1 (deflate) and 0 (null), but the server side reply with just one compression method: 0( null), thus the compression was disabled through the following communication. Why? are there any others issue can impact the behavior of Kamailio? Any hints will be appreciated. B.R. Rixin liu

1 0

Children, PKG_MEM and IMS
by Andrea Meroni 13 May '14

13 May '14

Hi all, i'm a little bit confused about the #children/PKG_MEM mechanism: in my setup (a simple IMS core with one P-CSCF, one I-CSCF, one dispatcher and 2 S-CSCFs) i wish to manage the highest possible #calls, but i don't know which is the ideal setup. I mean, i'm running all this components in VMs (with mem = 2048MB and smp = 8) inside a desktop machine with RAM = 32 GB and 8 processors; is it worth to define #children = #processors and setting an high value of PKG_MEM (for example 128MB) or use an higher #children (for example 64) with less PKG_MEM (for example 16MB)? Someone could explain me better how the mechanism works? Thank you, Andrea

1 0

Version support
by Jan Gaida 12 May '14

12 May '14

Hello, Which versions of Kamailio (OpenSER) are officially supported (e.g. actively patched) ? Is there a list of EOL dates for all versions or what is the general rule? Kind regards Jan -- *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.gaida(a)grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario.

2 1

Problem with rtpproxy and rtpproxy_manage()
by aft 12 May '14

12 May '14

Hi, I'm using kamailio from latest git-HEAD. The rtpproxy i'm using also from latest git. Our network topology is following : sip-softphone--------->kamailio/rtpproxy-------->softswitch---->gateway Because of saving bandwidth we need to use the "re-packetization" feature of rtpproxy. When we don't use it, it works perfectly without any glitches. Now if we use the repacketization feature like following : if (is_request()) { rtpproxy_manage("co"); } if (is_reply()) { rtpproxy_manage("z80"); } Then a bizzare thing happens. Suddenly rtp packets from gateway stops coming. So we don't have any voice in softphone, although everything seems ok at the "mobile phone's end." This is bizzare at multiple levels, as you can see we are using repacketize feature to increase the payload size from "proxy to softphone" leg. So we actually don't touch the packets from "softphone to proxy" leg. So how the gateway predicts this and decides not to send packets is beyond my understanding. As far as i can understand, rtpproxy_manage() is doing something to SDP which makes it impossible to send packets through the proxy. I will provide further information if anybody wants to look at it. Thanks in advance. -- -Cheers -Arif

2 5

Kamailio stopped, but kamctl online still get user info
by Eirc 12 May '14

12 May '14

Hello, all of users: I am confused here, my kamalio is stopped, but kamctl online and ul show still can see the user info: ============================================ root@kamailio:/home/admin_1# kamctl online 102 root@kamailio:/home/admin_1# kamctl online 102 root@kamailio:/home/admin_1# kamctl online 102 root@kamailio:/home/admin_1# service kamailio status Status of Kamailio: kamailio is not running. root@kamailio:/home/admin_1# kamctl ul show Domain:: location table=512 records=1 max_slot=1 AOR:: 102 Contact:: sip:102@192.168.1.105:5060 Q= Expires:: 313 Callid:: 363490571-5060-1(a)BJC.BGI.B.BAF Cseq:: 2117 User-agent:: Grandstream GXP2124 1.0.4.10 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:192.168.1.116:5060 Methods:: 7135 Ruid:: uloc-536d9a68-6a5-1 Instance:: <urn:uuid:00000000-0000-1000-8000-000B824017A0> Reg-Id:: 1 root@kamailio:/home/admin_1# ========================================== Does kamctl only receive the data from mysql, no matter kamctl status? -- *Best regards*

2 1

Kamailio Logged in users
by Neville D'Souza 12 May '14

12 May '14

Hi, I am having a slight issue with Kamilio server. How do I disconnect users without restarting the server. When I run the command kamctl ul users i get the list of users connected, some that are already disconnected. I am using the Linphone android client to coonect through 5060 port. Kindly advise Best Regards, Neville D'Souza Network Administrator r_gogopay_96x73 Gogopay Online Pvt Ltd. India +91 832 222 0119 ncd(a)gogopay.com www.gogopay.com

