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42 participants
37519 discussions

T.38 fax, rtpproxy, re-invite
by Yu Boot 02 Aug '17

02 Aug '17

Hello. I'm using almost default kamailio.cfg with nathelper, rtpproxy, PSTN dialing and other default features. I have following scheme: FXS gateway -> (NAT) -> Kamailio -> FreeSwitch Voice calls work perfectly, no one-direction sound and so on. But with fax machine there was a problem. In wireshark I saw UDPTL from FS to Kamailio IP, and no UDPTL at all to FXS gw. I've guessed that something's wrong with a re-INVITE request processing in Kamailio. I've added this to WITHINDLG route: else if ( is_method("INVITE") ) { record_route(); } And problem was magically solved, T.38 now works OK. Is this OK solution or this may break something elsewhere? Thanks.

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Kamailio CDR "best practices"
by Yu Boot 02 Aug '17

02 Aug '17

How to make Kamailio write "normal" CDRs? acc cdr module writes start time and end time in unixtime format, that is very uncomfortable. Fnd stcond question - how to make Kamailio to write CDR in ordinary text file?

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Usleep on Reply
by Nicolas Breuer 01 Aug '17

01 Aug '17

Hello All, I'm trying to code something on the on_reply part to do something if I receive a 183 ( Session in progress ) , followed immediately by a "180" I tried to sleep 1 second the 183 and recheck if I received before a 180 but seems the "sleep" function sleeps everything ( also the received 180 ) Is a developper can confirm that ? The goal is to detect if a 180 follow the 183.

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Getting Error - error reading: Connection reset by peer (104)
by Prem Kumar N 01 Aug '17

01 Aug '17

Hi All, I have a setup with two kamailio v5.0 and a load balancer(keepalived) It seems to check the heath of kamailio server load balancer establishing the tcp conneciton and resetting the connection once its got established. and due to resetting the connection I keep on getting following log message. Jul 31 18:14:18 localhost /usr/local/kamailio/sbin/kamailio[9388]: ERROR: <core> [core/tcp_read.c:291]: tcp_read_data(): error reading: Connection reset by peer (104) ([192.168.xxx.xxx]:46574 -> Jul 31 18:14:18 localhost /usr/local/kamailio/sbin/kamailio[9388]: ERROR: <core> [core/tcp_read.c:292]: tcp_read_data(): -> [192.168.yyy.yyy]:443) Jul 31 18:14:18 localhost /usr/local/kamailio/sbin/kamailio[9388]: ERROR: <core> [core/tcp_read.c:1319]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f6547989e78 r: 0x7f6547989ef8 Jul 31 18:14:18 localhost /usr/local/kamailio/sbin/kamailio[9389]: ERROR: <core> [core/tcp_read.c:291]: tcp_read_data(): error reading: Connection reset by peer (104) ([192.168.xxx.xxx]:44279 -> Jul 31 18:14:18 localhost /usr/local/kamailio/sbin/kamailio[9389]: ERROR: <core> [core/tcp_read.c:292]: tcp_read_data(): -> [192.168.yyy.yyy]:5060) Jul 31 18:14:18 localhost /usr/local/kamailio/sbin/kamailio[9389]: ERROR: <core> [core/tcp_read.c:1319]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f6547989e78 r: 0x7f6547989ef If any one can suggest, how can i get rid of these logs. Thanks Prem

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uac_auth credentials
by Volkan Oransoy 01 Aug '17

01 Aug '17

Hi all, I use UAC module for remote registrations on my proxy box. There is no problem with incoming calls and I successfully register and receive calls. I have a Freeswitch box behind this proxy box and I want my proxy to add authorisation headers for outgoing calls transparently. I already have credentials at uacreg table for remote registrations. Is it possible use same credentials at uac_auth() function? Or should I run SQL to retrieve necessary fields to feed uac_auth? Thanks! /Volkan

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call placement issue using is_subscriber
by Logeshwaran G 01 Aug '17

01 Aug '17

Hai, I am using the below script for call placements(Local as well as LCR), LCR is working fine, Local call placement is not working, Is there I am missing something?? Please guid me. route[CHECK] { if (!is_subscriber("$ru", "subscriber", "3")) { # callee is not a local subscriber route(LCR); exit(); } else { route(LOCATION); } } Thanks

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Need help to log the call trace/flow in file
by amit Kumar 01 Aug '17

