- sr-users - kamailio.org

log file
by Uzma Anjum 16 Aug '17

16 Aug '17

the log statements written in acc.c accounting module source code are not entered into syslog file.what can be wrong?

1 0

functions in accounting module
by Uzma Anjum 16 Aug '17

16 Aug '17

when the call answered,which methods are executed from acc.c source code and which method is inserted into database while call answer and call terminate.

1 0

Kamailio not listening on TLS port (5061)
by Deepak Jaiswal 16 Aug '17

16 Aug '17

Hi, I am new to kamailio and unable to debug the issue. Kamailio is not listening on TLS port. Could you please suggest some workaround? [root@uk-vap-002 ~]# ss -tnlp | grep kamailio LISTEN 0 1024 127.0.0.1:5060 *:* users:(("kamailio",29400,18)) LISTEN 0 1024 xxx.xxx.xxx.134:5060 *:* users:(("kamailio",29400,17)) LISTEN 0 1024 xxx.xxx.xxx.135:5060 *:* users:(("kamailio",29400,16)) LISTEN 0 1024 xxx.xxx.xxx:5060 *:* users:(("kamailio",29400,15)) Kamailio.cfg config done: listen=tls:xxx.xxx.xxx.133:5061 listen=tls:xxx.xxx.xxx.135:5061 listen=tls:xxx.xxx.xxx.134:5061 listen=tls:127.0.0.1:5061 2. #!ifdef WITH_TLS enable_tls=yes 3. #!ifdef WITH_TLS loadmodule "tls.so" #!endif 4. #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif Regards Deepak Jaiswal

2 1

Siremis Install: Unable to write in the session save path (CentOS 6)
by Trent 16 Aug '17

16 Aug '17

Hello. Doing a new install, and I am in the stage of accessing the web interface for Sermis which is located at /var/www/html/siremis Full error: Unable to write in the session save path [/var/www/html/siremis/session]Unable to write in the session save path [/var/www/html/siremis/session] I would think this is a single access problem, but I have checked ownership in httpd.conf, and the files, and directories. All have the apache ownership assigned to them. I have even given the "session" folder access of 777 with still no change in the error. -rw-r--r--. 1 apache apache 10176 May 18 05:18 ChangeLog drwxr-xr-x. 4 apache apache 4096 May 18 05:18 misc drwxr-xr-x. 7 apache apache 4096 May 18 05:18 openbiz -rw-r--r--. 1 apache apache 1019 May 18 05:18 README.md drwxr-xr-x. 14 apache apache 4096 May 18 05:18 siremis In addition, there is nothing about this appearing in the httpd error_log, nor access_log for an access problem. Any suggestions? Thanks!

3 3

UAC random_delay not that random
by Ivo Vastert 15 Aug '17

15 Aug '17

Hi, I'm using the UAC module in Kamailio to setup about 500 outbound registration towards a single host. The issue I'm currently experiencing is that when we start Kamailio all this 500 registrations are initiated at once towards the host which can't handle all the requests at the same time. Therefor I tried to spread the registrations using the reg_random_delay function: modparam("uac", "reg_random_delay", 60) I would expect that this would spread the registrations over about a minute but in reality the numbers are sprad as follow: 2017/08/09 21:59:14: 301 2017/08/09 21:59:15: 332 2017/08/09 21:59:16: 253 2017/08/09 21:59:18: 120 2017/08/09 21:59:22: 69

1 0

KSR.pv.get("$(hdr(P-Asserted-Identity){uri.user})") returns unable to get pv value
by Muhammad Zaka 15 Aug '17

15 Aug '17

Hi We are experiencing issue with python script KSR.pv.get using branch 5.0.2 as it fails "unable to get pv value" Example Code: if KSR.pv.get("$ai"): fromid = KSR.pv.get("$(hdr(P-Asserted-Identity){uri.user})") Error: Aug 14 18:38:23 sbc99-kam-dev /usr/sbin/kamailio[30459]: PYT {INVITE}: ERROR: app_python [apy_kemi.c:637]: sr_apy_kemi_f_pv_get(): unable to get pv value for [$(hdr(P-Asserted-Identity){uri.user})] Is anybody have any idea how to workaround? Many Thanks Regards Muhammad Zaka

