sr-users

sr-users@lists.kamailio.org

35 participants
37544 discussions

How to add route header
by Senthil K 02 Sep '14

02 Sep '14

Hi All, Kindly check my issue and please provide me your suggestions. In my requirement, I want to add a route header to the INVITE message which is sent out from the Kamailio server, Basically I will be making a call and that INVITE will reach the Kamailio, then the Kamailio will send the INVITE to another node, here at this situation when the Kamailio is sending the INVITE to the next node, I want to add a route header in INVITE. Please let me know how can I hard code a route header to the outgoing INVITE from Kamailio server, Kindly help me Kind Regards SenthilK

2 1

PhonerLite cant login
by Donovan 02 Sep '14

02 Sep '14

Hi, I hope you can help me. My login details are correct but it keeps saying timeout. It just suddenly stopped working. I have made no changes? https://url.odesk.com/_01tJ3v5df4REETK4PV-B3hQtMyc7vTyOQO Best, Don --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

2 1

To mark user as unreachable with nathelper.
by Igor Pavlov 02 Sep '14

02 Sep '14

Hi, colleagues. Is it possible to mark user as unreachable (ex. by setting flag in location table) with nathelper-module when it pinging user? I need something like asterisk's qualifying. ----- Best regards, Igor Pavlov

2 1

Adding a media attribute to SDP
by aft 01 Sep '14

01 Sep '14

Hi, I want to add this media attribute to SDP. a=ptime:<packet time> [1] Its not covered in SDPops module [2] its not added when i use rtpproxy_manage("z<packet size>") to repacketize the rtp stream. So how should i do it? -- -Cheers -Arif

2 4

Wrong ACK to Provider
by Yuriy Gorlichenko 01 Sep '14

01 Sep '14

Hello. I try to provide call scheme: internal client -> asterisk -> Kamailio -> provider -> external endpoint call when I make call I see this: asterisk kamailio provider invite --> invite --> <-- 407 ACK --> invite w/Auth --> <-- 100 <-- 100 <-- 180 <-- 180 <-- 183 <-- 183 <-- 200 <-- 200 ACK --> ACK --> My problem with last ACK, that I send to provider. Provider ignores it, and sends me some OK packets. As resultI can notend session ( answer to BYE 481 - transaction does not exists). I think it is wrong ACK but can not undrtand where I do mistake. Please help me to find it: My invite (with Auth creditans): IP 10.0.1.18.5068 > my.provider.ip.5060: UDP, length 1606 E...]. .@..R ...6........N0TINVITE sip:12345678900@my.provider.ip:5060 SIP/2.0 Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> Via: SIP/2.0/UDP my.external.ip:5068;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 Max-Forwards: 70 From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068> Contact:<provider_username@my.external.ip:5068> Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Wed, 27 Aug 2014 22:02:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 544 Proxy-Authorization: Digest username="provider_username", realm="my.provider.ip", nonce="U/5Wv1P+VZNjFBLf6fwPizgd6iLto5St", uri="sip:12345678900@my.provider.ip:5060", qop=auth, nc=00000001, cnonce="2888860875", response="9f23110471fe9ff751cd55466e70ded2", algorithm=MD5 v=0 o=root 1370647246 1370647246 IN IP4 12.34.56.78 s=Asterisk PBX 12.5.0 c=IN IP4 12.34.56.78 t=0 0 a=ice-lite m=audio 30296 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30297 a=ice-ufrag:p5k92ynl a=ice-pwd:FIOYKt96NlBfEqKsQipUuadUev1g a=candidate:vV3V06Tv Provider trying IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 500 E.........PX6... ..........ySIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP my.external.ip:5068;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1;rport=5068;received=12.34.56.78 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068> Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 provider ringing IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 1098 E..f......M.6... ........RV.SIP/2.0 180 Ringing Via: SIP/2.0/UDP my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE Contact: <sip:12345678900@67.192.253.160:5060;transport=udp> User-Agent: Plivo Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 Remote-Party-ID: "12345678900" <sip:12345678900@my.provider.ip >;party=calling;privacy=off;screen=no provider seesion in progress IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 1887 E..... ...,.6... ........g.DSIP/2.0 183 Session Progress Via: SIP/2.0/UDP my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE Contact: <sip:12345678900@67.192.253.160:5060;transport=udp> User-Agent: Plivo Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 742 Remote-Party-ID: "12345678900" <sip:12345678900@my.provider.ip >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1409149800 1409149801 IN IP4 67.192.253.160 s=FreeSWITCH c=IN IP4 67.192.253.160 t=0 0 a=msid-semantic: WMS uIWGGSqM8mUp5NEgQ9CU0svyzqjzisqD m=audio 27180 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=ssrc:326362635 cnam provider OK IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 2026 E..... ...,.6... ...........SIP/2.0 200 OK Via: SIP/2.0/UDP my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> Fл2rom: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE Contact: <sip:12345678900@67.192.253.160:5060;transport=udp> User-Agent: Plivo Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE SupлЛ o=FreeSWITCH 1409149800 1409149801 IN IP4 67.192.253.160 s=FreeSWITCH c=л2IN IP4 67.192.253.160 t=0 0 a=msid-semantic: WMS uIWGGSqM8mUp5NEgQ9CU0svyzqjzisqD m=audio 27180 RTP/AVP 0 my ACK IP 10.0.1.18.5068 > my.provider.ip.5060: UDP, length 614 E...]...@... ...6........n.hACK sip:12345678900@my.provider.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.external.ip:5068;branch=z9hG4bK48ba.4250e4d315c4aa6697b6d7f70e861b62.0 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK4d28fc11;rport=50600 Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> Max-Forwards: 70 From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH Contact:<provider_username@my.external.ip:5068> Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 So after this ACK provider still sends me 200 OK and my server still sends ACK.... tags and call-id always one. Thanks

