- sr-users - kamailio.org

BUG: ip_addr2sbuf: unknown address family 0
by AliReza Khoshgoftar Monfared 16 Jul '14

16 Jul '14

Hi, I am running a simple command on timer in a kamailio proxy. Here is the related parts of the config scripts: loadmodule "timer" > modparam("timer", "declare_timer", "tmr1=LOG_STATS,1,fast,enable"); > > route{ > ... > } > > route[LOG_STATS] { > exec_avp("netstat -ul | grep ':sip' | awk '{print > $$2}'","$avp(s:test)"); > xlog("L_ALERT","logged RecvQ as $avp(s:test)"); > return; > } > Each time I run the "netstat" using the exec() inside the timer route block, I get the following error from Kamailio in my log: kamailio[28546]: : exec [../../parser/../ip_addr.h:669]: ip_addr2sbuf(): > BUG: ip_addr2sbuf: unknown address family 0 > What is the meaning of this error, and what does kamailio mean by reporting it as a "BUG"? Is there any obvious reason, I am getting this? Is it related to calling netstat to survey the RecvQ? Thanks, Alireza

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have problem with iptel account
by Samhu 16 Jul '14

16 Jul '14

hi, This is Sam from Draytek Crop. We have a IPPBX device with iptel for SIP account as below. User B dial to IPPBX could heart the answer from Auto Attendant of IPPBX. Then user B cannot heart anything (after input the extension of user A) when user A pickup his telephone . At this momemt, user A could hear everything from user B. This Network Architecture worked fine before. And it still works fine now if we select the account of "draytel.org". Please show us how to solve the problem. Best Regards, Sam Hu

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SCA and PRESENCE
by Joel White 15 Jul '14

15 Jul '14

I am having an issue where I enable presence modules and the SCA module stops working. I disable presence and SCA works fine on Polycom phones. Does anyone know what I am missing or doing wrong? Thank you in advance Joel

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function to reset all flags or bflags?
by Juha Heinanen 15 Jul '14

15 Jul '14

there is a function to reset a flag or bflag, but i haven't found one that would reset all flags or bflags. is there a way to do it? if not, is it ok to add such function? -- juha

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Many instances of "new URI shorter than old URI" in log
by Alex Villacís Lasso 15 Jul '14

15 Jul '14

Jun 27 15:28:42 elx /usr/sbin/kamailio[25189]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:44 elx /usr/sbin/kamailio[25187]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:48 elx /usr/sbin/kamailio[25188]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:49 elx /usr/sbin/kamailio[25188]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI I am not really sure of what this message is supposed to indicate. It looks something related to uac_restore_[from,to] functions, but some debugging statements just before the use show that the message is not originating from there. This message is showing up hundreds of times per day.

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set_rtp_proxy_set behaviour
by Igor Potjevlesch 15 Jul '14

15 Jul '14

Hello, I set two entries of rtpproxy like : modparam("rtpproxy", "rtpproxy_sock", "1 == udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_sock", "2 == udp:127.0.0.1:7723") For a particular case, I want to use the second RTPProxy and the first one as the default one. If in a conditional block I set set_rtp_proxy("2"), all other call to rtpproxy_manage will use the first RTPProxy? Or there is a load-balancing introduce by default. Regards, Igor.

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Routing calls to Asterisk using dispatch
by Olli Heiskanen 15 Jul '14

15 Jul '14

Hello, I've started playing with an idea to add multiple asterisk servers and using dispatcher to balance the sip load between them. I added the code according to dispatcher module documentation ( http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html), but I think there's something off in my setup: kamctl ul output shows 2 AORs for one client: AOR:: 770(a)testers.com Contact:: sip:770@2.2.2.2:64340;rinstance=c634da314e12385f;transport=UDP Q= Expires:: 3221 Callid:: ZTE1MWYwYzM3NGNjNjMxMmEzM2JjYWNmNzQyZTdiNGI. Cseq:: 2 User-agent:: Z 3.2.21357 r21367 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 5087 Ruid:: uloc-53bfe447-35ae-2a2 Reg-Id:: 0 Last-Keepalive:: 1405174150 Last-Modified:: 1405174150 AOR:: 770(a)1.1.1.1 Contact:: sip:770@1.1.1.1:5070 Q= Expires:: 68 Callid:: 327fcf07641f80006e962821112a61b5(a)testers.com Cseq:: 754 User-agent:: Asterisk PBX 11.10.2 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 4294967295 Ruid:: uloc-53bfe447-35af-a82 Reg-Id:: 0 Last-Keepalive:: 1405174477 Last-Modified:: 1405174477 I don't think I should be seeing an AOR for 770 where Contact is the public address of my server (here 1.1.1.1) and User-Agent which is Asterisk. I'm using Asterisk Realtime integration, and by what I can tell the sip messages are going nicely, client authenticates with Kamailio and sends this message to Asterisk (which is on the same machine; Kamailio at 5060 and Asterisk at 5070): 1.1.1.1.sip > 1.1.1.1.vtsas: SIP, length: 374 REGISTER sip:1.1.1.1:5070 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKbc8a.f4473947000000000000000000000000.0 To: <sip:770@1.1.1.1> From: <sip:770@1.1.1.1>;tag=4a9c3f1c98b9f1c5704acfd1770d93d2-d0c1 CSeq: 10 REGISTER Call-ID: 7ffa0191-13742(a)1.1.1.1 Max-Forwards: 70 Content-Length: 0 Contact: <sip:770@1.1.1.1:5060> Expires: 3600 Currently I can make calls from 770 to 123 which is an Asterisk extension that answers, plays hello world and hangs up. However I can't call another sip clients when I route calls through Asterisk, they do work fine if I don't use Asterisk for handling calls, but I'd like Kamailio to be in the role of proxy/loadbalancer and Asterisk to handle calls. My config is the simple default config, added with realtime stuff and then dispatcher according to the documentation. I wonder if there's something wrong in the REGISTER that Kamailio sends to Asterisk, or maybe something else going wrong? Has anyone seen results like this and do you spot something here that needs fixing? Thanks, Olli

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siremis/ser/xrcmds error help
by MARY 15 Jul '14

15 Jul '14

sr-users，hello：　 http://127.0.0.1:8890/siremis/ser/xrcmds error info: [2014-07-14 08:49:30 (GMT)] An exception occurred while executing this script: Error message: #2, require_once(XML/RPC.php): failed to open stream: No such file or directory Script name and line number of error: /xxxx/siremis-4.1.0/siremis/modules/ser/service/asipto/libs/cmds/serxr.php:2 why? can not find the XML/RPC.php file。 2014-07-1417:56:16 MARY

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Re: [SR-Users] Forward UAC Registration with Authentication (like Upper Registration or Passthrough Registration)
by Angelica Juarez 14 Jul '14

14 Jul '14

Anjelicajuarez desde mi iPhone

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To Rewrite SDP when request goes from kamailio to outside network
by Anurag Mishra 14 Jul '14

14 Jul '14

Hi All, I have been struggling with this issue for a long and could not find a way to solve it. I have setup FreeSWITCH and Kamailio on the single server but on a private server with a private IP and fronted by firewall. Firewall has all the ports opened. FreeSWITCH is running with nonat. Kamailio has public IP advertised on port 5060 and private IP on 5070 for phones on LAN. Problem is with outbound calls there is no audio and reason I can see is contact field in SDP has the private IP of the server instead of the public IP. Internal device to device calls is working fine. I have tried to rewrite the SDP and have written below code, but got no success: request_route > { > xlog("L_DBG", "Request Route Enter"); > if (is_method("INVITE")) > { > xlog("L_DBG", "Request Route Enter in Invite"); > if (has_body("application/sdp")) > { > xlog("L_DBG", "Request Route Enter in SDP of > Invite"); > if (search("IN IP4 PRIVATE_IP")) > { > xlog("L_DBG", "Request Route Search IN IP4 > PRIVATE_IP"); > fix_nated_sdp("8", "PUBLIC_IP"); > fix_nated_sdp("2", "PUBLIC_IP"); > } > } > } > } > Can anyone help me to get rid of this. Thanks, Anurag Mishra

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Re: [SR-Users] how to remove non-digits using dp_translate
by Henry Fernandes 11 Jul '14

11 Jul '14

I don't know how to apply the global flag in Kamailio. Can you give me an example? -H From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: <miconda(a)gmail.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Date: Thursday, 3 July, 2014 1:47 AM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] how to remove non-digits using dp_translate Hello, you have to replace [^0-9] with empty, using the global flag. Cheers, Daniel On 02/07/14 23:11, Henry Fernandes wrote: > > I'd like to use the dialplan module to remove all non-digits from a string. > For example, "1-306-555-1234" becomes "13065551234" and "(306) 555-1234" > becomes "3065551234". > > > > > Unfortunately, I can't figure out the regular expressions for subst_exp and > repl_exp. I can easily replace one instance of a non-digit, however I can't > figure out how to replace all non-digits. > > > > > -H > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listi > nfo/sr-users > -- Daniel-Constantin Mierla - http://www.asipto.comhttp://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

