[SR-Users] Howto send rtp stats infos to homer

2016-11-23 Thread Oliver Roth
Hi there We are using homer to analyce and monitor traffic. I did not get a solution to send x-rtp or p-rtp stats to homer. How can I find infos about jitter, sent/received packages, lost packages, etc.? Our current infos are based on Rtpstat provided by kamailio. At the moment I do

Re: [SR-Users] ***SPAM***Re: SDP Codec not removed with RTPengine - but with rtpproxy it worked

2016-11-23 Thread Oliver Roth
: [SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked Move the record_route() function to be executed somewhere after the msg_apply_changes(). Cheers, Daniel On 18/11/16 10:10, Oliver Roth wrote: > Found the problem with msg_apply_changes: > cannot apply msg changes

Re: [SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked

2016-11-18 Thread Oliver Roth
RTPEngine, you should call msg_apply_changes() after removing the codec. Thanks, Carsten 2016-11-18 9:39 GMT+01:00 Oliver Roth : > Hi, > > > > The codec is removed before sending it to rtpengine … > > See the log below > > > > > > > > Nov 18 09:37

Re: [SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked

2016-11-18 Thread Oliver Roth
telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - Any ideas? Kr, Oli Von: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Gesendet: Freitag, 18. November 2016 09:27 An: Oliver Roth ; Kamailio (SER) - Users Mailing List Betreff: Re: AW: [SR-Users] SDP Codec not removed with RTPengine

Re: [SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked

2016-11-16 Thread Oliver Roth
with RTPengine - but with rtpproxy it worked Hello, are you executing rtpengine_manage() before or after removing the codec? Cheers, Daniel On 16/11/16 10:03, Oliver Roth wrote: Hi there I have the following problem – I need to remove a codec in the initial INVITE. This happens since I

[SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked

2016-11-16 Thread Oliver Roth
Hi there I have the following problem - I need to remove a codec in the initial INVITE. This happens since I changed from rtpproxy to rtpengine. I changed all rtpproxy_manage() to rtpengine_manage(). Originating INVITE with the "clearmode" m=audio 9196 RTP/AVP 8 0 125 101 a=rtpmap:0 PCMU/8000 a

[SR-Users] Advice for routing and sbc functions

2016-09-12 Thread Oliver Roth
Hi all I am using kamailio for several years now and I am really happy with it! Now an new scenario to solve: Sip-carrier A --> Kamailio loadbalancer --> multiple kamailio routing gateways --> freeswitch/sbc/media-gw/whatever --> sip - carriers B,C & D Kamailio gateways: They are doing differ

[SR-Users] Contact header / bye

2016-08-15 Thread Oliver Roth
Hi all I do have a problem with getting a propper interconnection to our new carrier.. Situation as following User --> Class5 switch --> Kamailio --> carrier I always get the ip of the class5 switch in the contact. But my carrier needs the kamailios contact ip - because of the ack handling. I r

Re: [SR-Users] ACK / BYE transaction problem

2016-07-01 Thread Oliver Roth
fault kamailio.cfg for latest kamailio versions. Cheers, Daniel On 29/06/16 17:20, Oliver Roth wrote: Yes the ip for the [carr] is missing. But I thought, the [gw ] should create the ack based on the transaction and send it to the [carr] My situation Class5 system [c5] --> Loadbalancer

Re: [SR-Users] ACK / BYE transaction problem

2016-06-29 Thread Oliver Roth
sers-boun...@lists.sip-router.org] On Behalf Of Oliver Roth Sent: 29 June 2016 16:04 To: Kamailio (SER) - Users Mailing List mailto:sr-users@lists.sip-router.org>> Subject: Re: [SR-Users] ACK / BYE transaction problem I removed the changes for the to header - so it is not touched all the time

Re: [SR-Users] ACK / BYE transaction problem

2016-06-29 Thread Oliver Roth
headers.. I had a similar issue 3 weeks ago. Cheers, Francisco. From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Oliver Roth Sent: 29 June 2016 15:55 To: sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org> Subject: [SR-Users] ACK / BYE transaction proble

