- sr-users - kamailio.org

Use Kamailio as (SIP-TLS) Gateway to connect to (non TLS) fritz.box for fixed-line calls
by Lukas Wygasch 02 Sep '14

02 Sep '14

Dear Kamailio-Community, i would be very pleased when someone could help me with my private non-profit project. I do not want other people to do my work, but i'm not as good with SIP as i should to solve this problem by myself. So here is my Goal: Use Kamailio as (TLS-SIP-)Gateway to my (non-TLS)-FritzBox to do (local-country) fixed-line Calls from all over the World. In Detail: Connect to (Portforwarded / NAT behind Fritzbox-Router) Kamalio-Server with User from "Subsriber-List" by TLS-… [View More]

2 1

Regrading Kamailio
by Saurabh Srivastava 02 Sep '14

02 Sep '14

Hello Kamailio Team, I am using kamailio server successfully for some time now. I have a specific set of problem. We have a Kamailio Server setup in Amazon EC2 Cloud, the users are able to register on the server and make use of this. *Now the problem:* If I am trying to register multiple SIP clients from same Public_IP:Private_IP (i.e. alice and bob from same computer) on a kamailio server which is sitting behind Amazon EC2 Cloud Based Server, then only the first user (who sends REGISTER … [View More]

2 1

Roadmap to next major release - v4.2.0
by Daniel-Constantin Mierla 02 Sep '14

02 Sep '14

Hello, during the last devel meeting on IRC, done before the summer, we set the 4.2.0 release time frame for this autumn. As planned by that moment, we have to decide the development freezing date and more accurate milestones at the end of summer, therefore it is about the time. My proposal is to freeze development by end of September 10, 2014, have about one month of testing and release sometime around mid of October. If anyone has constraints to meet this date, just send here another … [View More]

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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 … [View More]

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 &… [View More]lt;-- 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 [View Less]

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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)… [View More]

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. … [View More]

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 … [View More]

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

Kamailio as Websocket bridge to Asterisk, and Asterisk-sent OPTIONS
by Alex Villacís Lasso 29 Aug '14

29 Aug '14

As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but otherwise has accounts with all of the avfp and dtls settings for websockets. Additionally, I have removed the bindaddr=127.0.0.1 from sip.conf and instead put a deny=0.0.0.0/0.0.0.0 and permit=127.0.0.1/255.255.255.0 in order to restrict SIP signaling to localhost. This allows asterisk to bypass rtpproxy … [View More]when signaling through a websocket. I have already established calls originating from the browser. However, I have an issue with the registration. In my setup, Kamailio receives the REGISTER from whatever source, and forwards this through UDP to Asterisk, after the multiple-domain transformation. Therefore, Asterisk sees the following in its SIP port (all traffic through localhost): REGISTER sip:pbx.villacis.com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 Max-Forwards: 69 To: <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080> From: "Alex Villac..s" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=b5c0lq4kac Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Contact: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:6b0c58ee-bdc5-47c0-aff0-963132dc0cad>";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,outbound User-Agent: SIP.js/0.6.2 Content-Length: 0 Asterisk answers this through UDP, and Kamailio forwards it through the websocket: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 From: "Alex Villac..s" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=b5c0lq4kac To: <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=as5ae2df76 Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;expires=600 Date: Thu, 28 Aug 2014 22:21:15 GMT Content-Length: 0 Then Asterisk sends this through UDP, and Kamailio again forwards it through the websocket: NOTIFY sip:fnuql6ft@192.168.3.2:47307;transport=ws SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as43c12840 To: <sip:fnuql6ft@192.168.3.2:47307;transport=ws> Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.12.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0) Since I have not implemented handling of voicemail indications, the browser answers this: SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport=5080 To: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;tag=ggu5etber9 From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as43c12840 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY Supported: outbound Content-Length: 0 After that, Asterisk wants to send an OPTIONS packet. From the point of view of Asterisk (sip set debug on), it is already sent, but never gets a response. However, tcpdump shows that the packet is never sent through the localhost interface in the first place. It is also not sent through any other interface. My guess is that since the REGISTER has a contact with transport=ws , Asterisk wants to send this through a websocket (which is disabled). So I could have to generate a contact without transport=ws . I have worked around this by setting qualify=no in the account for the websocket, but I would like a better solution, one that allows the OPTIONS packet to reach the browser, and to get the response. What is the proper way to deal with this? [View Less]