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Comparing strings
by Alex Balashov 11 Jul '14

11 Jul '14

Is there any compelling reason to use textops' cmp_str()/cmp_istr() versus the 'eq' operator (or the implicit string comparison in "a" == "b")? -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

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Would like to upgrade to 4.1.4+wheezy -- when will this be available?
by Will Ferrer 11 Jul '14

11 Jul '14

Hi We would like to upgrade or debian instances to the apt repo for 4.1.4+wheezy. Is this release soon to be stable and ready for use? We would like to upgrade so we can get around the memory leak that arises when many reloads occur as described here: http://sip-router.org/tracker/index.php?do=details&task_id=436 Thank you very much for any advice any one can offer us. All the best. Will Ferrer

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Update Interval for Rtimer
by AliReza Khoshgoftar Monfared 11 Jul '14

11 Jul '14

Hi, I have a basic yet weird problem with Rtimer module. I cannot make it execute a route block for me at an interval of my choice (which is what it is supposed to do, right?) Here is a simplified example with relevant parts of my config script: Assume I want to print "hello" every 100 ms: I have a script like this (I've put all my global variables, in case they matter and my relevant route blocks...) Is there anything I am doing wrong here in the usage of rtimer? *####### Global Parameters #########* > > *#!ifdef WITH_DEBUG* > *debug=4* > *log_stderror=yes* > *#!else* > *debug=-2* > *log_stderror=no* > *#!endif* > > *memdbg=5* > *memlog=5* > > *log_facility=LOG_LOCAL0* > > *fork=yes* > *children=4* > > */* comment the next line to enable TCP */* > *disable_tcp=yes* > > */* uncomment the next line to disable the auto discovery of local aliases* > * based on revers DNS on IPs (default on) */* > *auto_aliases=no* > > */* add local domain aliases */* > *# alias="mysipserver.com <http://mysipserver.com>"* > *dns=no # (cmd. line: -r)* > *rev_dns=no # (cmd. line: -R)* > > *port=5060* > > */* uncomment and configure the following line if you want Kamailio to * > * bind on a specific interface/port/proto (default bind on all > available) */* > *listen=eth0:5060* > > *sip_warning=no* > > *####### Modules Section ########* > > *#set module path* > > *#mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/"* > *loadpath "/usr/lib/x86_64-linux-gnu/kamailio/modules"* > > > *#loadmodule "db_mysql.so"* > *loadmodule "mi_fifo.so"* > *loadmodule "kex.so"* > *loadmodule "tm.so"* > *loadmodule "tmx.so"* > *loadmodule "sl.so"* > *loadmodule "rr.so"* > *loadmodule "pv.so"* > *loadmodule "maxfwd.so"* > *loadmodule "textops.so"* > *loadmodule "siputils.so"* > *loadmodule "xlog.so"* > *loadmodule "xprint.so"* > *loadmodule "sanity.so"* > *loadmodule "ctl.so"* > *loadmodule "mi_rpc.so"* > *loadmodule "acc.so"* > *loadmodule "dispatcher.so"* > *loadmodule "exec.so"* > *loadmodule "rtimer.so"* > *loadmodule "benchmark.so"* > > > *# ----------------- setting module-specific parameters ---------------* > > > *# ----- mi_fifo params -----* > *modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")* > > > *# ----- 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)* > > > *# ----- acc params -----* > *modparam("acc", "log_flag", 1)* > *modparam("acc", "failed_transaction_flag", 3)* > *modparam("acc", "log_extra", * > * > "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si")* > > *# ----- tm params -----* > *modparam("tm", "fr_timer", 2000)* > *modparam("tm", "fr_inv_timer", 40000)* > > *# ----- dispatcher params -----* > *modparam("dispatcher", "list_file", > "/home/ubuntu/Downloads/dispatcher.list")* > *modparam("dispatcher", "flags", 2)* > *modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")* > *modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")* > *modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")* > > *# ----- xlog params -----* > *modparam("xlog", "buf_size", 8192)* > *modparam("xlog", "log_facility", "LOG_LOCAL0")* > > *# ----- rtimer params -----* > *modparam("rtimer", "timer", "name=ta;interval=1;mode=1;")* > *modparam("rtimer", "exec", "timer=ta;route=8")* > > > *####### Routing Logic ########* > > > *# main request routing logic* > *route {* > * ...* > * # calls various routing blocks, but no explicit call to route("8")* *}* *route[8] {* * # to be executed on timer * * xplog("L_ALERT","hello");* *}*

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INVITE to 200, INVITE to 404, INVITE to 487
by Luis F Urrea 11 Jul '14

11 Jul '14

Hi all, I have seen Siremis charts that account stats such as INVITE to 200 and INVITE to 404. I would like to have this sort of accounting but without the use of the Siremis UI, so I am wondering how is this achieved, is this done through the ACC module and correlating responses? TIA

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Path attribute not updated in location table when user switches between two access points behind NAT
by Prasun Bheri 11 Jul '14

11 Jul '14

Hi Group, I stumbled upon a possible bug in Kamailios (4.0.0) Path module. Setup: Proxy server is behind another Kamailio Load Balancer and both (Proxy & LB) are configured to use Path module. When client registers for the first time path is inserted into Location table successfully however when client switches from currently Access Point to another AP (which are both behind a NAT), Path attribute in Location table is not updated with newer <ip>:<port> When client switched from one AP to another, IP remained same however port changed from 48238 to 33038 and subsequent update query issued by Kamailio is as follows: update location set expires='2014-07-11 19:27:32',q=-1.00 ,cseq=1805331004,flags=0,cflags=64,user_agent='xxxxxxxxxx',received='sip:xx5.xx1.xx1.219: *33038*;transport=tcp',path='<sip:xx3.xx1.xx9.53;transport=tcp;lr;received=\'sip:xx5.xx1.xx1.219:33038;transport=tcp\'>',socket='tcp:xx3.xx1.xx9.28:5060',methods=16351,last_modified='2014-07-11 18:27:32',ruid='uloc-53bf7f9c-4254-e',instance='<urn:imei:911304251588897>',reg_id=1 where username='testuser' *AND **contact='sip:testuser@xx5.xx1.xx1.219:33038;transport=tcp' *AND callid='7983ce39-2ffd3a92-8f82f477-e6313128-84b64128' AND domain=' vvoip.xxxxxxxx.com' In the where clause contact should have been old value taken from initial registration, however contact from new registration is used, so where condition always fails and entry is never updated. as a result no further calls or messages could be received by the client. is there any workaround for this issue? PS: Here is the configuration of path module. # set path to lazy mode modparam("registrar", "path_mode", 1) # set to 1 to use path header modparam("registrar", "use_path", 1) # set to 1 to use path received parameter modparam("registrar", "path_use_received", 1) modparam("path", "use_received", 1) Thanks & Regards, Prasun

1 0

Problem with installing rtpengine
by Muhammad Shahzad 11 Jul '14

11 Jul '14

Hi, I am trying to upgrade from mediaproxy-ng to rtpengine trunk version. The compilation steps go well and i have deb packages created. However when i try to install them (on same machine where they compiled), i get this error for every deb package, ngcp-rtpengine-daemon pre-depends on ngcp-system-tools * ngcp-system-tools is not installed.* I can't find this ngcp-system-tools package anywhere, neither on source nor in built deb packages. So, i am stuck. Can anyone help? Thank you.

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sip_trace
by Rafal Skrzeczkowski 10 Jul '14

10 Jul '14

Hi, I'm trying to make configuration that will trace registrations. I can trace REGISTER messages using sip_trace. But how can i trace responses generated by pv_www_authenticate using sip_trace? Rafal Skrzeczkowski

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Listen to to 2 NIC's IP addresses and establish call between
by Raj Roy Ghandhi 10 Jul '14

10 Jul '14

Hi Friends, I have to configure a Kamailio in to meet following requirements 1. PC having 2 NICs (NIC-A and NIC-B) with 2 different IPs (IP-A and IP-B) from 2 different service providers. Totally 2 different IPs. NIC-A attached to provider is a private SIP traffic provider and having IP-A. (no internet access) NIC-B attached to internet service provider and having IP-B. 2. I get SIP traffic to IP-A of NIC-A from SIP traffic provider. 3. I need to forward that SIP traffic to another SIP carrier in remote location using IP-B of NIC-B through internet and have the call establish without any issue. How can I achieve this with single instance of Kamailio running on same machine. Please advice me. Best Regards, Roy.