2 4

Kamailio 4.3.4: receive_msg(): no via found in reply
by Grant Bagdasarian 15 Aug '17

15 Aug '17

Hello, We're testing kamailio 4.3.4 with two kamailio's communicating with each other. One of them is a gateway with rtpengine at the carrier side, the other is a basic sip router. When calling a destination and cancelling the INVITE a 487 is received from the carrier and directly ACK'd by the first kamailio and in turn is forwarded to the second. But the 487 response received by the 2nd kamailio doesn't have a Via header which results in kamailio printing the error: receive_msg(): no via found in reply. Any ideas why the forwarded 487 doesn't contain a Via header and how to fix this? Note: IP's are changed. Carrier: 10.0.0.1 Kamailio 1: 10.14.0.1 Kamailio 2: 10.14.0.2 U 10.0.0.1:5060 -> 10.14.0.1:5060 SIP/2.0 487 Request Cancelled. Via: SIP/2.0/UDP 10.14.0.1;branch=z9hG4bK8e28.642c68037d059634308d4be8fe3f0cdc.0. From: "+31612345678" <sip:+31612345678@10.14.0.3:5060>;tag=109ac722. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. Contact: <sip:10.0.0.1:5060>. CSeq: 2 INVITE. Server: sbc_5. Content-Length: 0. . # U 10.14.0.1:5060 -> 10.0.0.1:5060 ACK sip:003112345678@10.0.0.1:5060 SIP/2.0. Via: SIP/2.0/UDP 10.14.0.1;branch=z9hG4bK8e28.642c68037d059634308d4be8fe3f0cdc.0. Max-Forwards: 67. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. From: "+31612345678"<sip:+31612345678@10.14.0.3:5060>;tag=109ac722. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. CSeq: 2 ACK. Content-Length: 0. . # U 10.14.0.1:5060 -> 10.14.0.2:5060 SIP/2.0 487 Request Cancelled. From: "+31612345678" <sip:+31612345678@10.14.0.3:5060>;tag=109ac722. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. Contact: <sip:10.0.0.1:5060>. CSeq: 2 INVITE. Server: sbc_5. Content-Length: 0.

4 16

Reinvite, rewrite reply SDP?
by Yu Boot 15 Aug '17

15 Aug '17

Finally I find a root of a problem with my T.38 reinvites. A->(NAT)->Kami->PSTN When NATted caller does reinvite, and remote side answers with 200 OK+SDP, Kami (why?) doesn't rewrite SDP for its own IP (so NATted caller can receive RTP) and leaves remote's IP as media address, so of course there is no RTP/UDPTL. All other NAT related stuff works fine. Even if remote side sends me T.38 reinvite first, everything works properly. Config file is default from Kami sources. Any ideas?

2 2

RTPEngine recording
by Alex Balashov 14 Aug '17

14 Aug '17

I have gathered that RTPEngine has recently, or perhaps some time ago, evolved a recording feature set: https://kamailio.org/docs/modules/5.0.x/modules/rtpengine.html#rtpengine.f.… Does anyone have any experience using it with Kamailio? How does it work? Any gotchas or pitfalls? It looks like recording is done via a separate daemon specifically for that purpose. Does it emit directly in a playable format, or more or less just dump the raw, RTP-encapsulated frames? -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 8

DNS Geo Location Kamailio servers, registration and routing
by Sadeesh Radhakrishnan 14 Aug '17

14 Aug '17

We have kamailio servers that do DNS-Geo-Location on different locations, using same domain name. So the users from a country will be preferred to his nearby server by DNS routing. Similarly users from different countries will be registering to their nearby kamailio sip servers. All the servers share a common user location table on database. Now the users are getting registered fine, with no issue. We want the users at one location should be able to call user@other-location using the user location contact information. When we tried, the INVITE is sent on common domain name and that reverse resolves dns to same server and gets a 404 not found. I need some suggestion / ideas in order to make it happen. Thanks

2 1

set digest mime
by wsotest.512 14 Aug '17

14 Aug '17

2 1

Inconsistency in sems v1.6 installation: sems.conf is missing
by leslie mekiyess 14 Aug '17