3 11

Re: [SR-Users] Presence - How to force Kamailio to use one subscription
by Daniel-Constantin Mierla 01 Sep '14

01 Sep '14

Hello, if you have a patch and want to get it in the main repository, send it and if all looks ok, we can commit it -- that will ensure the feature is there with new releases of kamailio. Cheers, Daniel On 01/09/14 10:28, Surendra wrote: > Hi Veerabhadra, > > Kamailio doesn't support this. We had done this by changing the code > logic. > > Regards > surendra > > > > > > > _______________________________________________ > 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 0

[OT] vim auto-complete plugin for kamailio.cfg
by Daniel-Constantin Mierla 01 Sep '14

01 Sep '14

Hello, I published a vim plugin offering basic autocomplete for kamailio configuration file. It is available via githup at: * https://github.com/miconda/vim-extensions More details about the plugin (including how to install and use) can be read at: * https://github.com/miconda/vim-extensions/blob/master/plugin/kamailio/READM… It was written more as a proof of concept so far, hopefully interested people will join development to bring it to a 'smarter auto-complete' plugin. Right now it relies on custom autocomplete feature from vim -- once activated, as you type a word, by pressing CTRL+x CTRL+u a pop up box presents possible completion by matching the typed part of the word. The dictionaries include module functions and parameters, script variables, core functions and parameters (they were generated with some scripts grepping various files, thus they might not be complete). It is not (yet) aware of context, so it displays all matches (e.g., even when in a route block, it displays core parameters), so again, everyone is welcome to contribute/fork/pull requests to develop the plugin further. The readme lists some other goals as well. Hopefully it is going to be useful at least for some people in the community! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 0

Re: [SR-Users] Presence - How to force Kamailio to use one subscription
by Surendra 01 Sep '14

01 Sep '14

Hi Veerabhadra, Kamailio doesn't support this. We had done this by changing the code logic. Regards surendra

1 0

Presence - How to force Kamailio to use one subscription per user?
by Veerabhara Gundu 30 Aug '14

30 Aug '14

Hello Friends, I have a scenario, where clients are going away and re-registering with new IP and Port numbers. Presence server keep sending NOTIFY messages for both old Subscription and new Subscription for same clients till the old subscription expires. It's unnecessary traffic and processing on both client and server. I am looking for a way to force Kamailio to use one subscription per user similar to mode:04 of location table. If it's not possible, looking for a way to remove subscription from config file when I detect this error.We have an option to remove automatically Subscription when NOTIFY failed, but this is not working for TCP based clients. In TCP mode, when Kamailio not event trying to send NOTIFY if Kamailio already knows that TCP connection is broken and replying with 500 error. Any alternative suggestion also welcome. Thanks, Veera Thanks, VG

1 3

What does "::" operator do in statements related to hashtable?
by aft 30 Aug '14

30 Aug '14

Hi, >From kamailio documentation, the usage of hashtable is given as : modparam("htable", "htable", "a=>size=4;") ... $sht(a=>test) = 1; $sht(a=>$ci::srcip) = $si; I get it that in the first statement, a is the hashtable, a new key-value pair is added to one of its empty bucket (test,1). What is happening in the second statement? -- -Cheers -Arif

3 3

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