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crash after 100 trying on NOTIFY
by Kristian F. Høgh 10 Jul '14

10 Jul '14

Hi, We've had the following crash. Kamailio 4.0.4 Client sends SUBSCRIBE, kamailio answers 202. Kamailio sends NOTIFY, client answers 100 trying. fr_timer is 5000. The client doesn't send "200 OK", so kamailio generates 408. Any suggestions? Regards, Kristian Høgh (gdb) bt #0 0x00007f3ab0c91852 in cancel_branch (t=0x7f3a9c779190, branch=0, reason=<value optimized out>, flags=4) at t_cancel.c:284 #1 0x00007f3ab0cd9c9b in reply_received (p_msg=0x7f3ab26ac560) at t_reply.c:2198 #2 0x0000000000458924 in do_forward_reply (msg=0x7f3ab26ac560, mode=<value optimized out>) at forward.c:799 #3 0x00000000004a108e in receive_msg ( buf=0x8dbd80 "SIP/2.0 100 Trying\r\nVia: SIP/2.0/UDP 10.10.10.10;branch=z9hG4bK83f6.32f26ad2.0\r\nTo: sip:554@<domain>;tag=7ce8fb6e\r\nFrom: sip:144@<domain>;tag=93983715339d3cf39b"..., len=<value optimized out>, rcv_info=0x7fff70f07090) at receive.c:272 #4 0x0000000000533ed6 in udp_rcv_loop () at udp_server.c:557 #5 0x000000000046964a in main_loop () at main.c:1638 #6 0x000000000046c4e2 in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2566

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Need help with Routing presence NOTIFY messages
by Veerabhara Gundu 09 Jul '14

09 Jul '14

Hello Experts, I have two Kamailio servers with built in presence servers with load balanced with DNS, both sharing same data base. Registration, authentication and messaging are working without any problem. When it comes to Presence, NOTIFY messages generated by Kamailio is getting rejected by firewall if the client didn't register to that particular Kamailio. In MESSAGES case, I read the socket info from Location table, if the client is not registered to that Kamailio forwarded to the server which serving that user. But in NOTIFY case, these messages are not coming to routing logic at all, Kamailio is directly sending messages by looking at the contact information from active-watchers table. One possible solution I am thinking is, separating presence logic and starting that as separate server. That way I can play with contact header. Any other thoughts, I appreciate any help with this. I have spent enough time on this. Thanks, Veera Thanks, Veera

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Re: [SR-Users] What is q-value ??
by Daniel Grotti 09 Jul '14

09 Jul '14

Hi, q value is used in location table for serial forking. If you have a subscriber with 2 contact registered you can use the q value to load one contact first (the one with the higher q) and afterwards load and call the second one. Highest is the value stored, highest is the priority. Daniel On 9 Jul 2014 06:20, aawaise <owaisulhaq(a)hotmail.com> wrote: > > There is a mention of highest "q value" in documentation of registrar > module's function "lookup(domain [, uri])". > Can some one please explain what this "q value" is ?? > It is used in 4.2 section of following link. > http://kamailio.org/docs/modules/stable/modules/registrar.html#idp1976344 > > Any help will be appreciated. Thanks. > > > > -- > View this message in context: http://sip-router.1086192.n5.nabble.com/What-is-q-value-tp128895.html > Sent from the Users mailing list archive at Nabble.com. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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What is q-value ??
by aawaise 09 Jul '14

09 Jul '14

There is a mention of highest "q value" in documentation of registrar module's function "lookup(domain [, uri])". Can some one please explain what this "q value" is ?? It is used in 4.2 section of following link. http://kamailio.org/docs/modules/stable/modules/registrar.html#idp1976344 Any help will be appreciated. Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/What-is-q-value-tp128895.html Sent from the Users mailing list archive at Nabble.com.

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Using 'reg_fetch_contacts' in Kamailio-3.1.1
by aawaise 09 Jul '14

09 Jul '14

I want to extract domain of a user from Location List For this I found "reg_fetch_contacts(table, uri, profile)" function to be the most suitable function. I want to know when was it added to registrar module ? Can it be used in kamailio version 3.1.1 ? Or I should update my kamailio version ?? Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Using-reg-fetch-contacts-in-Kamaili… Sent from the Users mailing list archive at Nabble.com.

1 0

Access the attributes of a location record
by aawaise 09 Jul '14

09 Jul '14

I want to extract domain attribute of a user from Location Table. I have gone through the readme of registrar module, that has an example of using reg_fetch_contacts() and how to access the attributes of a location record ? But I get the callee domain = NULL when I log using command xlog("callee=>domain: $(ulc(callee=>domain)))\n". Can anyone please help me with the correct syntax. Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Access-the-attributes-of-a-location… Sent from the Users mailing list archive at Nabble.com.

1 0

Help me to configure kamailio on EC2
by Jayaraman, Kamalakannan 09 Jul '14

09 Jul '14

Hi, I had installed kamailio on my AWS EC2, by following the steps in http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x . Since I installed in EC2, i changed the listen=<my-private-ip> advertise=<my-public-ip> in kamailio.cfg file too. I open the 5060 port in AWS console too. Kamailio starts without any error, but I can't able to connect in Jitsi client. Please help me out to do this as soon as possible.

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No sound between chrome and firefox
by Yuriy Gorlichenko 08 Jul '14

08 Jul '14

Hello. I have Kamailio 4.1.3+rtpengine_rtpproxy-ng as module for rtpengine. Kamailio installed as frontend (registration, auth, proxy ) of asterisksk servers. WEBRTC users registred at kamailio and asterisk works as media server. When I try to call from Jssip from Firefox to chrome to way audio is fine. When I call from chrome - I see rtp packets only from firefox. Not from chrome. At kamailio log when I call from chrome log the same as whe i try to call from firefox (I can not see anithing wrong) So somebody else have same issue and may be somebody know reasons of this. Thanks for advice

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sip_trace and registration responses
by Rafal Skrzeczkowski 08 Jul '14

08 Jul '14

Hi, I'm trying to make configuration that will trace registrations. I can trace REGISTER messages using sip_trace. But how can i trace responses generated by pv_www_authenticate using sip_trace? Rafal Skrzeczkowski

1 0

Kamailio, CPL, SIP CGI and alternatives.
by LAA 08 Jul '14

08 Jul '14

Hi all, I'm looking for a mechanism to let users personalize the behaviour of their profile, uploading configuration changes to Kamailio. For example voice mail redirecion, time-based routing, etc. I'm testing CPL scripts (cpl-c module with Kamailio 4.1.3), but I have problems with the behaviour of some features. "reject" and "redirect" seem to be working fine, but "proxy" is not working. ¿Has anybody experienced the same problems? I could hardly find any post regarding CPL scripting with Kamailio. What are the most commonly used alternatives for this purpose with Kamailio? ¿SIP CGI? ¿any other? Best regards. Luis.

2 2

SDP_NAT on Retry
by Timo Klecker 08 Jul '14

08 Jul '14

Hello everybody, we have got a problem with sdp-nat on retries of a 200 OK packet. Due to slow response of our trunk the 200 ok is resend (up to 3 times) but sdp-nat is only done on the first two packets. We are running Kamailio 4.1.0 with the following setup: Trunk --> NAT --> Kamailio --> Voip Bridge The rtp stream from the trunk (public ip) is supposed to go to one of our media gateways (behind 1:1 nat). We use sdp_mangle_ip to replace the private IPs of our Mediagateway with the public ones. This goes well with the first 200 OK packet but not with a retry. A.A.A.A (public trunk media IP) B.B.B.B (our public media IP - 1:1 natted to I.I.I.I) INVITE Trunk --> Kamailio Connection Information (c): IN IP4 A.A.A.A INVITE Kamailio --> Voip Bridge Connection Information (c): IN IP4 A.A.A.A 200 OK Kamailio <-- Voip Bridge Connection Information (c): IN IP4 I.I.I.I 200 OK Trunk <-- Kamailio Connection Information (c): IN IP4 B.B.B.B 200 OK Kamailio <-- Voip Bridge Connection Information (c): IN IP4 I.I.I.I 200 OK Trunk <-- Kamailio Connection Information (c): IN IP4 I.I.I.I ACK Trunk --> Kamailio --> Voip Bridge Our configuration used is the following: route[RELAY] { xlog("L_INFO"," In route Block RELAY \n"); # enable additional event routes for forwarded requests if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (!t_relay()) { xlog("L_NOTICE", " (ROUTE_RELAY) Relay failed, sending error \n"); sl_reply_error(); } exit; } onreply_route[MANAGE_REPLY] { xlog("L_INFO"," In route Block MANAGE_REPLY \n"); if ( isflagset(20) ) { -- we have to do nat route(SDP_NAT); force_send_socket(KAMAILIO_TO_TRUNK_IP); } else { force_send_socket(KAMAILIO_TO_BRIDGE_IP); } } How can we handle the 200 OK retry? Is there a way to do this? Kind regards Timo Klecker