01 Aug '17

Hi Developers, I need help to log the call flow in a file. Our purpose to have only one entry per call in a file based system at the end of the call. For example: CALL-ID TO FROM START TIME END TIME .... MISSED COMPLETED 1 sip:1@abc.com sip:100@abc.com 2017-07-28 18:29:14 2017-07-28 18:29:14 yes no 2 sip:2@abc.com sip:100@abc.com 2017-07-28 18:29:14 2017-07-28 18:29:14 no yes 3 sip:3@abc.com sip:100@abc.com 2017-07-28 18:29:14 2017-07-28 18:29:14 no yes I have gone through various trace module, but its appear that all module record the call flow/logs in database. ie siptrace,acc,dialoge module. Could you please help me to log the information. one of the approach I have thought to store all the variable in AVP variables in request_route,failure_route,on_reply route..etc For ex. $avp(from)=$fu; $avp(to)=$tu; $avp(starttime)=startime $avp(endtime)=endtime etc... //log at call end in a seperate logs file and xlog("LOG_INFO",$avp(from),,$avp(to),$avp(starttime),$avp(endtime)) Please let know if this approach is correct or any other way to achieve this, Thanks, Amit

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Weird issue with kamailio relaying messages to itself
by George Diamantopoulos 31 Jul '17

31 Jul '17

Hello all, I have been toying with kamailio lately, and I thought I had gotten to the point where I had a mostly working (tm) prototype. I gave it a test drive with some real calls, however, and an issue manifested at least once, where homer received packets originating from the kamailio host, and whose destination was also the kamailio host. The dialog this manifested in is a generally problematic one, with many retransmissions occurring because of slow database access (I haven't implemented htable caching yet). No packet capture over the network is actually taking place, I'm copying everything to homer with "trace_mode"set to 1. Homer shows these messages like in the screenshot: https://imagebin.ca/v/3VGJAmovRBmo. Here's an example of a packet: 2017-07-28 14:32:29 +0300 : 2.3.4.5:5060 -> 2.3.4.5:5060 INVITE sip:1234567890@kamailio-server.org SIP/2.0 Record-Route: <sip:2.3.4.5;lr;ftag=as491cec82> Via: SIP/2.0/UDP 2.3.4.5;branch=z9hG4bK0cf.07799e3eb71f33a9ef91178ac760ebd2.0 Via: SIP/2.0/UDP 2.3.4.5;branch=z9hG4bKsr-BnCVyrWoUladI14pU7Pry-Pzy-ONy-VSQ74NU7HzeYazq2nFU2Oc5FIWiGIKq-HXex1oONpam-C6IrIXkr4FQxZRh7sM Max-Forwards: 69 From: <sip:subscriber@5.4.3.2:5061>;tag=as491cec82 To: <sip:1234567890@kamailio-server.org> Contact: <sip:2.3.4.5;line=sr-eNC05xhKefQnON1EglFrOkF0OfpzV2s9UzM9UXMNQXMn5-B0Q-s*> Call-ID: 4e5d22c3369704b97d866c8d0f3798f8@192.168.201.2:5061 CSeq: 103 INVITE User-Agent: Asterisk PBX 11.13.1~dfsg-2~bpo70+1 Date: Fri, 28 Jul 2017 11:32:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "9876543210" <sip:9876543210@192.168.201.2 >;party=calling;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 500 v=0 o=root 242468242 242468243 IN IP4 172.17.130.13 s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1 c=IN IP4 172.17.130.13 t=0 0 m=audio 59426 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:50 a=sendrecv a=rtcp:59427 a=ice-ufrag:hffIoy6x a=ice-pwd:nXB8ip7Qz8ZG5yyZRr97kGJbej a=candidate:igct4zWHEzhGCjWc 1 UDP 2130706431 172.17.130.13 59426 typ host a=candidate:igct4zWHEzhGCjWc 2 UDP 2130706430 172.17.130.13 59427 typ host Can something like this be triggered by misconfiguration in the routing scripts? Should it worry me and should I dig into it more? Could it be a bug of the siptrace module and nothing bad actually took place? I'm not sure where to start with this, so any input would be greatly appreciated. Thanks!

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Rate limiting
by NITESH BANSAL 31 Jul '17

31 Jul '17

Hello, I plan to add some rate limiting in the Kamailio. I'm already using Pike to rate limit based on the source IP. But in this case, i want to do the rate limiting per customer(regardless of the source IP), I can identify the customer based on a custom header in the INVITE, idea is to limit the customer to 'x' number of requests per second. Nitesh

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Configuration to use rtpengine for SRTP
by David Cunningham 31 Jul '17

31 Jul '17

Hello, We're configuring Kamailio 4.2 with rtpengine to act as a midpoint between a telephone using TLS/SRTP and Asterisk. There are examples out there for TLS/SRTP with WebRTC, but we're using a plain hard phone, not WebRTC. Would anyone be able to point us towards a Kamaiio configuration which: a) Tests if the destination phone (stored using usrloc) uses TLS. b) Sends RTP for calls to a TLS phone to rtpengine to be encrypted. We can assume all phones using TLS want to use SRTP. Thanks very much in advance. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019

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