14 Aug '17

For some reason, the installation creates the etc directory under the /usr/local/etc/sems/ : make[2]: Entering directory `/usr/src/sems-1.6.0/apps/conf_auth' mkdir -p **/usr/local/etc/sems/**etc**/** installing conf_auth.conf make[2]: Leaving directory `/usr/src/sems-1.6.0/apps/conf_auth' Then at the make install ends the following is printed: *** install complete. Run SEMS with *** *** /usr/local/sbin/sems -f /**usr/local/etc/sems/sems.conf** At bottom libe, while following the spain document instructions, I figured out that the sems.conf wasn;t created at the /usr/local/etc/sems/ path. I can create a new one or even copy this one found : /usr/src/sems-1.6.0/apps/mobile_push/load_test/sems_cfg/core/etc/sems.conf but I am afraid of further files/directory inconsistencies. Any assistance is very appreciated.

2 1

acc billsec
by Aidar Kamalov 14 Aug '17

14 Aug '17

Hello, is there any way to log in cdr duration and billsec? -- Aydar A. Kamalov

3 3

UAC RPC Commands problem
by Volkan Oransoy 14 Aug '17

14 Aug '17

Hi, I use Kamailio 5.0. I try to run RPC commands for UAC module but some of them doesn’t work like documented. kamcmd uac.reg_dump kamcmd uac.reg_reload These commands work as expected. But commands with extra arguments give error. kamcmd> uac.reg_refresh 908508850000 error: 400 - Invalid Parameters kamcmd> uac.reg_info l_uuid 908508850000 error: 404 - Record not found Do I miss something? Thanks, /Volkan

2 3

Carrierroute, fallback, reinvites
by yu＠yu-boot.ru 12 Aug '17

12 Aug '17

Hello. Trying to implement fallback route for carrierroute and got sick to make reinvites working correctly. Until fallback route triggers, everything is OK. Anyone has working config for that purposes (NAT traversal, reinvites, carrierroute ), please? And abstract question - is it generally a good idea to implement a simple softswitch (not PBX like Asterisk) like Yate or Freeswitch with Kami? Now I just want to learn Kami and learn SIP deeper in general.

2 1

Route calls from Kamailio to Asterisk behind NAT
by vs＠cell.tel 12 Aug '17

12 Aug '17

3 5

Re: [SR-Users] [OpenSIPS-Users] ClueCon Slides: Industrial Grade FreeSWITCH, Scaling, Balancing, High Availability, SIP, WebRTC
by Abdul Basit 11 Aug '17

11 Aug '17

nice presentation!!! -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On 10 August 2017 at 23:12, Giovanni Maruzzelli <gmaruzz(a)gmail.com> wrote: > *Industrial Grade FreeSWITCH Scaling, Balancing and High Availability for > SIP and WebRTC* > > Scaling your FreeSWITCH platform to serve a growing user base is a > critical challenge. We'll go through the best techniques, practices, and > implementations for Voice and Video Calls, Conferencing, WebRTC, SIP, > Chatting, Presence and Instant Messaging > Slides can also be downloaded from http://opentelecom.it/cluecon2017.pdf > > Enjoy ClueCon! > > -giovanni > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _______________________________________________ > Users mailing list > Users(a)lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >

1 0

Dispatcher call id hash for server side invite
by Ryan Wagoner 11 Aug '17

11 Aug '17

For a call invite from a phone to an load balanced asterisk server farm I can use ds_select_dst with hash over callid for the algorithm. What I don't understand is what happens for a server side invite. Say user A calls user B. The server will send and invite to user B's device. User B's device will reply and ds_select_dst won't have the call id hash so it will choose a random server, which might not be the server sending the invite. How do I mark these server side invites so the call hash is known by kamailio? Or am I thinking about this the wrong way? Thanks, Ryan