[SR-Users] ACK / BYE transaction problem

2016-06-29 Thread Oliver Roth
Hi all Follow scenario Class5 system [c5] --> Loadbalancer kamailio (dispatcher module) [lbl] ---> gateway kamailio [gw] --> carrier [carr] I get Invites from [c5] with Request ,To, from, contact, pid in national format 0794445566 [lbl] dispatches this to [gw] For the [carr] I need internati

[SR-Users] Loadbalancer/dispatcher with dialog/cnxcc

2014-04-23 Thread Oliver Roth
Hi all Following situation 1 dispatcher/loadbalancer getting all the inbound traffic and sending it to 3 different gateway (round robin). The loadbalancer has no (or even very few) business logic. Just "in" - split to different gateway - "out" 3 sip gateway doing all the business logic like aut

Re: [SR-Users] Planning release of v4.1.3

2014-04-22 Thread Oliver Roth
Hi One point from my side https://sip-router.org/tracker/index.php?do=details&task_id=353 I did not hear anything from the developers ... KR, Oli -Ursprüngliche Nachricht- Von: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Daniel-Co

[SR-Users] kamailio / rtpproxy - remove codec

2014-04-06 Thread Oliver Roth
Hi all We use kamailio 3.3.7 and rtpproxy for enduser call-termination. In case of a fax call, we get an invite from our carrier for codecs G711a/u and T38. As our termination carrier does not support T38 and because the invite contains G711 and T38 we get back error 488. How is it possible to

[SR-Users] Module not up to date in apt-get repository

2014-03-19 Thread Oliver Roth
Hi all We updated from kamailio 3.1 to kamailio 3.3. As shown in module description the function rtpproxy_manage() should be available starting version 3.2 But if the update is done to 3.3 - we get the following error when trying to start kamailio * Not starting Kamailio SIP server: invalid c

Re: [SR-Users] Carrierroute - scan_prefix

2014-02-03 Thread Oliver Roth
Hi all Nobody an idea? To measure asr/ner ratio this is really important to us! Regards, Oli Von: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Oliver Roth Gesendet: Montag, 13. Januar 2014 09:26 An: Kamailio (SER) - Users Mailing List

[SR-Users] pdd value

2014-01-21 Thread Oliver Roth
Hi all Question about an additional cdr information. We need to get the pdd value - means a time value till the user gets a ringback or an error- to track the delay, till a connection is done --> quality issue for customer. 5.6.1 Definition of Post Dial Delay Post Dial Delay (PDD) is experience

[SR-Users] Carrierroute - scan_prefix

2014-01-13 Thread Oliver Roth
Hi all I need a solution to get the used scan_prefix in carrierroute module. We have different carriers for our destinations, sometimes split up 50%/50% or other splittings - all routes with failure routes. Now if one destination is not reachable, then I would like to get this information and r

Re: [SR-Users] call limit by trunk/user

2013-12-18 Thread Oliver Roth
#idp130984 Regards, On Wed, Dec 18, 2013 at 9:42 AM, Oliver Roth mailto:oliver.r...@triotel.ch>> wrote: Hi all I need a solution to limit the amount of concurrent calls by trunk or user/subscriber. We have the following situation: Kamailio loadbalancer ==> 3 kamailio gateway wit

[SR-Users] call limit by trunk/user

2013-12-18 Thread Oliver Roth
Hi all I need a solution to limit the amount of concurrent calls by trunk or user/subscriber. We have the following situation: Kamailio loadbalancer ==> 3 kamailio gateway with routing/business logic Kamailio is version 3.3 for all 4 systems. Loadbalancer has only the absolute minimum config t

[SR-Users] rtpproxy / sdp / problem with signalling

2013-11-27 Thread Oliver Roth
Problem with signalling - RTP gets lost! Rtpproxy not working properly? I am absolutely stuck ... cause this happens in a live environement. I have the following situation A calls B over carrier 1 - number is not valid and I get back error 404 from carrier and now freeswitch should play a messa

Re: [SR-Users] Loosing rtp only in carrierroute failureroute

2013-11-26 Thread Oliver Roth
No idea? -Ursprüngliche Nachricht- Von: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Oliver Roth Gesendet: Dienstag, 29. Oktober 2013 12:50 An: Kamailio (SER) - Users Mailing List Betreff: [SR-Users] Loosing rtp only in carrierroute