2 1

One way audio
by aft 12 May '14

12 May '14

Hi, The network topology is : SipSoftphone(caller)----->kamailio/rtpproxy---->Softswitch---->sipGateway-->mobile phone(callee) The rtpproxy is managed in kamailio script by : if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if (is_request()) { rtpproxy_offer("co"); } if (is_reply()) { rtpproxy_answer("coz80"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } Now if in rtpproxy_answer(), i don't resize rtp packets, then everything works fine. If i put "z80" then only one way audio can be heard. In SipSoftphone(caller) side, nothing can be heard. I have sniffed the SIP packets and it looked fine to me. The rtp packets from callee arrives fine in kamailio/rtpproxy box. But it does not leave that box. I've sniffed the rtp packets. Before the call is picked up by the callee, there is an ivr played. The ivr arrives at the caller(softphone) end fine. But after picking up the phone, rtppackets are not delivered to the caller(softphone)'s side. This is the rtpproxy's log: May 11 23:54:29 65429 rtpproxy[24207]: INFO:rxmit_packets: callee's address latched in: 85.13.205.178:13716 (RTP) May 11 23:54:41 65429 rtpproxy[24207]: DBUG:get_command: received command "27802_12 Uc18,96 2b15632de8c94ff488e9cdc1cdbcb85d 182.16.156.25 4000 74d009e3a57246138e2e0124cbe6ee89;1" May 11 23:54:41 65429 rtpproxy[24207]: INFO:handle_command: adding strong flag to existing session, new=1/0/0 May 11 23:54:41 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 54920/64810, session timer restarted May 11 23:54:41 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27802_12 54920 64.120.57.111#012" May 11 23:55:42 65429 rtpproxy[24207]: INFO:process_rtp: session timeout May 11 23:55:42 65429 rtpproxy[24207]: INFO:remove_session: RTP stats: 1523 in from callee, 300 in from caller, 396 relayed, 0 dropped May 11 23:55:42 65429 rtpproxy[24207]: INFO:remove_session: RTCP stats: 0 in from callee, 8 in from caller, 8 relayed, 0 dropped May 11 23:55:42 65429 rtpproxy[24207]: INFO:remove_session: session on ports 54920/64810 is cleaned up May 11 23:58:58 65429 rtpproxy[24207]: DBUG:get_command: received command "27802_13 Uc18,96 008e2697e52a4adda7be16358b318def 182.16.156.25 4002 907ba9691f6a47f3bfb0bb4143ef7dc7;1" May 11 23:58:58 65429 rtpproxy[24207]: INFO:handle_command: new session 008e2697e52a4adda7be16358b318def, tag 907ba9691f6a47f3bfb0bb4143ef7dc7;1 requested, type strong May 11 23:58:58 65429 rtpproxy[24207]: INFO:handle_command: new session on a port 53876 created, tag 907ba9691f6a47f3bfb0bb4143ef7dc7;1 May 11 23:58:58 65429 rtpproxy[24207]: INFO:handle_command: pre-filling caller's address with 182.16.156.25:4002 May 11 23:58:58 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27802_13 53876 64.120.57.111#012" May 11 23:58:59 65429 rtpproxy[24207]: DBUG:get_command: received command "27803_10 LZ80c18 008e2697e52a4adda7be16358b318def 85.13.205.178 26728 907ba9691f6a47f3bfb0bb4143ef7dc7;1 6611E63499885;1" May 11 23:58:59 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 53876/42866, session timer restarted May 11 23:58:59 65429 rtpproxy[24207]: INFO:handle_command: pre-filling callee's address with 85.13.205.178:26728 May 11 23:58:59 65429 rtpproxy[24207]: INFO:handle_command: RTP packets from callee will be resized to 80 milliseconds May 11 23:58:59 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27803_10 42866 64.120.57.111#012" May 11 23:58:59 65429 rtpproxy[24207]: INFO:rxmit_packets: callee's address filled in: 85.13.217.158:26728 (RTP) May 11 23:58:59 65429 rtpproxy[24207]: INFO:rxmit_packets: guessing RTCP port for callee to be 26729 May 11 23:58:59 65429 rtpproxy[24207]: INFO:rxmit_packets: caller's address latched in: 182.16.156.25:4003 (RTCP) May 11 23:58:59 65429 rtpproxy[24207]: INFO:rxmit_packets: caller's address filled in: 182.16.156.25:1036 (RTP) May 11 23:58:59 65429 rtpproxy[24207]: DBUG:get_command: received command "27801_10 LZ80c18 008e2697e52a4adda7be16358b318def 85.13.205.178 26728 907ba9691f6a47f3bfb0bb4143ef7dc7;1 6611E63499885;1" May 11 23:58:59 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 53876/42866, session timer restarted May 11 23:58:59 65429 rtpproxy[24207]: INFO:handle_command: RTP packets from callee will be resized to 80 milliseconds May 11 23:58:59 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27801_10 42866 64.120.57.111#012" May 11 23:59:00 65429 rtpproxy[24207]: DBUG:get_command: received command "27802_14 LZ80c18 008e2697e52a4adda7be16358b318def 85.13.205.178 26728 907ba9691f6a47f3bfb0bb4143ef7dc7;1 6611E63499885;1" May 11 23:59:00 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 53876/42866, session timer restarted May 11 23:59:00 65429 rtpproxy[24207]: INFO:handle_command: RTP packets from callee will be resized to 80 milliseconds May 11 23:59:00 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27802_14 42866 64.120.57.111#012" May 11 23:59:02 65429 rtpproxy[24207]: DBUG:get_command: received command "27800_13 LZ80c18 008e2697e52a4adda7be16358b318def 85.13.205.178 26728 907ba9691f6a47f3bfb0bb4143ef7dc7;1 6611E63499885;1" May 11 23:59:02 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 53876/42866, session timer restarted May 11 23:59:02 65429 rtpproxy[24207]: INFO:handle_command: RTP packets from callee will be resized to 80 milliseconds May 11 23:59:02 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27800_13 42866 64.120.57.111#012" May 11 23:59:06 65429 rtpproxy[24207]: DBUG:get_command: received command "27801_11 LZ80c18 008e2697e52a4adda7be16358b318def 85.13.205.178 26728 907ba9691f6a47f3bfb0bb4143ef7dc7;1 6611E63499885;1" May 11 23:59:06 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 53876/42866, session timer restarted May 11 23:59:06 65429 rtpproxy[24207]: INFO:handle_command: RTP packets from callee will be resized to 80 milliseconds May 11 23:59:06 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27801_11 42866 64.120.57.111#012" May 11 23:59:10 65429 rtpproxy[24207]: DBUG:get_command: received command "27802_15 LZ80c18 008e2697e52a4adda7be16358b318def 85.13.205.178 26728 907ba9691f6a47f3bfb0bb4143ef7dc7;1 6611E63499885;1" May 11 23:59:10 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 53876/42866, session timer restarted May 11 23:59:10 65429 rtpproxy[24207]: INFO:handle_command: RTP packets from callee will be resized to 80 milliseconds May 11 23:59:10 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27802_15 42866 64.120.57.111#012" May 11 23:59:20 65429 rtpproxy[24207]: INFO:rxmit_packets: callee's address latched in: 85.13.217.158:26728 (RTP) I don't what is going wrong here, but somehow rtpproxy is not relaying the packets from callee's end. But the addresses seems fine and whenever i remove the "z80" inside rtpproxy_answer(), It works fine. At first i thought its just that the resizer is broken. But when the ivr is played, the rtppackets of the ivr works just fine. -- -Cheers -Arif