1 0

SIP domain translation between providers
by Jeremy Visser 08 Jul '14

08 Jul '14

G'day folks Struggling to work out whether Kamailio can help me or not in this situation, and I hope someone can tell me whether I'm barking up the wrong tree. I have hundreds of users which register to an old legacy Asterisk box that needs retiring. The "users" are customers consisting of VoIP ATAs, PBXes, softphones, and other hardware phones. I've been told I need to retire the Asterisk system in the next month and move customers to our new provider, who provide the SBC themselves (it's a Broadsoft-based system). Obviously a customer once ported to the new provider requires new SIP proxy and SIP domain settings configured in their CPE. For obvious reasons, I cannot require hundreds of customers to change their configuration overnight. However, there is not enough time to assist hundreds of customers changing their SIP proxy/registrar/domain details. So in the meantime I'm trying to work out a transition solution. I cannot simply change my SIP registrar DNS to be a CNAME for the new provider's registrar, as the domain is different, and the new provider doesn't allow domains other than their own to register with them (even if the username/password portion is correct). My new provider has previously told me they are unwilling to accept my old domain whilst authenticating (they are much bigger than me, and I have no power to change them). So I was hoping to use Kamailio and possibly the Path or Dispatch modules to proxy SIP registration to the new provider, rewriting the domain part of the SIP register string as I go But I can't for the life of me work out whether that's even possible. I have spent weeks poring over various Kamailio configurations posted to the interwebs, but nobody seems to have done this before. Specifically the "translation proxy" -- which I am hoping to make Kamailio -- will need to be stateless, and perform no local authentication. Individual registrations will need to be passed through to the new provider (including authentication!), while still translating the domain in the Request-URI, To, From, and various other SIP headers. Because the current legacy Asterisk box has nat=yes set for all users, I will need to proxy media, because if I start proxying SIP traffic to the new provider, I will be confusing their own NAT hacks. I don't understand the documentation for either the Path or Dispatch modules, and the information on the wiki is confusing and poorly written. I am very familiar with SIP, and have no trouble configuring Asterisk, but trying to work with Kamailio is leaving me feeling like an idiot. However, Kamailio's config is very flexible, so despite this I suspect what I'm trying to achieve may be possible. Any pointers in the right direction? Cheers, Jeremy.

3 2

Strange avp_check behavior
by Klaus Darilion 08 Jul '14

08 Jul '14

Hi! Kamailio 4.1.4. When using avp_check with fnmatch and AVP the result is always TRUE if it is true for 1 time. It only happens with $avp(), but not with $var(). Is this a bug or some special behavior of AVPs? xlog("L_ERR", "======= with $$avp(...)================"); $avp(rpid) = "+499539110"; $avp(pattern) = "+499539[2-5]"; # NO MATCH if (avp_check("$avp(rpid)", "fm/$avp(pattern)/g")) { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): YES"); } else { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): NO"); } $avp(pattern) = "+499539*"; # MATCH if (avp_check("$avp(rpid)", "fm/$avp(pattern)/g")) { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): YES"); } else { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): NO"); } $avp(pattern) = "+499539[2-5]"; # NO MATCH if (avp_check("$avp(rpid)", "fm/$avp(pattern)/g")) { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): YES"); } else { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): NO"); } xlog("L_ERR", "======= with $$var(...)================"); $var(rpid) = "+499539110"; $var(pattern) = "+499539[2-5]"; # NO MATCH if (avp_check("$var(rpid)", "fm/$var(pattern)/g")) { xlog("L_ERR", "$var(rpid) matches $var(pattern): YES"); } else { xlog("L_ERR", "$var(rpid) matches $var(pattern): NO"); } $var(pattern) = "+499539*"; # MATCH if (avp_check("$var(rpid)", "fm/$var(pattern)/g")) { xlog("L_ERR", "$var(rpid) matches $var(pattern): YES"); } else { xlog("L_ERR", "$var(rpid) matches $var(pattern): NO"); } $var(pattern) = "+499539[2-5]"; # NO MATCH if (avp_check("$var(rpid)", "fm/$var(pattern)/g")) { xlog("L_ERR", "$var(rpid) matches $var(pattern): YES"); } else { xlog("L_ERR", "$var(rpid) matches $var(pattern): NO"); } --> The result is: <script>: ======= with $avp(...)================ <script>: +499539110 matches +499539[2-5]: NO <script>: +499539110 matches +499539*: YES <script>: +499539110 matches +499539[2-5]: YES <script>: ======= with $var(...)================ <script>: +499539110 matches +499539[2-5]: NO <script>: +499539110 matches +499539*: YES <script>: +499539110 matches +499539[2-5]: NO Thanks Klaus

1 1

Where can I get kamctl.oracle
by Konstantin Khatskevich 08 Jul '14

08 Jul '14

Hello! I attempt to dispose the bond ODBC-Oracle, but I cannot find kamctl.oracle (I have any other)& Where can I download this kamctl.oracle? -- Best regards, Konstantin Khatskevich

2 1

Re-Compiling and Re-Installing kamailio server
by aawaise 08 Jul '14

08 Jul '14

I am trying to make changes in code and recompile kamailio server project. Problem is when I try the following command make PREFIX='/usr/local/kamailio-3.1/' FLAVOUR=kamailio.cfg I get the message; 'kamailio' is up to date. I want to know, how to make my machine forget about the older installation, so that new compilation goes good. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Re-Compiling-and-Re-Installing-kama… Sent from the Users mailing list archive at Nabble.com.

2 1

Help me to configure Kamailio on EC2, not able to connect through Jitsi
by Jayaraman, Kamalakannan 08 Jul '14

08 Jul '14

Hi, I had installed kamailio on my AWS EC2, by following the steps in http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x . Since I installed in EC2, i changed the listen=<my-private-ip> advertise=<my-public-ip> in kamailio.cfg file too. I open the 5060 port in AWS console too. Kamailio starts without any error, but I can't able to connect in Jitsi client. Please help me out to do this as soon as possible. -- *Kamalakannan Jayaraman* *Associate Software Engineer* Pearson English kamalkannan.jayaraman(a)pearson.com +91 8754 7754 53 - Mobile kamalakannan.jayaraman.ge - Skype 6th Foor, 10th Block, DLF IT park, Chennai - 600089 India Pearson English is the world's largest provider of English language learning.

1 0

need help on kamailio-snmpstats module
by Allen Zhang 07 Jul '14

07 Jul '14

Hi, I'm having trouble on installing kamailio-snmpstats module. I'm using kamailio 4.0. I want to install kamailio's snmpstats module from RPM. I compiled net-snmp 5.4 from source and built some RPMS from it. I've installed net-snmp 5.4 from RPM on the target machine. And when I run 'yum install kamailio-snmpstats', I get: Error: Package: kamailio-snmpstats-4.0.0-0.el6.x86_64 () Requires: net-snmp-agent-libs Where do I get this net-snmp-agent-libs? Can I buid this from net-snmp source somehow? Regards, Allen

2 2

Measuring Response Time for a SIP Message
by AliReza Khoshgoftar Monfared 07 Jul '14

07 Jul '14

Hi, I am trying to do two types of measurements on a packet (say an INVITE or BYE) that arrives at a kamailio proxy. 1) The time that packet is spending in Kamailio to be processed 2) The total time from when the packet arrives at the machine that runs the proxy till it leaves that machine. I believe kamailio shall be able to help me with (1) and for (2) , I may need to use some linux tools Have anybody tried to measure response time with either definition? and in case that you have, how have you done it? Thanks Alireza

2 1

$avp(received)
by Kelvin Chua 07 Jul '14

07 Jul '14

is there any way of getting the received ip/port/proto from nathelper? modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") set_contact_alias(); xlog("$avp(RECEIVED)"); this gives me <null> Kelvin Chua

2 3

kamailio server over internet
by Gaurav Chaudhary 07 Jul '14

07 Jul '14

sir , i have set-up a kamailio server on a machine , every thing is in default configuration. but the issue is only local clients are able to register. i have forwarded the port 5060 correctly but the clients from some other network over the internet are not able register every request just got time out at the client side. do i nee to enable some specific setting to make it happen.

2 1

Extract value from a function
by aawaise 07 Jul '14

07 Jul '14

I came across this function of registrar module reg_fetch_contacts (domain, uri, profile) How can I log domain of a user using this function ?? Need help in syntax. Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Extract-value-from-a-function-tp128… Sent from the Users mailing list archive at Nabble.com.