[SR-Users] voice prompts / early media and kamailio

2013-11-26 Thread Oliver Roth
Hi all Based on my problem reported with subject "error handling" I have some other questions. I think it is a conceptual question - and I do not see any solution. I would like to handle Kamailio with carrierroute / carrierfa

Re: [SR-Users] error handling

2013-11-24 Thread Oliver Roth
20:44 schrieb "Oliver Roth" mailto:oliver.r...@triotel.ch>>: Sorry - I do not understand what you mean ... I added the following code at the end of the failure_route - seems to work if (t_grep_status("404")){ xlog("L_INF

Re: [SR-Users] error handling

2013-11-24 Thread Oliver Roth
eff: Re: [SR-Users] error handling Acc_db_request Sent from my iPhone On Nov 24, 2013, at 11:20 AM, Oliver Roth mailto:oliver.r...@triotel.ch>> wrote: Hi all Question about error handling with kamailio. We send call to carrier and get back error 404. In carrierfailureroute we catch up

[SR-Users] error handling

2013-11-24 Thread Oliver Roth
Hi all Question about error handling with kamailio. We send call to carrier and get back error 404. In carrierfailureroute we catch up this error and send call to an internal freeswitch that plays a voiceprompt saying: "destination not available" In accounting this calls is collected like a "no

[SR-Users] Billing

2013-11-18 Thread Oliver Roth
Hi all I am looking for a free billing solution usable with Kamailio. I currently use the sp from siremis - so far it works fine! I need to get a solution where I can add credit limits for users/trunks or credit limits for carriers (eg. If we made a prepayment). I thing JBilling could be great -

[SR-Users] extract part of string / INVITE

2013-11-13 Thread Oliver Roth
Hi all I need to extract a part of the INVITE msg: INVITE sip:+4179615@82.197.185.185;user=CSC10824 SIP/2.0 I need 10824 in a avp variable. Is there a regex function where I can extract this? Or how can this be done? Something like: Avp(myCSC) = string after CSC - length 5 Please be aware,

Re: [SR-Users] WG: Carrierroute module

2013-11-11 Thread Oliver Roth
iling List Betreff: Re: [SR-Users] WG: Carrierroute module http://sip-router.1086192.n5.nabble.com/Error-loading-carrierroute-in-Kamailio-4-0-x-td122311.html On 10 November 2013 10:46, Oliver Roth wrote: > Nobody an idea how to fix this? > > > > > > > > Hi > > > &

Re: [SR-Users] implemented specifiers not processed

2013-11-11 Thread Oliver Roth
e and replace xlog() with appropriate new function name Cheers, Daniel On 11/10/13 1:01 PM, Oliver Roth wrote: Hi all Having some problems with the following part of the script in a timer: We have some kamailios running doing more or less the same job - they are used by a loadbalancer kamailio. Act

[SR-Users] implemented specifiers not processed

2013-11-10 Thread Oliver Roth
Hi all Having some problems with the following part of the script in a timer: We have some kamailios running doing more or less the same job - they are used by a loadbalancer kamailio. Actually I wanted to check the ip or systemname of the current system do perform some actions depending on the

Re: [SR-Users] Regex question

2013-11-03 Thread Oliver Roth
(SER) - Users Mailing List Cc: kamai...@aaronlux.com Betreff: [SR-Users] Regex question Oliver Roth writes: > I need to do some string operations in kamailio.cfg. > How can I get the cli from the following string: > sip:+41523940347@195.216.67.103;user=phone > > I only would nee

[SR-Users] Regex question

2013-11-03 Thread Oliver Roth
Hi all Maybe a very simple question - but I cannot see the solution - I am more or less newbie ;) I need to do some string operations in kamailio.cfg. How can I get the cli from the following string: sip:+41523940347@195.216.67.103;user=phone I only would need +41523940347 in a variable Some

[SR-Users] Loosing rtp only in carrierroute failureroute

2013-10-29 Thread Oliver Roth
Hi all We do have a strange problem with loosing rtp in case of carrierroute - failureroute. If we send traffic directly to the failure gateway, we do have rtp without any problem Situation: Ua --> Freeswitch --> kamailio 3.3 --> gw 1 Error 404 --> failureroute --> media gateway voiceprompt f