1 0

Help diagnosing non-response from incoming call
by Alex Villacís Lasso 09 May '14

09 May '14

I have this setup for kamailio + asterisk, in which kamailio is supposed to listen on all ethernet interfaces on UDP port 5060, and will forward traffic from/to asterisk running on the same machine, and listening on localhost, udp port 5080. The scenario for the problematic call is somewhat like this: SIP-PROVIDER<--->NAT<--192.168.0.0/16-->eth0:192.168.10.10<--KAMAILIO-->127.0.0.1<--ASTERISK Our firewall/NAT has been configured to redirect SIP traffic from the SIP provider to the kamailio+asterisk machine at IP 192.168.10.10. The attached file sip-traffic-from-eth0.txt shows a tcpdump capture of an incoming call at eth0. Then, kamailio is supposed to redirect this traffic to asterisk. The attached file sip-traffic-from-localhost.txt shows a tcpdump capture of the same call at 127.0.0.1. The issue is that the INVITE is received, then routed to asterisk, which sends back the 200 OK, but then there is no ACK from the SIP provider. From the point of view of the caller of the SIP provider, the destination just keeps ringing until timeout. Am I right in assuming that the two Record-Route headers should not appear on the traffic as seen from eth0, and that they are the source of the trouble? Can you see any additional issues I might have not seen in the traffic?

1 0

Multiple port configuration
by Alex Villacís Lasso 09 May '14

09 May '14

How do I configure kamailio to listen to multiple ports, on all known interfaces? I have tried "listen=udp:*:5060", "listen=udp:*:5062" on separate lines, but I get the following errors: Not starting : invalid configuration file! 0(7806) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 190, column 12: syntax error 0(7806) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 190, column 12: ip address, interface name or hostname expected 0(7806) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 190, column 13: ERROR: bad config file (3 errors) I have also tried "port=5060", then "port=5062" on separate lines, but kamailio only listens on the port indicated by the last line. I do not want to have to enumerate every single interface on the configuration file.

2 1

route(RELAY) - what exactly execute?
by Patrik Kristel 09 May '14

09 May '14

Hello, I would like to ask what exactly means route(RELAY)? I don't found some declaration of it. I{ found it in many method before exit; I think its probably last handle of SIP message, am I right? Thank you Patrik

1 0

/tmp/kamailio_fifo is missing
by Patrik Kristel 09 May '14

09 May '14

Hello all, I trying to debug Errorwith mi_fifo an d I found out that file kamailio_fifo is missing from the directory /tmp, also I not found it in my Debian by locate command. I have correctly following lines in my conf files: in kamailio.cfg: loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") in kamctlrc: OSER_FIFO="/tmp/kamailio_fifo" So, I dont know, it maybe my mysql crashed? Thank you for you help! Patrik

2 2

kamailio load balancer despatch to asterisk12 plus PJSIP Stack Mediagateways
by Rainer Piper 09 May '14

09 May '14

Hi, I'm using kamailio as a load balancer to dispatch to 5 asterisk Media gateways with pjsip Stack The old asterisk sip Stack is turned off. hier my asterisk pjsip.conf to trust kamailios IP Addr. pjsip.conf [transport-udp] type=transport protocol=udp bind=0.0.0.0:5060 tos=cs3 cos=3 [kamailio] type=endpoint context=outgoing-kamailio disallow=all allow=g722,alaw,ulaw transport=transport-udp direct_media=no disable_direct_media_on_nat=yes aors=kamailio [kamailio] type=aor qualify_frequency=60 [kamailio] type=identify endpoint=kamailio match=[IP-Addr-of-your-kamailio] extensions.conf exten => _XXXX.,1,Dial(PJSIP/${EXTEN}@trunkprofil) od exten => _XXXX.,1,Dial(PJSIP/trunkprofil/${EXTEN}) works great and pjsip is very fast !!! love it !!! Regards Rainer -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 <callto:004922897167161>

1 0

Starting Kamailio with init script
by VOIP Tests 09 May '14

09 May '14

Hello, can someone let me know how I can start Kamailio with the init script? In the /etc/default/kamailio file I have commented out RUN_KAMAILIO=yes, this starts Kamailio but my phones do not register. If I stop the init script and start kamailio with kamctl then my phones register, what am I doing wrong here? Also, if my understanding is correct, if I start Kamailio with the init script I should not start kamailio with kamctl, in that case how do I run the kamctl commands, for eg like kamctl ul show ? Thanks for the help. Arun