2 1

Re: [SR-Users] UAC remote registration - refreshing database
by Dan Christian Bogos 07 Jul '14

07 Jul '14

Hey Daniel, I totally agree on improving versus adding new module. The only thing which made me going towards porting was not knowing how easy would be enhancing vs porting something which is implemented. Also the fact that Alexa has mentioned it as an often requested feature made me think that it could take a while until someone would be able to do it in the existing module. Anyway, thanks for your input! DanB PS: Ovidiu, I appreciate your feedback as well. Good to know that you are around ;).

2 1

MSILO - dumped messages get duplicated
by Marc M. 06 Jul '14

06 Jul '14

Hi, I have a Kamailio 3.2 setup up and running. Recently I configured msilo for offline message store. Essentialy the setup runs fine. Messages to users without location info getting stored. In case the user has location info, but cannot be reached, I made failover_route, so the messages getting stored in this case too. Messages getting dumped at the next REGISTER. However, I have a group of new mobile clients, with unstable connection. It happens many times that they send a REGISTER, but the connection brakes afterwards. It this case the message is getting dumped, but since it cannot be delivered, it runs on the failover_route which stores the message again. However, the previous message is not removed from the db, so the same message is now twice in the db. Next time the message either gets delivered in double, or in case of a new failure, it gets stored 2x2= four times. I need to avoid this duplicates but still make sure the message get delivered sooner or later. Any help is highly appreciated. Here is my routing table: thanks Marc ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } if (is_method("MESSAGE")) { log(1,"-------------------------------------------------------->WE HAVE A NEW MESSAGE\n"); route(MSILO_MESSAGE); } t_check_trans(); # authentication 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")) { 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); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } ##storing offline messages route[MSILO_MESSAGE]{ if (lookup("location")){ t_on_failure("FAIL_MESSAGE"); t_relay(); xlog("L_NOTICE","---------------------------------MI: ($mi), MB: ($mb)\n"); }else{ xlog("L_NOTICE","------------------MESSAGE PROCESSING: location not found, storing offline message ID: ($ru)\n"); m_store("$ru"); sl_send_reply("202", "ACCEPTED"); xlog("L_NOTICE","------------------MESSAGE PROCESSING: location not found, stored offline message ID: ($ru)\n"); }; exit; }; failure_route[FAIL_MESSAGE]{ xlog("L_NOTICE","------------------MESSAGE PROCESSING: user cannot be reached, storing offline message ID: ($ou)\n"); m_store("$ou"); t_reply("202", "ACCEPTED"); xlog("L_NOTICE","------------------MESSAGE PROCESSING: user cannot be reached, stored offline message ID: ($ou)\n"); 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 $fu (IP:$si:$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()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard 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(); # MSILO - dumping user's offline messages m_dump(); log(1,"-------------------------MESSAGE PROCESSING: dumping offline messages\n"); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!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 if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rtpproxy_offer(); } if (!has_totag()) add_rr_param(";nat=yes"); #!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; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { rtpproxy_answer(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }

3 4

RTP to SRTP encryption with Kamailio.
by Dani Kamailio 04 Jul '14

04 Jul '14

Hello, I need to build a VoIP system who receives SIP and RTP traffic in a public IP and encrypt both of them with TLS and SRTP respectively. The main point is to have security inside of the local network (I know it may sound unuseful). So, I was trying to build the whole system in Kamailio but I got stucked with the RTP to SRTP bridge and I do not really know how to do it. I know that there are some modules like rtpproxy-ng and rtpengine as media relay that can handle SRTP... any ideas? I just want to make sure that I am in the right way. If it could be done in Kamailio, have I to write the code to encrypt RTP, like the algorithm or something? Would it be convenient to send the RTP packets to Asterisk? (and Asterisk would somehow encrypt them?) I am pretty new with all of this, any help will be apreciate, thank you.

2 2

no rtpengine.so module
by Yuriy Gorlichenko 04 Jul '14

04 Jul '14

Hello. I install kamailio 4.1.3 and it works fine. But I neen proxy DTLS SRTP to backend media servers through kamailio. rtpproxy no not work with dtls srtp? so I installed rtpengine but can not find rtpengine.so file for copiyng to kamailio modules directory. Does anyone can install rtpengine as kamailio module?

3 4

FW: Module TOPOH: Does not unmask CALL-ID on request BYE in one direction
by Samuel F. 04 Jul '14

04 Jul '14

Hi, We are setting up a Kamailio Proxy with the Topoh module in place masking everything including CALL-ID. It seems to be masking everything just fine except for CALL-ID in BYE requests (verified with PCAPs) when received from downstream. Upstream works fine. Looking at the source code for the module 4.1.2 we see the unmasking being done like this: if(dialog) { /* dialog request */ th_unmask_ruri(&msg); th_unmask_route(&msg); th_unmask_refer_to(&msg); if(direction==1) { th_unmask_callid(&msg); } } It only seems to be unmasking for one direction. From my understanding BYE can be initated from both directions? Might be something I don’t understand here how the source code works but I wanted to ask for any pointers of help here.. been troubleshooting this for a good while now. Kind regards, Samuel

3 2

Re: [SR-Users] UAC remote registration - refreshing database
by Dan Christian Bogos 04 Jul '14

04 Jul '14

Hey Alex, Many thanks for so fast answer. Could be I have missed the past interest (hence re-posting). Anyway, I wonder what would be the chances that we get Ovidiu's interest so he can port registrant module to kamailio maybe? Unfortunately without reloads it is hard to push remote registration in enterprise solutions. Thanks again, DanB On 03.07.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > On 07/02/2014 06:22 AM, Dan Christian Bogos wrote: >> >Anybody aware if it is possible to refresh the list of remote >> >registrations from the database without restarting the whole server? > This gets asked a lot, and the answer is no. But it is probably a widely > desired feature set by now. > > -- 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 2

Calls per second
by Olle E. Johansson 03 Jul '14

03 Jul '14

Hi! Have you implemented a per-customer rate limit in Calls per second? If so - how? I've played with ratelimit/pipelimit and it seems like I can define a database with one pipe per customer - but have to restart Kamailio to add customers. There are warnings for low timer settings, like 1 second, but I don't know how up-to-date those warnings are. I guess I could play with hash tables and implement something pike-like there, but it seems like a workaround for something pretty common. So the question remains - how are you limiting on a cps per customer? /O

6 15

sipcapture / raw_interface
by András Stefler 03 Jul '14

03 Jul '14

Hello all, I am new in using Kamailio. I have installed the version 4.0.1 and using it as SIP capture node. My server has lots of physical interfaces where I want to enforce capturing on one of the only. This is called 'csm1'. This is the corresponding part from kamailio.cfg modparam("sipcapture", "capture_node", "sipserv") modparam("sipcapture", "db_url", "mysql://****) modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 0) #modparam("sipcapture", "raw_socket_listen", "192.168.0.1:5060") modparam("sipcapture", "raw_interface", "csm1") modparam("sipcapture", "raw_ipip_capture_on", 0) modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "raw_sock_children", 4) modparam("sipcapture", "raw_moni_capture_on", 1) modparam("sipcapture", "promiscious_on", 1) modparam("sipcapture", "raw_moni_bpf_on", 1) ******************************** stefler@sipserv:~$ ifconfig csm1 csm1 Link encap:Ethernet HWaddr ac:16:2d:96:3f:e0 inet addr:1.0.8.1 Bcast:1.0.8.3 Mask:255.255.255.252 inet6 addr: fe80::ae16:2dff:fe96:3fe0/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 My problem is that capturing is activated on all of the interfaces and only on the selected one. Have you seen this problem before? Regards Andras Stefler

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How to check message queue (buffer) in Kamailio
by AliReza Khoshgoftar Monfared 03 Jul '14

03 Jul '14

Hi, I had another simple question: In a kamailio server (proxy), how do I check the number of messages currently waiting for processing? Is there a variable that I can monitor, say, if I want to make a routing decision in my config based on the number of messages in the queue? Also, is it possible to get a head count by method? or is it only possible after fully parsing the message? I see that "ratelimit" module uses similar information, but I am not sure how to get the status of these queues that the module uses. Thanks

3 5

Server generated 408 time out goes to onreply route?
by Allen Zhang 03 Jul '14

03 Jul '14

Hi, Does local server generated 408 time out reply go to onreply_route? If so, can I change the status code to a 200 class code by doing: $rs = "202"? If I can change it, does it still go to failure_route? (It's not a 400 class response anymore.) The scenario I want to test it's easy to produce. Otherwise I would just try it...... Regards, Allen

