sr-users August 2017

sr-users@lists.kamailio.org

89 participants
144 discussions

by Logeshwaran G

Hi, It is the correct way of using the cr reload command in xmlrpc through postman? <?xml version='1.0'?> <methodCall> *<methodName>cr.reload</methodName>* </methodCall> Thanks & Kind Regards, Logeshwaran G

6 years, 8 months

Save original $fU, $rU

by yu＠yu-boot.ru

Have a problem with Asterisk. When callee behind Asterisk hangs up first, it sends back BYE with swapped From and To fields, which affects Kami CDRs rendering them useless. I've tried to save original From/To to AVPs on initial INVITE but they are empty when CDR is written. How to solve this? -- Отправлено из myMail для Android

6 years, 8 months

dp_translate_cr_route, R-URI

by Yu Boot

Hello. #cut off prefix "9" dp_translate("1"); ... #searching in db by called number without prefix... if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ ... and failing. Is it normal behavior that cr_route "sees" original $rU even after dp_translate manipulations?

6 years, 8 months

Kamailio / Asterisk and multidomain

by Cyrille Demaret

Hi, I'm new to Kamailio and I'm trying to use it as a front SIP proxy to one or more asterisk. Unlike to the "Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration" tutorials, I would like to let Kamailio handle registrations, calls between users and other basic functionalities. Asterisk will only handle advanced features like voicemail, advanced dialplan configuration,. My problem is that It needs to be multidomain. My network will be as follow with 2 network interfaces for Kamailio and a private lan between Kamailio and Asterisk : Public interface ip 1.2.3.4 ----> [Kamailio] <----- Private interface 192.168.100.10 <----------------> 192.168.100.11 [Asterisk] When I send a call to asterisk, the domain is sent in the from field of the INVITE and I can do what I need on the Asterisk dialplan (I can get the SIP domain using the ${SIPDOMAIN} variable). My problem is when I need to send a call back to Kamailio for example to reach another user of the domain. I'm using Asterisk 14 with PJSIP with the following config : [kamailio] type=endpoint transport=transport-udp context=from-kamailio disallow=all allow=ulaw aors=kamailio [kamailio] type=aor contact=sip:192.168.100.10:5060 [kamailio] type=identify endpoint=kamailio match=192.168.100.10 If I use this dial string in my Asterisk dialplan "PJSIP/ kamailio /sip:200@testdomain.com", Asterisk contact directly testcomain.com without going through the local IP of my Kamailio. I can't send a domain to Kamailio in the INVITE request. Does anyone can help me on this or maybe simply tell me that I'm not going to the good direction? :) Thank you, Cyrille

6 years, 8 months

Siremis - can't add carrier without prefix/suffix

by Yu Boot

Hello Suddenly revealed that I can't add carriers without rewrite_prefix/rewrite_suffix via Siremis. But I don't need these fields! And kamctl adds these rules fine. What's a problem? Screenshot with error http://tinypic.com/r/68v493/9

6 years, 8 months

problems with use_dst_blacklist and ICMP "Destination Unreachable"

by Valentin Christoph

Hi all, Question: * Does the TM destination blacklist work, if destination host responds with ICMP "Destination Unreachable"? Discussion: I have got an IMS scenario, where I think the destination blacklist is not working and try to find the reason. I use UDP between two Kamailio SIP Proxies (P-CSCF and S-CSCF) and send a SIP request to the primary instance of the S-CSCF. However the destination Kamailio has been stopped with kamctl stop before and so the destination machine responds with ICMP "Destination Unreachable". DNS Failover towards secondary S-CSCF instance works perfectly. Destination Blacklist does not work. I.e. the next SIP Request is sent to the primary host again, before failover to secondary host applies. I would have expected the request would be sent to secondary host immediately. >From the description in doc/dst_blacklist.txt [...] A destination is added to the blacklist when an attempt to send to it fails (e.g. timeout while trying to send or connect on TCP), or when a SIP timeout occurs while trying to forward statefully an INVITE (using tm) and the remote side doesn't send back any response.[...] I think the destination blacklist does not work, because an ICMP Response "Destination Unreachable" is received. Second question: * How could I avoid the problem? Thanks, Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

6 years, 8 months

rtpproxy: One direction rtp not being relayed

by Jon Bonilla (Manwe)

Hi all After a migration, a customer is reporting one way audio on incoming calls. A tcpdump capture shows 65 rtp packets in both directions but rtpproxy log shows RTP stats: 1 in from callee, 64 in from caller, 65 relayed, 0 dropped Any thoughts? cheers, Jon

6 years, 8 months

sercmd dispatcher.list command giving an error: 500 - No Destination Sets

by Deepak Jaiswal

Hi, I am new to Kamailio and trying to debug an issue. We have two servers. On one server, I am getting destination like below. sercmd dispatcher.list { SET_NO: 1 SET: { SET_ID: 1 DEST: { URI: sip:xxx.xxx.xxx.139:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1503397149 } DEST: { URI: sip:xxx.xxx.xxx.:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1503397150 } } } However, on another server, I am getting below error. sercmd dispatcher.list error: 500 - No Destination Sets sercmd -s unixs:/tmp/kamailio_ctl dispatcher.list error: 500 - No Destination Sets [root@uk-vap-002 ~]# sercmd system.listMethods cfg.add_group_inst cfg.commit cfg.del cfg.del_delayed cfg.del_group_inst cfg.diff cfg.get cfg.help cfg.list cfg.rollback cfg.set cfg.set_delayed cfg.set_delayed_int cfg.set_delayed_string cfg.set_now_int cfg.set_now_string cfg.seti cfg.sets core.arg core.echo core.flags core.info core.kill core.printi core.prints core.ps core.psx core.pwd core.sctp_info core.sctp_options core.shmmem core.tcp_info core.tcp_options core.udp4_raw_info core.uptime core.version ctl.connections ctl.listen ctl.who dispatcher.list List going on.................................... I tried to find dispatcher.list file on both server. However, it is not there on both the server. Could you please help me on this? Thanks and Regards Deepak K. Jaiswal Tech support Engineer My Phone : +91- 95351-23352, +91- 88675-01385 My Movius work line: +1 (470) 298-7210 [ovius]<http://www.moviuscorp.com/> www.moviuscorp.com<http://www.moviuscorp.com/>

6 years, 8 months

append_branch and tls

by James Cloos

[I see I posted this to -dev instead of -users; fixing that -JimC] When using rewrite_ruri() and append_branch() to set a list of next hops to try, what is the proper syntax to specify that a given route should use tcp or tls/tcp? I've mostly used a single next hop via t_relay_to_tcp() or t_relay_to_tls(), but need route stacks now. -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

6 years, 8 months

Kamailio+SEMs conference call setting

by leslie mekiyess

Hi, I have a Kamailio server , a SEM one, an SQL server and Siremis installed and working. I am able to perform p2p calls with a softmobile. But I didn't succeed to perform conference / group / speed dial calls. Besides defining some records in the database, I don't know how to complete the necessary (Kamailio/SEM ...) configuration in order to perform conference call, group and speed dial . Any assitance would very appreciated. Leslie :-)

6 years, 8 months

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sr-users August 2017