3 4

Re: [SR-Users] issue with usr_loc db_mode
by Daniel-Constantin Mierla 09 May '14

09 May '14

Hello, can you enable query log in mysql and verify if they happen in such case? I checked and for db_mode=1, it should immediately do the query. For db_mode=3, th queries are done on time. Also, is there any other application that could potentially delete records? Because after the first registration of a phone (which does an INSERT), there will be UPDATE queries, and if the record is not there, there is nothing to update. Look at usrloc module parameters, there are some parameters to try to do insert if the update doesn't affect records, you can try with this one to see if there is a difference. Cheers, Daniel On 08/05/14 18:20, Henry Fernandes wrote: > There are no errors in the log about this. > > Yesterday, I tried db_mode=2 (with default 'timer_interval' of 60 seconds) and it's better now, although still not working properly. There are still some records that are not in the 'location' table in the database. > > For example, I have a phone that has a registration expirations are set to 1 hour (3600 seconds). Currently, this record is not in the database however "kamctl ul show" shows the following: > > AOR:: phoneUsername > Contact:: sip:phoneUsername@a.b.c.d:62544 Q= > Expires:: 3123 > Callid:: f1e0aa72-eb0d28e2-c3737132(a)10.1.0.14 > Cseq:: 52 > User-agent:: PolycomVVX-VVX_500-UA/4.1.6.4835 > State:: CS_SYNC > Flags:: 0 > Cflag:: 0 > Socket:: udp:a.b.c.d:5060 > Methods:: 8159 > Ruid:: uloc-536a9e57-3137-2 > Reg-Id:: 0 > > In other words, it's been more than 60 seconds but the database is not up to date. This is similar to what I see with db_mode=1. In that case, the database is often missing more than 50% of the registered phones. > > Here are the parameters I'm using for the USRLOC module. > > modparam("usrloc", "db_url", WRITE_DBURL) > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "use_domain", 0) > > -H > > > On 2014-05-08, at 1:00 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > >> Hello, >> >> any db_mode>0 in 3.3 is safe for not losing registrations upon kamailio restart (e.g., db_mode=2 writes to db on time and at shutdown, so nothing is lost as well). db_mode=1 should do that in realtime, indeed. >> >> I'm not aware of any issue with db_mode=1, being used in quite some deployments. Can you check the syslog to see if there is any error reported there? >> >> If you use mysql, then you can enable logging the queries and then check if for each registration there is a mysql query to insert/update/delete a record. >> >> Cheers, >> Daniel >> >> On 08/05/14 00:08, Henry Fernandes wrote: >>> I'm using Kamailio 3.3 and the USRLOC module to track registrations. I have set "db_mode" to 1 because I want to store all registration info to the database and I want the database to be up to date (in case I need to restart Kamailio). >>> >>> Unfortunately, this doesn't work as I expect. When I query the database, it has far fewer entries in the "location" table than are registered to Kamailio (which I find with a "kamctl ul show"). Also, when I restart Kamailio, I lose some of the registrations for my phones. >>> >>> Am I misunderstanding the behaviour for db_mode=1? Isn't it supposed to keep the database up to date? >>> -H >>> _______________________________________________ >>> 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> >> >> _______________________________________________ >> 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

What do next?
by Jaafar Hamza 09 May '14

09 May '14

Hi, I am a student in Programming, I develop for project of the end of study a sip phone(Swing application). I use Sip Jain api. I installed kamailio in Ubuntu Server 14.04 using this command : apt-get install kamailio*. Now, I don't know what to do to configure this server like a registrar and I would like use the presence module. Please help me, I need it. Thanks Hamza Jaafar

2 1

sip trunk help
by Joli Martinez 09 May '14

09 May '14

Hello, I am new to Kamailio. I need to get a SIP trunk registered to Kamailio. Can someone explain how to set this up. Thanks

3 3

issue with usr_loc db_mode
by Henry Fernandes 08 May '14

08 May '14

I'm using Kamailio 3.3 and the USRLOC module to track registrations. I have set "db_mode" to 1 because I want to store all registration info to the database and I want the database to be up to date (in case I need to restart Kamailio). Unfortunately, this doesn't work as I expect. When I query the database, it has far fewer entries in the "location" table than are registered to Kamailio (which I find with a "kamctl ul show"). Also, when I restart Kamailio, I lose some of the registrations for my phones. Am I misunderstanding the behaviour for db_mode=1? Isn't it supposed to keep the database up to date? -H