2 1

Replys though onsend_route?
by Jöran Vinzens 03 Jul '14

03 Jul '14

Hi all, is there a possibility to pass reply and failure though onsend_route as well? We try to capture via siptrace the incoming and outgoing packet. So far this is only possible with Requests. I've found this: http://lists.sip-router.org/pipermail/sr-users/2012-May/073224.html Can anybody confirm this is still the behavior? Does anybody have a solution passing reply though onsend_route or any other way to capture the outgoing reply via siptrace? Thanks a lot for your help. BR Jöran Vinzens -- Jöran Vinzens - vinzens[at]sipgate.de <vinzens(a)sipgate.de> sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk

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how to remove non-digits using dp_translate
by Henry Fernandes 03 Jul '14

03 Jul '14

I'd like to use the dialplan module to remove all non-digits from a string. For example, "1-306-555-1234" becomes "13065551234" and "(306) 555-1234" becomes "3065551234". Unfortunately, I can't figure out the regular expressions for subst_exp and repl_exp. I can easily replace one instance of a non-digit, however I can't figure out how to replace all non-digits. -H

2 1

Making RLS presence work with Blink and Kamailio 4.1.4
by Alex Villacís Lasso 02 Jul '14

02 Jul '14

I am having trouble making all of the supposed features of Blink work with Kamailio 4.1.4. My kamailio.cfg file is attached. Specifically, what I am having trouble is with presence (the way Blink wants to implement it), and MSRP. Ordinary voice calls work correctly. With presence, I have managed to make XCAP storing and retrieval work, in the sense that a document that is being stored with PUT can be later retrieved with GET on the same URL. However, I cannot make Blink use this support in order to actually monitor presence. As far as I understand, when user A subscribes to the presence state of user B, and user B is registered, user B is supposed to receive a notification that user A requests authorization for presence subscription. However, this is not working so far when both A and B use Blink. From what I can see through a packet capture, neither adding a new contact on the Blink window or having existing contacts result in any SUBSCRIBE packets for the contact users being sent to the Kamailio instance. If the subscription is meant to be performed some other way, I am not seeing it being done either. Of course, this results in presence changes not being exchanged between contacts. The only SUBSCRIBE packet I see from Blink is the one for the SIP account used for Blink. With MSRP, I have copied the example from the msrp.so documentation, but I have replaced the authentication shown with a static password, with an actual query using auth_db.so, the very same query used to authenticate other SIP packets. Through the packet capture, when user A tries starting a MSRP chat with user B, an INVITE is sent from user A to Kamailio, which has a special check to NOT send this invite to Asterisk, but rather route it itself, and it gets "correctly" routed to user B. Then user B attempts to start a MSRP session with Kamailio (not user A) without authentication credentials, Kamailio challenges user B for the credentials, and... that's it. User B makes no attempt to repeat the MSRP session with the required credentials, and the communication eventually times out. The presence failure might have something to do with the following exchange I see in the packet capture: GET /xcap-root/pidf-manipulation/users/sip:avillacisIM@pbx.elastix.com/index HTTP/1.1 Accept-Encoding: identity Authorization: Digest username="avillacisIM", realm="pbx.elastix.com", nonce="U6xFz1OsRKNuU11dgljLCgGppv5lK6Eu", uri="/xcap-root/pidf-manipulation/users/sip:avillacisIM@pbx.elastix.com/index", response="3dfdf556c9812f4bebb8b0cc82578ef9", algorithm="MD5" Host: pbx.elastix.com:5060 Connection: close Accept: application/pidf+xml User-Agent: Blink 0.8.2 (Linux) HTTP/1.1 404 Not found Via: SIP/2.0/TCP 192.168.3.2:55567 Server: kamailio (4.1.4 (x86_64/linux)) Content-Length: 0 That is, the Blink client tries to fetch the pidf-manipulation document, and Kamailio answers with a 404. In other cases Blink has responded with a PUT of a minimal document at the required URI (which gets stored to the database), but in this case (pidf-manipulation) this is not done. I am not sure of what is supposed to happen here. This 404 is the same regardless of whether there are other (Blink or otherwise) SIP clients connected. I think this has been asked before, but is there a working implementation of MSRP+presence for Kamailio that works with Blink? Or failing that, a complete kamailio.cfg that has been shown to work with at least one MSRP implementation?

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UAC remote registration - refreshing database
by Dan Christian Bogos 02 Jul '14

02 Jul '14

Hey Guys, Anybody aware if it is possible to refresh the list of remote registrations from the database without restarting the whole server? Ta, DanB

2 1

TCP rest by peer
by Rob Moore 02 Jul '14

02 Jul '14

Hi All, Since we've been using sipPhones as a proxy for registrations and calls to phones over TCP. I've been seeing a couple of these crop up in the logs. /usr/sbin/kamailio[4255]: ERROR: <core> [tcp_read.c:1131]: ERROR: tcp_read_req: error reading /usr/sbin/kamailio[4255]: ERROR: <core> [tcp_read.c:293]: error reading: Connection reset by peer (104) We haven't had any complaints of problems with the phones so if these are not a problem I'd like to know how to clean them up. Any suggestions on cause or solution would be fantastic. Thanks in advance. Rob Kind regards, Robert Moore Telephony System Architect [Description: Aeriandi] m. +44 (0)7766 838 040 t. +44 (0)845 108 0308 [cid:image002.png@01CF8C82.823C49E0]<http://www.retailweektechnologyawards.com/winners-2014> Aeriandi Ltd / Prama House, 267 Banbury Road, Oxford OX2 7HT / +44 (0)845 108 0308 / info(a)aeriandi.com<mailto:info@aeriandi.com> / aeriandi.com<http://www.aeriandi.com/> This transmission is intended solely for the addressee and may be confidential. If you are not the named addressee, or if the message has been addressed to you in error, you must not read, disclose, reproduce, distribute or use this transmission. Please delete the message or contact the sender. Delivery of this message to any person other than the named addressee is not intended in any way to waive confidentiality. Please note that neither Aeriandi Ltd nor the sender accepts any responsibility for any viruses, which may be transmitted, and it is your responsibility to scan attachments (if any).

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msilo possible bug
by Kelvin Chua 02 Jul '14

02 Jul '14

Hi guys, i think i have stumbled on an msilo bug. m_dump() is not working. kamctl fifo get_statistics all | grep mess msilo:dumped_messages = 0 msilo:failed_messages = 0 msilo:stored_messages = 1 so here we can see there is a message to be delivered. i turned on debug mode, then the intended destination of message logs in. then this is what i see on logs DEBUG: msilo [msilo.c:1082]: m_dump(): dumping [1] messages for <sip:XX85X80X74@1X8.1X9.1X7.1X3>!!! DEBUG: msilo [ms_msg_list.c:169]: msg_list_check_msg(): checking msgid=405 DEBUG: msilo [ms_msg_list.c:207]: msg_list_check_msg(): msg already in sent list. DEBUG: msilo [msilo.c:1089]: m_dump(): message[0] mid=405 already sent. using ngrep, no message was ever seen on the wire. then, kamctl fifo get_statistics all | grep mess msilo:dumped_messages = 0 msilo:failed_messages = 0 msilo:stored_messages = 1 still the same. anybody experienced this? Kelvin Chua

2 2

Fedora 18-20 GeoIP missing?
by Roberto Fichera 02 Jul '14

02 Jul '14

Hi All, is there any reason why the GeoIP module is missing in the Fedora repo? Thanks in advance. Roberto Fichera.

2 2

Possible GPL violation by third party (4PSA)
by Matthew Costa 02 Jul '14

02 Jul '14

4PSA develop and sell a software suite called VoipNow Professional (version 2.5) which, amongst other things, includes it's own packaged copy of Kamailio. This version is derived from Kamailio version 1.5.4. The software installs a binary release from a 4PSA repository. The RPM information contains the following: # rpm -qi voipnow-kamailio-1.5.4-130312.41.rhel5 Name : voipnow-kamailio Relocations: (not relocatable) Version : 1.5.4 Vendor: Rack-Soft, Inc <devel(a)4psa.com> Release : 130312.41.rhel5 Build Date: Tue 12 Mar 2013 18:31:30 EST Install Date: Sun 02 Feb 2014 00:15:14 EST Build Host: rhel5-1.64b.build.4psa.ro Group : System Environment/Daemons Source RPM: voipnow-kamailio-1.5.4-130312.41.rhel5.src.rpm Size : 6599403 License: GPL Signature : RSA/SHA1, Tue 12 Mar 2013 18:35:28 EST, Key ID 42ba8c472f75de11 Packager : Rack-Soft, Inc <devel(a)4psa.com> URL : http://kamailio.org/ Summary : Kamailio, a very fast and flexible SIP Proxy I've made a direct request to 4PSA for the source code for this package, only to be told they don't release their sources. The request, and subsequent response, can be found at http://my.4psa.com/4psa/topics/where_can_i_find_the_source_code_for_the_voi pnow_2_5_x_modified_version_of_kamailio. For those interested, the following packages are included with VoipNow which are derived from GPL software. There is no source available for any of these from the vendor. # rpm -qa |grep voipnow voipnow-spandsp-0.0.6-100708.10.rhel5 voipnow-asterisk-1.6.1.20-130312.07.rhel5 voipnow-asterisk-addons-1.6.1-130131.30.rhel5 voipnow-php-2.5.5-130312.11.rhel5 voipnow-sox-14.3.1-100723.30.rhel5 voipnow-asterisk-sounds-1.4.22-100723.25.rhel5 voipnow-kamailio-1.5.4-130312.41.rhel5 voipnow-ejabberd-2.1.4-100723.29.rhel5 voipnow-asterisk-debuginfo-1.6.1.20-120417.09.rhel5 voipnow-asterisk-extra-2.5.5-130312.11.rhel5 My requests are falling on deaf ears. Does anybody know how to make them take this matter seriously? Cheers, Matthew Costa Infrastructure Manager <http://www.greenlight-itc.com/> Suite 703, Level 7 815 Pacific Highway Chatswood 2067 tel: 02 8412 0000 fax: 02 8412 0001