3 6

Issue in Dialplan Module
by Logeshwaran G 10 Aug '17

10 Aug '17

Hi, I am facing an Problem in Dialplan Module. Scenario: ======== I am Dialling 00091xxxxxxxxxx number, I need to remove only the prefix 000, then the call placement will be 91xxxxxxxxxx. My dialplan rule: ============== MariaDB [kamailio]> select * from dialplan; +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | 1 | 1 | 1 | 1 | ^000$ | 0 | ^000$ | | | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ But the prefix 000 is not replaced. Please Guide me to Resolve this issue. Do I need to change somet rules?? Additional info: ============= But, With additional to the above dialplan data rule I used the below rule. +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | 1 | 1 | 1 | 1 | ^000$ | 0 | ^000$ | | | 111 +----+------+----+----------+-----------+-----------+-----------+----------+-------+ The above rule replacing only the 000, If we Dial 000xxxx its omitting, While Dialing 000 its replacing that with 111. Thanks & Regards, Logeshwaran

4 23

BUG: invalid branch
by Yasin CANER 10 Aug '17

10 Aug '17

2 1

Kamailio: how to route RTP media directly to endpoint
by wsotest.512 10 Aug '17

10 Aug '17

Hi all, We have usual config Kamailio + Asterisk where Kamailio play as sip and rtp proxy. Kamailio have public IP, asterisk - no. All calls between clients now going like that: UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> Kamailio (rtpproxy) --> Asterisk --> UserB All clients of course from Internet and behind Nat. Main problem is amount of traffic going through Kamailio and Asterisk. We need to pay for every additional GB behind limit in tariff plan to hosting provider. So we decided to try route all rtp traffic between users directly. UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> --> --> UserB Is it possible at all? Maybe someone already did it . -- BR, Alex

3 2

IMPORTANT: acc module - SQL define removed and diameter code relocated
by Daniel-Constantin Mierla 09 Aug '17

09 Aug '17

Hello, I just pushed a bit of clean up to acc module, therefore be aware of following changes: 1) the define conditions on SQL_ACC were removed -- this was enabled for more than 10 years and only made the code look complex and hard to follow up its logic. 2) the code related to DIAMETER accounting was relocated to acc_diameter (new) module. It was a consistent size of code that was not enabled for sooo... long. It is now a dedicated module, similar to acc_radius. The diameter accounting code, even a new module now, is in the same stage, compiling but not tested, in pair with auth_diameter module, it may work, but very likely not. In summary, what's important for those using the acc module -- it offers the same functionality as it was enabled by default in the past 10 years or more, only the unused code was relocated. It offers the functionality of writing accounting records to syslog and sql databases. The acc module is now slimmer, only with the code that it needs, therefore easier to maintain and enhance for the future. Hopefully, there was no side effect with this update -- anyhow, if you find any issue, just open a bug report on github project. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Kamailio: how to route RTP media directly to endpoint
by Aleksey Hodin 09 Aug '17

09 Aug '17

Hi all, We have usual config Kamailio + Asterisk where Kamailio play as sip and rtp proxy. Kamailio have public IP, asterisk - no. All calls between clients now going like that: UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> Kamailio (rtpproxy) --> Asterisk --> UserB All clients of course from Internet and behind Nat. Main problem is amount of traffic going through Kamailio and Asterisk. We need to pay for every additional GB behind limit in tariff plan to hosting provider. So we decided to try route all rtp traffic between users directly. UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> --> --> UserB Is it possible at all? Maybe someone already did it ... -- BR, Alex

1 0

Registrar limit
by Yu Boot 09 Aug '17

09 Aug '17

How to limit one user login with one AOR (IP) and always use IP from last registration attempt?

2 1

RFC: Moving misc/obsolete/ to its own git repository
by Daniel-Constantin Mierla 09 Aug '17

09 Aug '17

Hello, trying to make the git repository a bit slimmer and avoid unuseful indexing of unused modules, I am considering to move misc/obsolete to its own github repository (planned to be named kamailio-obsolete). The initial idea was to keep there modules that were duplicated or no longer maintained, just in case someone will come later and wants to revive it. However, iirc, it is more than 4 years and nobody touched them, so very likely it requires a bit of work to revive any of them, so a fresh re-import will be the same from an external repository. If anyone still wants to keep them in the main kamailio repository, respond on the mailing list and present the benefits. If not, I will do the operation in the next several days. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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