3 3

n00b question -- RADIUS authentication.
by Måns Nilsson 08 May '14

08 May '14

[first post to list] Greetings, I'm in the process of getting a Kamailio 3.3.2 installation authenticating its SIP accounts against a RADIUS database. There are -- at the moment -- no plans to do any fancy accounting nor any authorisation beyond simple authentication. I've set up and tested a FreeRadius 2.2.3_1 server on a dedicated server. After a fairly steep learning curve involving RADIUS dictionaries I've come so far that kamailio sends out a RADIUS Access-Request message that is received by FreeRadius, processed, and returned to Kamailio which promptly ignores it and continues to send 401 to the client; the SIP message exchange with the client being: (some identifing info has been redacted) > REGISTER sip:my.domain SIP/2.0 < SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.25.191.24:41688;branch=z9hG4bK-d8754z-eac09e6c626d4c4d-1---d8754z-;rport=41688 > REGISTER sip:my.domain SIP/2.0 Via: SIP/2.0/UDP 10.25.191.24:41688;branch=z9hG4bK-d8754z-4f25c643f4b93465-1---d8754z-;rport < SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.25.191.24:41688;branch=z9hG4bK-d8754z-4f25c643f4b93465-1---d8754z-;rport=41688 The RADIUS exchange: 10:07:10.861063 IP (tos 0x0, ttl 64, id 14964, offset 0, flags [none], proto UDP (17), length 270) 10.24.194.198.63712 > 10.24.194.196.1812: [udp sum ok] RADIUS, length: 242 Access Request (1), id: 0x05, Authenticator: 4215e95809551826eda76972be4106c4 Username Attribute (1), length: 18, Value: mtu-06(a)my.domain 0x0000: 6d74 752d 3036 4069 706b 2e73 722e 7365 Unknown Attribute (207), length: 10, Value: 0x0000: 0a08 6d74 752d 3036 Unknown Attribute (207), length: 13, Value: 0x0000: 010b 6970 6b2e 7372 2e73 65 Unknown Attribute (207), length: 36, Value: 0x0000: 0222 5532 6448 326c 4e6e 5271 3677 4353 0x0010: 6463 6775 5056 3050 516e 3936 324d 5635 0x0020: 6d34 Unknown Attribute (207), length: 17, Value: 0x0000: 040f 7369 703a 6970 6b2e 7372 2e73 65 Unknown Attribute (207), length: 12, Value: 0x0000: 030a 5245 4749 5354 4552 Unknown Attribute (207), length: 8, Value: 0x0000: 0506 6175 7468 Unknown Attribute (207), length: 12, Value: 0x0000: 090a 3030 3030 3030 3031 Unknown Attribute (207), length: 36, Value: 0x0000: 0822 3933 3832 3333 3333 3530 3162 3238 0x0010: 6439 3236 3739 3863 3964 3038 6539 3134 0x0020: 3733 Unknown Attribute (206), length: 34, Value: 0x0000: 3538 3665 3336 3763 3230 3163 3137 6438 0x0010: 6261 3265 3830 3533 3763 6433 3562 3761 Service Type Attribute (6), length: 6, Value: #15 0x0000: 0000 000f Unknown Attribute (208), length: 8, Value: 0x0000: 6d74 752d 3036 NAS Port Attribute (5), length: 6, Value: 5060 0x0000: 0000 13c4 NAS IP Address Attribute (4), length: 6, Value: 10.24.194.198 0x0000: c079 c2c6 10:07:10.863964 IP (tos 0x0, ttl 64, id 28916, offset 0, flags [none], proto UDP (17), length 48) 10.24.194.196.1812 > 10.24.194.198.63712: [bad udp cksum 0x06ac -> 0x44c0!] RADIUS, length: 20 Access Accept (2), id: 0x05, Authenticator: 8f07de871a066aacfbe822e20a9b96c1 The RADIUS part of the Kamailio config is: if (is_method("REGISTER") || from_uri==myself) #if (is_method("REGISTER") ) { # authenticate requests xlog("L_INFO", "authenticate [$fd]\n"); ### RADIUS ### if (!radius_www_authorize("my.domain")) { $var(ret) = $rc; xlog("L_INFO", "response code: [$var(ret)]\n"); switch ($var(ret)) { case -7: send_reply("500", "Server Internal Error"); exit; case -1: send_reply("400", "Bad Request"); exit; default: }; if (defined($avp(digest_challenge)) && ($avp(digest_challenge) != "")) { append_to_reply("$avp(digest_challenge)"); }; send_reply("401", "Unauthorized"); exit; }; # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } Any clues? What is missing from my narrative? -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 Actually, what I'd like is a little toy spaceship!!

2 3

kamctl online and ul show are empty!
by 5060 08 May '14

08 May '14

Hello: after i starting kamailio for while, the system shows: 3(1466) DEBUG: mi_fifo [mi_parser.c:84]: end of fifo input tree 3(1466) DEBUG: mi_fifo [fifo_fnc.c:507]: done parsing the mi tree 1(1464) DEBUG: usrloc [urecord.c:327]: Binding '102',' sip:102@192.168.1.105:5060' has expired 3(1466) DEBUG: mi_fifo [mi_parser.c:84]: end of fifo input tree 3(1466) DEBUG: mi_fifo [fifo_fnc.c:507]: done parsing the mi tree 3(1466) DEBUG: mi_fifo [mi_parser.c:84]: end of fifo input tree 3(1466) DEBUG: mi_fifo [fifo_fnc.c:507]: done parsing the mi tree ---------------------------------------------------------------------- kamctl online and ul show are empty. how can i adjust the time to resolve the problem? thanks!

2 3

Re: [SR-Users] UPDATE --- Can "presence" module handle publish reqs with several dialog entries? + Problem SOLUTION
by Klaus Feichtinger 08 May '14