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Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls
by Carlos Rangel 01 Jul '14

01 Jul '14

Hello List Hopefully someone can help. This is the problem when the call is hug up 20-30 seconds after it initiates. The call is only hung on when the remote extension initiates the call. If the remote extension receives the call there is no problem the call is not hung on. I changed the remote cisco phone for a yealink and it is the same behavior. It thought it was the phone. This is what I am using in kamailio.cfg #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ASTERISK #!define WITH_USRLOCDB #!define WITH_ANTIFLOOD Remote User Internet Internal network Yealink IP TG28P ----DSL router ---|------Internet --------|-----Cisco ASA 5500 FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production Server --------|------ PSTN Thanks Carlos Rangel De: Carlos Rangel [mailto:crangel@globaltelesourcing.com] Enviado el: jueves, 26 de junio de 2014 01:27 p.m. Para: miconda(a)gmail.com; 'Kamailio (SER) - Users Mailing List' Asunto: RE: [SR-Users] Kamailio Freepbx Integration Dropping Calls Hi Daniel Thank you so much for your response. Here is the SIP trace of one of the calls, I am not sure where the call initiates but you can see at the end of the file in bold X-Asterisk-HangupCause: No user responding. I am not sure why is it sending this message though. The variables are Extension/Username=XXXXX Ext_IP= Public IP Internal_IP= Asterisk/Kamailio internal IP Sorry for the long file but again I am not sure where the call initiates This is the part where that call is hung on. U 2014/06/26 13:36:11.831965 Kamailio_IP:5080 -> Kamailio_IP:5060 BYE sip:XXXXX@65.190.71.203:5060;user=phone;transport=udp SIP/2.0. Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47. Route: <sip:Kamailio_IP;lr=on;ftag=000653dc394000970f227678-1fafb4e2>. Max-Forwards: 70. From: <sip:919707249077@Kamailio_IP:5060>;tag=as2e670ea4. To: "User" <sip:XXXXX@Kamailio_IP:5060>;tag=000653dc394000970f227678-1fafb4e2. Call-ID: 000653dc-3940000b-33caf1b2-20ccd185(a)192.168.0.22. CSeq: 102 BYE. User-Agent: FPBX-2.11.0(11.10.2). X-Asterisk-HangupCause: No user responding. X-Asterisk-HangupCauseCode: 18. Content-Length: 0. . U 2014/06/26 13:36:11.832260 Kamailio_IP:5060 -> 65.190.71.203:5060 BYE sip:XXXXX@65.190.71.203:5060;user=phone;transport=udp SIP/2.0. Via: SIP/2.0/UDP Kamailio_IP;branch=z9hG4bKcf68.d6ef5aa9cc5bd0fb0ab13a563b7cf284.0. Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47. Max-Forwards: 69. From: <sip:919707249077@Kamailio_IP:5060>;tag=as2e670ea4. To: "User" <sip:XXXXX@Kamailio_IP:5060>;tag=000653dc394000970f227678-1fafb4e2. Call-ID: 000653dc-3940000b-33caf1b2-20ccd185(a)192.168.0.22. CSeq: 102 BYE. User-Agent: FPBX-2.11.0(11.10.2). X-Asterisk-HangupCause: No user responding. X-Asterisk-HangupCauseCode: 18. Content-Length: 0. Descripción: Description: Description: DLR-Logo-No-TextCARLOS RANGEL | INFORMATION TECHNOLOGY DIRECTOR Global Telesourcing México, S. de R.L. de C.V. | Aarón Sáenz #1891-1 | Monterrey, N.L., México Direct 703 894 1667 | Mobile US 703 894 1667 | Mobile MX +52 1 812 000 7362 | crangel(a)globaltelesourcing.com ________________________________________________________________________________________________________________ The information contained in this e-mail and any attached documents may contain information that is confidential or otherwise protected from disclosure. If you are not the intended recipient of this message, or if this message has been sent to you in error, please immediately alert the sender by reply e-mail and then delete this message, including any attachments. Any dissemination, distribution or other use of the contents of this message by anyone other than the intended recipient is strictly prohibited. De: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] En nombre de Daniel-Constantin Mierla Enviado el: jueves, 26 de junio de 2014 03:12 a.m. Para: Kamailio (SER) - Users Mailing List Asunto: Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls Hello, can you gran the SIP trace on kamailio server for such case? You can use ngrep, like: ngrep -d any -qt -W byline port 5060 and send the output to the mailing list. You can replace any sensitive information (e.g., ip address) before sending to mailing list. The typical call drop after 30-40 secs is when ACK is not routed properly, but we have to see that in the sip trace. Cheers, Daniel On 25/06/14 18:50, Carlos Rangel wrote: Hello I have successfully (I believe) implemented Kamailio 4.1.4 integration with Freepbx 5.2.11 taking as a guide Daniel’s tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. I just did not create the voicemail tables because voice mail is handled by Freepbx. I installed the system in a separate box for testing and connected to the Freepbx Production server via IAX trunk. The system is behind a Cisco Firewall and looks like this Remote User Internet Internal network Cisco 7960 ----DSL router ---|------Internet --------|-----Cisco ASA 5500 FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production Server --------|------ PSTN I have configured the FW to allow UDP and TCP traffic from the corresponding IP as well as tfpt that is needed for the Ciscos to pick up the configuration from the server. I have a few remotes Cisco 7960 phones that can register remotely in Kamailio as long as the user is added with kamctl add user password and as long as the extension is created in Freepbx. The problem that I have is when try to make a call from the remote Ciscos the call is dropped after 30 or 40 seconds. I can see from the logs that the problem appears to be that the server is not receiving responses from the phone 06-25 10:57:30] WARNING[1814] chan_sip.c: Retransmission timeout reached on transmission 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32001ms with no response [2014-06-25 10:57:30] WARNING[1814] chan_sip.c: Hanging up call 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Is this something that we can adjust in kamailio or could it be related to the FW configuration?? Sorry but I am very new to kamailio and sip. Thanks Carlos _______________________________________________ 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

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installation siremis error
by Hossein Aghaie 01 Jul '14

01 Jul '14

hi after install kamailio 4.1.0 on ubuntu server,I try to install siremis-4.1.0 of this Doc : http://kb.asipto.com/siremis:install41x:main but to end setup,show this error : http://paste.fedoraproject.org/114029/ how to solved this problem? tnx

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Kamailio Development Workshop, July 9-10, 2014, Paris, France
by Daniel-Constantin Mierla 01 Jul '14

01 Jul '14

Hello, I want to announce the next Kamailio Development Workshop, to take place in Paris, France, during July 9-10, 2014. This event is the next in its series targeting to show the internals of Kamailio and enable more people to become developers as well as let users of Kamailio to get more knowledge about the design of the application which can have relevant impact in operating deployment. Be aware that it is not a workshop about Kamailio installation and administration. Its content is about writing C code to extend Kamailio. More details can be found at: - http://www.kamailio.org/w/2014/06/kamailio-development-workshop-july-9-10-2… Similar to the past events, we are considering to have a social networking event in the evening of July 9 (most probably a dinner meetup at a nice place in Paris), which is open for everyone, each participant taking care of own expenses. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 2

how to compile module db_oracle on Centos 6.5
by Konstantin Khatskevich 01 Jul '14

01 Jul '14

Hello! 5. Installation Because it dependes on an external library, the oracle module is not compiled and installed by default. You can use one of the next options. * - edit the "Makefile" and remove "db_oracle" from "excluded_modules" list. Then follow the standard procedure to install Kamailio: "make all; make install". * - from command line use: 'make all include_modules="db_oracle"; make install include_modules="db_oracle"'. But "db_oracle: requires non-free Oracle client SDK" (http://lists.sip-router.org/pipermail/sr-dev/2013-October/021803.html) What can I do? -- Best regards, Konstantin Khatskevich

2 2

Changes in registrar module source code
by aawaise 01 Jul '14

01 Jul '14

Hello, I am a user of kamailio server. I am in need to pick domain of particular online users from location list. I found out that usrloc module is used for Location Table items. And we cant directly export parameters from usrloc module. I decided to use registrar module to approach domain parameter of usrloc module. I want to understand complete flow control in registrar module source code, so that I can add my commands to pick domain parameter from usrloc module. How can it be done ? Secondly, if I make a change in registrar module, to see its reaction, I will have to recompile kamailio server or is there any shortcut to immediately see effect of changes made in registrar module ?? Any help will be appreciated. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Changes-in-registrar-module-source-… Sent from the Users mailing list archive at Nabble.com.