08 May '14

Pls find attached the patch for version 4.1.3. The three lines of code are equal for all affected version (incl. 3.2.4, 3.3.5, 4.0.6 and 4.1.3). regards, Klaus > Daniel-Constantin Mierla <miconda(a)gmail.com> hat am 7. Mai 2014 um 15:45 > geschrieben: > > Hello, > > can you send the patch to be reviewed? > > Cheers, > Daniel > > On 07/05/14 13:15, Klaus Feichtinger wrote: > > > > Hi Daniel, > > > > I think we´ve found the reason, why this problem occurs! > > > > The problem is caused in the "agregate_xmls" function in file > > "notify_body.c" of the "presence_dialoginfo" module: > > > > /* loop over all bodies and create the aggregated body */ > > for(i=0; i<j; i++) > > { > > /* LM_DBG("[n]=%d, [i]=%d, [j]=%d xml_array[i]=%p\n", n, i, j, > > xml_array[j] ); */ > > p_root= xmlDocGetRootElement(xml_array[i]); > > if(p_root ==NULL) { > > LM_ERR("while geting the xml_tree root element\n"); > > goto error; > > } > > if (p_root->children) { > > for (node = p_root->children; node; node = node->next) { > > if (node->type == XML_ELEMENT_NODE) { > > LM_DBG("node type: Element, name: %s\n", > > node->name); > > /* we do not copy the node, but unlink it and then > > add it ot the new node > > * this destroys the original document but we do > > not need it anyway. > > * using "copy" instead of "unlink" would also copy > > the namespace which > > * would then be declared redundant (libxml > > unfortunately can not remove > > * namespaces) > > */ > > if (!force_single_dialog || (j==1)) { > > xmlUnlinkNode(node); > > if(xmlAddChild(root_node, node)== NULL) { > > LM_ERR("while adding child\n"); > > goto error; > > } > > > > It seems to be not the best idea to "unlink" the XML node (= > > "xmlUnlinkNode(node);"), as then no "node->next" is available any more. > > Therefore, this loop will _always_ stop after one dialog entry and ignore > > any additional one! > > > > But we "solved" the problem in this way that the loop will not directly > > use "node->next", but a variable, which is set within the loop. It looks > > like this: > > xmlNodePtr next_node = NULL; > > [...] > > if (p_root->children) { > > for (node = p_root->children; node; node = next_node) { > > next_node = node->next; > > if (node->type == XML_ELEMENT_NODE) { > > [...] > > > > This solution has been tested with 2 and 3 dialog entries in a single > > PUBLISH request and it is working fine. We should discuss if this is a > > problem that should be solved generally or if it is a "private" problem in > > our use case. > > > > What do you mean? > > > > regards, > > Klaus > > > > > > >

2 1

Problems this morning using Iptel.org
by swill 08 May '14

08 May '14

Hi, I use a configuration of ipkall and iptel.org, Most of the time this works out as a great incoming solution, however, today, it seems to be ending the call as i answer, I use only one softpphone at a time, but have tried it on Zoiper, 3cx, and Phonerlite, It does the same thing, call it, it rings, answer it and it immediately goes to "Call ended". I've done powercycle reboots on my HTC EVO 4G LTE, and on my PC it acts the same on all three. It registers fine on them all, I just cant determine if it is an IPKall or an Iptel problem..Is anyone else having an issue?? Are there any suggestions on what further troubleshooting I could do?? Thanks Michael

2 1

Kamailio auto restart if stopped
by JR Richardson 07 May '14

07 May '14

Hi All, Just had a strange occurrence where kamailio has been up and running fine for several months and just died with a strange error: root@sip-router3:~# kamailio -V version: kamailio 4.1.1 (x86_64/linux) ac7d55 May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2275]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7faf565586d0), called from <core>: mem/shm_mem.c: sh_realloc(88), first free <core>: mem/shm_mem.c: sh_realloc(88) - aborting May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2271]: ALERT: <core> [main.c:775]: handle_sigs(): child process 2275 exited by a signal 6 May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2271]: ALERT: <core> [main.c:778]: handle_sigs(): core was not generated May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2271]: INFO: <core> [main.c:790]: handle_sigs(): INFO: terminating due to SIGCHLD May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2279]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2277]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2273]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2272]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2274]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2276]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2278]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2271]: ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Not sure what that means, memory leak? I started started it right back up with no indication of a problem. Running Kamailio for many years, I have never had an instance of it just shut down before. So I'm wondering if there are any safe scripts that check the running process for kamailio and restart if automatically if not running? I dug around a bit but not seeing any hits for auto start kamailio scripts. And another question would be if this occurrence of shutting down should be of concern or not? Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Can "presence" module handle publish reqs with several dialog entries?
by Klaus Feichtinger 07 May '14

07 May '14

Hi, during simulating scenarios in relation with the notify problem I have so far, I became unsure, if Kamilio (in principal) is supporting handling of PUBLISH requests (event "dialog") that contain _more than_ one (1) dialog entry. I know that it can generate notify requests with a message body that contains more than one dialog entry. But that´s not automatically the same as handling publish requests.... So - is Kamailio able or not? Did anybody ever use it in such a scenario? regards, Klaus

2 7

Problem initial configuration of Kamalio
by Mr.Gab 07 May '14

07 May '14

Hello everybody, i'm new to kamalio and i'm trying to install on a centos server hosted on a open-stack IaaS platform. I followed the official wiki adding the "advertise statement" and define WITH_NAT. I added 3 user as test 100 101 102 that i can see typing kamctl db show subscriber but if i try to register with Bria i get Failed. Theese are sip packets that i sniffed with tcpdump. http://pastebin.com/sa74HJqM can someone help a noob to understad how to debug this kind of problem with kamalio? i tried googling but i didn't find anything. Thank you all in advice