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Siremis 4.1.0 menu adjustment for different users
by Fred 01 Jul '14

01 Jul '14

Hi, I'm setting up a Registrar server using Kamailio 4.1.3 with Siremis 4.1.0 as WebGUI. On Siremis I struggle to set up different users which only see limited menu items. What I want to do is e.g. an user called "CustomerService" which only sees the "Subscriber_List" menu item and a second user called "SystemAdmin" which only sees the "Dispatcher_List" menu item. I tried different things with new roles, groups and adjustments in the menu administration but didn't manage to get the desired result. Does anyone have a hint for me if that is possible and how? Thanks Fred

1 0

Child Processes Stopped kamailio 4.0.6
by Jonathan Hunter 30 Jun '14

30 Jun '14

Hi Guys, Sorry thought I better raise this in new stream to avoid confusion. We have another crash today, not sure if the below is of much use without debug? Thanks Jon Jun 26 09:38:09 pbx /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:788]: handle_sigs(): child process 18532 exited by a signal 11Jun 26 09:38:09 pbx /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:791]: handle_sigs(): core was not generatedJun 26 09:38:09 pbx /usr/sbin/kamailio[18529]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLDJun 26 09:38:09 pbx /usr/sbin/kamailio[18531]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:788]: handle_sigs(): child process 18532 exited by a signal 11Jun 26 09:38:09 pbx9 /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:791]: handle_sigs(): core was not generatedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18529]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLDJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18531]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18533]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18537]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18534]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18535]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18538]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18530]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18539]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18536]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 kamailio[2076]: ERROR: <script>: Failed Reg Block Traffic from 194.72.34.18Jun 26 09:38:09 pbx kamailio[2075]: ERROR: <script>: Failed Reg Block Traffic from 194.72.34.18Jun 26 09:38:09 pbx kamailio[2074]: ERROR: <script>: Routing Call To Asterisk serverJun 26 09:38:09 pbx kamailio[2074]: INFO: <script>: Routing Call to Dispatcher group 1 -Asterisk Media ServersJun 26 09:38:10 pbx kamailio[2072]: ALERT: <core> [main.c:788]: handle_sigs(): child process 2075 exited by a signal 11Jun 26 09:38:10 pbx kamailio[2072]: ALERT: <core> [main.c:791]: handle_sigs(): core was generatedJun 26 09:38:10 pbx kamailio[2072]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLDJun 26 09:38:10 pbx kamailio[2074]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2078]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2076]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2077]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2079]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2073]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2080]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2081]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2082]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2072]: ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: No such file or directory (2)Jun 26 09:43:33 pbx kamailio: INFO: <core> [tcp_main.c:4846]: init_tcp(): init_tcp: using epoll_lt as the io watch method (auto detected)

2 2

How to extract information from location table
by aawaise 30 Jun '14

30 Jun '14

I want to know is there any indirect method to extract information from location table that is kept by Usrloc module ?? I want to know about the username and domain name of users from Location Table on arrival of INVITE packet. How can it be done ? -- View this message in context: http://sip-router.1086192.n5.nabble.com/How-to-extract-information-from-loc… Sent from the Users mailing list archive at Nabble.com.

2 1

route [Location] Documentation
by aawaise 30 Jun '14

30 Jun '14

I want to know that how route [Location] works when an INVITE packet approaches kamailio server. Basically route [Location] gathers information whether callee is a valid user or not. Secondly if it is online or not. I want to understand this whole procedure. Any help will be appreciated. Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/route-Location-Documentation-tp1286… Sent from the Users mailing list archive at Nabble.com.

2 1

Voice quality issues when registering to Kamailio
by Travis Dillon 30 Jun '14

30 Jun '14

Hello all, I have an environment setup as follows: VoIP routers -> Kamailio 4.0 -> Asterisk 11.6 When I register the phones to Asterisk the call quality is excellent. So to be clear, the call comes in the VoIP router, routes through Kamailio to the phone registered to Asterisk. When I register the phone to Kamailio the call quality is significantly reduced. So clarity here, the call comes in the VoIP router, routes through Kamailio to the phone registered to Kamailio. Why would the call quality diminish when registered to Kamailio? Where can I start looking for this issue? Thank you, Travis R. Dillon

2 1

UAC module refresh registration, could not reuse callid/fromtag !!!
by nevious 30 Jun '14

30 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?

4 4

Strange application hang querying DB
by Tim Chubb 27 Jun '14

27 Jun '14

Hi, I'm currently wracking my brains trying to deduce where an application hang can be coming from. I have ported the PostGres schema to be compatible with MS Sql Server and have had no issues getting a working dev environment working. However when I try and use SQL Azure as the backend db as opposed to SQL Server 2012 (running locally on my dev machine) Kamailio will seem to hang after about 10 - 15 mins of flawless operation. I have recreated the SQL Azure schema from the working schema running on my dev machine and copied all data in a 1:1 manner so I can be certain it's not a data quality issue. I am using the same FreeTDS/UnixODBC config that I am using for Asterisk realtime (which doesn't exhibit this behaviour). The really strange thing is that I can see the queries being sent by Kmailio in the freetds trace log, and the response from SQL Azure, however its as if the db response handler is deadlocked/hung. The kamailio process continues to work fine, and as long as its not required to commiunicate with the DB all appears well i.e. no error messages in the log file. This behaviour only happens when im using SQL Azure as opposed to SQL Server, so there must be something causing the issue, that's unique to SQL azure as opposed to SQL Server. As far as I can tell this is not something I can create a backtrace for as no core dumps are produced, as far as I can tell there is no easy way debug what the lock is. If anyone has any suggestions I would be very grateful to hear them. One thing that occurs to me is that the db_unixodbc module doesn't have half as many parameters as its equilent in Asterisk, and that connection pooling may be in use, this could be a bit of a show stopper AFAIK you have to disable connection pooling with SQL server, certainly any of the guides I have come across configuring Asterisk to work with SQL server suggests that. Tim Chubb Developer [VS-logo-email] The Coach House, Heywood House, Park Lane, Heywood, Westbury, BA13 4NA Direct: +44 (0)333 0110164 Email: tim.chubb(a)voicesimplified.com<mailto:tim.chubb@voicesimplified.com> Web: www.VoiceSimplified.com<http://www.voicesimplified.com/> Voice Simplified Ltd (registered in England & Wales: 07171825) registered office: Curzon House, 2nd Floor, 24 High Street, Banstead, Surrey. SM7 2LJ. The information in this email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. If you are not the intended recipient, please return the message to the sender and delete it from your records.

2 3

Error when install Kamailio 4.1.4 on Mac OS 10.9.
by Son Truong 27 Jun '14

27 Jun '14

Hi Kamailio, - I meet an error when I am trying install Kamailio 4.1.4 on Mac OS 10.9. The error: touch /usr/local/lib64/kamailio/modules/usrloc.so install -m 755 usrloc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp.so install -m 755 xhttp.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp_rpc.so install -m 755 xhttp_rpc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xlog.so install -m 755 xlog.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xprint.so install -m 755 xprint.so /usr/local/lib64/kamailio/modules /bin/sh: -c: line 1: syntax error: unexpected end of file make: *** [install-cfg] Error 2 How can I fix it? I am new in Kamailio, I want to learn Kamailio. -- Thanks and best regards, Phạm Sơn Trường Android, iOS Team Lead at ME Corp. Email1: truonguit2010(a)gmail.com Email2: truongps(a)mecorp.vn Mobile: 01218816208

1 0

Processing of Invite Packet
by aawaise 27 Jun '14

27 Jun '14

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

1 0

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

26 Jun '14

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

1 0

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

26 Jun '14

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

2 2

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

26 Jun '14

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

2 2

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

26 Jun '14

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

3 4

Trouble with rtpproxy
by Igor Potjevlesch 26 Jun '14

26 Jun '14

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

2 3

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

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