1 0

Routing incoming calls to Asterisk trunk
by Teleport Engineer 07 May '14

07 May '14

Hello, I'd setup a test environment where Kamailio accepts external invites and routes them to an internal Asterisk (actually working as an SBC). My dialplan works fine routing the INVITE to the internal Asterisk, but the SIP message goes into the defaul sip context (the one configured in sip general), and not in the context defined for the sip trunk between Kamailio and Asterisk. The sip trunk is configured like this: [kamailio] disallow=all host=kamailio port=5060 insecure=port,invite type=friend context=from-sip-external fromdomain=mydomain qualify=yes allow=alaw allow=g722 So I expect that all the sip invites coming from kamailio should be executed in from-sip-external context. But it doesn't work like this since each invite goes in the default context. The invite that goes to Asterisk is like this <--- SIP read from kamailio-external:5060 ---> INVITE sip:extension@asterisk:5060;transport=UDP SIP/2.0 Record-Route: <sip:kamailio-external:5060;nat=yes;ftag=905e4921;lr=on> Via: SIP/2.0/UDP kamailio-external:5060;branch=z9hG4bK277a.aea3ad87.0 Via: SIP/2.0/UDP caller:28021;rport=28019;received=caller;branch=z9hG4bK-d8754z-d11f1fa1d0fb2ab4-1---d8754z- Max-Forwards: 69 Contact: <sip:524@caller:28019;transport=UDP> To: <sip:extension@kamailio-external> From: <sip:524@caller:28007>;tag=905e4921 Call-ID: YzEzMDM1NDZhZTllMTRiMDJmY2MyZTE0ZWZjMzE5OTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.2.21357 r21367 Allow-Events: presence, kpml Content-Type: application/sdp Content-Length: 260 What I think is that in the sip message there is no information about kamailio internal ip address but only about kamailio-external address and then Asterisk put it into the default context. If this is the reason, should I change the trunk ip address in Asterisk and should I do some manipulation on the incoming packet on kamailio? Or maybe I should use kamailio-external as fromdomain in the trunk configuration? Thanks.

2 1

Asterisk-Kamailio set up
by VOIP Tests 06 May '14

06 May '14

Hello, I have a set up with Asterisk-Kamailio as explained in http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. This set up has been working well for us for sometime now. We are now moving Asterisk to a stand alone server ( not installing it on the same box as asterisk ) but we are running into some issues. I have just made some changes to the Asterisk bind IP address and port and also set up Kamalio as a peer on asterisk. When I make an extension to extension call the calls are failing from the FROMASTERISK route ( call is getting cancelled ). Can someone let me know what changes I need to make to correct the issue? Some pointer will help. Thank you, Arun

2 2

How to enable TLS compression
by 刘日新 06 May '14

06 May '14

Hi, all. I has configure kamailio.cfg as below: #!ifdef WITH_TLS modparam("tls", "session_id", "vic22") modparam("tls", "session_cache",1) modparam("tls", "tls_disable_compression",0) modparam("tls", "config", "/usr/cfg/tls.cfg") #!endif I was sure that this configure segment was included , in case that I set a wrong path to modparam("tls", "config", "/wrong paths/tls.cfg") the kamailio can`t boost and raised error. But, when I debug this configure with gdb, I found that, within method : init_tls_compression(), Line 384 of tls_init.c 384 } else if (cfg_get(tls, tls_cfg, disable_compression)){ 385 LOG(L_INFO, "tls: init_tls: disabling compression...\n"); 386 sk_SSL_COMP_zero(comp_methods); The param value from cfg was always 1, namely: * print ((struct cfg_group_tls *)tls_cfg)->disable_compression 1 Why that? Are there something wrong? B.R. Rixin liu

4 3

Dispatcher question
by Volkan Oransoy 06 May '14

06 May '14

Hi all, I have a multidomain setup that I share load, among two freeswtich servers with a kamailio loadbalancer using dispatcher. For specific applications like queues, I need to match a sip domain with a freeswitch server. A domain should match same freeswitch server on every request. Which algorithm should I use with ds_selecy_dst, or what is the most appropriate way to do this? Thanks. /Volkan

2 1

Port 5061 problem
by aawaise 05 May '14

05 May '14

Hello, I have configured TLS on kamailio server. Problem is that in kamailio.cfg, when I use command port =5061. Kamailio doesnot start and error given in logs is >> ERROR: tcp_init : bind(a, 0x9007f4, 16) on MyIP:5061 : Adress already in >> use >> ERROR: Error while initializing TCP port of TLS Socket MyIP:5061 I tried the command " netstat -lntp | grep 5061 " to see what service or app is using the port 5061. I found no result. It is not used by any application apparently. Any help will be highly appreciated. Thanks. Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Port-5061-problem-tp127464.html Sent from the Users mailing list archive at Nabble.com.

5 6

Location table not updated
by VOIP Tests 05 May '14

05 May '14

Hello, can someone let me know the reason that the location table does not get updated? Out of 15 extension only a few registered phones details get updated in this table and the rest are not. But I can see all the extension in Kamailio when I run kamctl ul show --brief. Thank you, Arun

3 2

SIP over TLS
by aawaise 05 May '14

05 May '14

Hello, I am running a kamailio server with SIPS over TLS. Now I have to connect an android based SIP client to server. The SIP client has no support for SIPS over TLS. But it can run SIP over TLS. First I want to know the difference between the two ?? Secondly how can I make my server accept SIP connection over TLS ? Thanks. Cheers, Owais -- View this message in context: http://sip-router.1086192.n5.nabble.com/SIP-over-TLS-tp127469.html Sent from the Users mailing list archive at Nabble.com.

2 3

sip trunk
by Joli Martinez 04 May '14

04 May '14

Hello, I am setting up Kamailio as class4 softswitch for wholesale minutes. I am trying to figure out how to add a SIP trunk to kamailio. thanks,

1 0

Kamailio and mysql connection error: db_mysql [km_my_con.c:109]: driver error:
by toasterisk 04 May '14

04 May '14

hello, all of users: I am using this link for kamalio+asterisk+mysql: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb i do everything based on the link, only different thing is I installed Kamalio+mysql at same server, and asterisk in another server. currently, my asterisk can connect with the remove MYSQL server, but kamailio can not connect with that. when start up the kamailio, i got the error message: ------------------------------------------ 0(3739) DEBUG: <core> [sruid.c:100]: root for sruid is [uloc-5365e604-e9b-] (0 / 18) 0(3739) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (1): ctl 0(3739) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (1): mi_rpc 0(3739) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (1): acc 0(3739) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (1): auth_db 0(3739) DEBUG: <core> [db.c:316]: connection 0xb71aab90 not found in pool 0(3739) DEBUG: db_mysql [km_my_con.c:94]: opening connection: mysql://xxxx:xxxx@localhost/asterisk 0(3739) ERROR: db_mysql [km_my_con.c:109]: driver error: Access denied for user 'asterisk'@'localhost' (using password: YES) 0(3739) ERROR: <core> [db.c:320]: could not add connection to the pool 0(3739) ERROR: auth_db [authdb_mod.c:174]: unable to connect to the database 0(3739) ERROR: <core> [sr_module.c:896]: init_mod_child(): Error while initializing module auth_db (/usr/local/lib/kamailio/modules_k/auth_db.so) 0(3739) ERROR: <core> [main.c:1433]: main_dontfork: init_child failed 4(3744) INFO: <core> [main.c:851]: INFO: signal 15 received 3(3743) INFO: <core> [main.c:851]: INFO: signal 15 received 0(3739) INFO: <core> [main.c:797]: INFO: dont_fork turned on, living on 0(3739) ALERT: <core> [main.c:782]: child process 3743 exited normally, status=0 0(3739) ALERT: <core> [main.c:782]: child process 3744 exited normally, status=0 0(3739) INFO: <core> [main.c:797]: INFO: dont_fork turned on, living on 0(3739) DEBUG: <core> [db_pool.c:97]: connection still kept in the pool 0(3739) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(3739) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(3739) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(3739) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(3739) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done 0(3739) DEBUG: <core> [mem/shm_mem.c:242]: shm_mem_destroy 0(3739) DEBUG: <core> [mem/shm_mem.c:245]: destroying the shared memory lock --------------------------------------------------------------------------------------------------------------------------- [MySQL-asterisk] Description = MySQL Asterisk database Trace = Off TraceFile = stderr Driver = MySQL SERVER = 192.168.1.116 // IP for kamailio and mysql USER = asterisk PASSWORD = 111111 PORT = 3306 DATABASE = asterisk -----------------use isql test--------------- root@kamailio:/etc# isql -v ODBC [IM002][unixODBC][Driver Manager]Data source name not found, and no default driver specified [ISQL]ERROR: Could not SQLConnect root@kamailio:/etc# ----------------------------------------------------------- anyone knows the problem? thanks!

3 2

Location table not getting updated
by VOIP Tests 04 May '14

04 May '14

Hello, can someone let me know what could be the reason that the location table in the kamailiio db does not get updated? There are some extensions that I can see in the location table but not others, but I am still able to make extension to extension calls to all extensions. This is a multi domain set up. Thank you, Arun

2 5

xmlops & XPath question
by Alex Balashov 04 May '14

04 May '14

Hello, I don't know a lot about XML or XPath, so I was hoping someone could help me out. I'm trying to parse out 'Event: dialog' state notifications from NOTIFYs using xmlops. They look like this: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="5" state="full" entity="sip:abalashov-roaming@xxx.xxx.xxx.xxx"> <dialog id="0588e460-d293-11e3-8563-538b9e69fee5" direction="initiator"> <state>terminated</state> </dialog> </dialog-info> So, I do: $xml(u=>doc) = $rb; xlog("L_INFO", "State: $xml(u=>xpath:/dialog-info/dialog[1]/state/text())\n"); This produces a blank result. I've tried numerous other variations, such as: xpath:/dialog-info/dialog/state/text() None of them work, either. When I upload the above XML body to various online XPath validation tools and try my expressions, they work fine. Any ideas as to what I'm missing? Do I need to indicate the namespace in the XPath expression somehow? If so, how do I do it? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 5

[sr-user] textops module - search_append_body(re, txt) - append new line
by Cockhootec Yahrabee 03 May '14

03 May '14

Hi, I am trying to append new field in SDP and I need to append new line but I don't know how. I tried to google for solution but I didn't find anything. I want to route my RTP and RTCP packet to port 5060 (this is for development of my school project, I know that in the context of correct SIP this has no sense but this is only experimental). My sample kamailio.cfg code: if (is_method("INVITE")) { # route RTP packets to 5060 if (search_body('m=audio [0-9]+')) { subst_body('/^m=audio [0-9]+/m=audio 5060/g'); } # route RTCP packets to 5060 if (search_body('a=rtcp:[0-9]+ IN IP4 [0-9]+.[0-9]+.[0-9]+.[0-9]+')) { subst_body('/^a=rtcp:[0-9]+ IN IP4 [0-9]+.[0-9]+.[0-9]+.[0-9]+/a=rtcp:5060 IN IP4 192.168.11.129/g'); } else { search_append_body('m=audio [0-9]+ RTP/AVP [0-9]+ [0-9]+ [0-9]+ [0-9]+ [0-9]+ [0-9]+ [0-9]+', ' *{NEW_LINE_NEEDED_HERE}*a=rtcp:5060 IN IP4 192.168.11.129'); } } } Could someone please help me resolve what should I replace the {NEW_LINE_NEEDED_HERE} with to add new line after 'm=audio' field? Also If you have some idea how should I optimize my regexp for finding 'm=audio'/'a=rtcp' fields I would appreciate your help.

1 0

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users