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How to use the ip parameter in rtpproxy_manage()
by Sebastian Damm 29 Aug '14

29 Aug '14

Hi, I have the following setup: UAC ---> LB ---> Proxy ---> GW In NAT scenarios the loadbalancer detects it, but the proxy communicates with the RTP proxy. I want to send the original caller IP as detected by the loadbalancer (and transported to the proxy) to the RTP proxy. As far as I understand the documentation of the rtpproxy module, I could call rtpproxy_manage with a second parameter, indicating which IP address should be sent to the RTP proxy. I tried sending a … [View More]

2 3

Help analysing segmentation fault
by Charles Chance 29 Aug '14

29 Aug '14

Hi All, I wonder if some one could help me to diagnose a recurring issue? It happens at random times/intervals and under varying load. But always, just before the time of crash, I see the same critical error in log: CRITICAL: dialog [dlg_hash.c:841]: log_next_state_dlg(): bogus event 6 in state 1 for dlg 0xb0632134 [1367:5814] with clid ' 0695dd7a346188dd24e7520e6c01092c(a)sip.sipcentric.com' and tags 'as77c89620' '' Analysing the core dump reveals: Core was generated by `/usr/sbin/… [View More]kamailio -P /var/run/kamailio.pid -m 128 -M 4 -u kamailio -g kamailio'. Program terminated with signal 11, Segmentation fault. #0 0x081a737c in parse_uri (buf=0x3a70006e <Address 0x3a70006e out of bounds>, len=275, uri=0xbfa2fd2c) at parser/parse_uri.c:389 389 scheme=buf[0]+(buf[1]<<8)+(buf[2]<<16)+(buf[3]<<24); (gdb) frame 1 #1 0x008fe5dd in dialog_publish (state=0x903f37 "Trying", ruri=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180, initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0, localtarget=0x0, remotetarget=0x0, do_pubruri_localcheck=1) at dialog_publish.c:275 275 if (parse_uri(ruri->s, ruri->len, &ruri_uri) < 0) { (gdb) p *ruri $1 = {s = 0x3a70006e <Address 0x3a70006e out of bounds>, len = 275} (gdb) up #2 0x008ff277 in dialog_publish_multi (state=0x903f37 "Trying", ruris=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180, initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0, localtarget=0x0, remotetarget=0x0, do_pubruri_localcheck=1) at dialog_publish.c:387 387 dialog_publish(state,&(ruris->s),entity,peer,callid,initiator,lifetime,localtag,remotetag,localtarget,remotetarget,do_pubruri_localcheck); (gdb) p *ruris $2 = {s = {s = 0x3a70006e <Address 0x3a70006e out of bounds>, len = 275}, next = 0x0} (gdb) up #3 0x0090187a in __dialog_created (dlg=0xb0632134, type=2, _params=0x6db064) at pua_dialoginfo.c:470 470 dialog_publish_multi("Trying", dlginfo->pubruris_caller, &(dlg->from_uri), (include_req_uri)?&(dlg->req_uri):&(dlg->to_uri), &(dlg->callid), 1, dlginfo->lifetime, 0, 0, 0, 0, send_publish_flag==-1?1:0); (gdb) p *dlginfo->pubruris_caller $3 = {s = {s = 0x31590014 <Address 0x31590014 out of bounds>, len = 275}, next = 0xb0b5fd00} (gdb) p *dlginfo->pubruris_caller->next $4 = {s = {s = 0x3a70006e <Address 0x3a70006e out of bounds>, len = 275}, next = 0x0} In config, for pua_dialoginfo we are enabling the option "use_pubruri_avps" and setting "pubruri_caller_avp" and "pubruri_callee_avp" accordingly. Therefore, in pua_dialoginfo.c it is using get_str_list() function to set dlginfo->pubruris_caller from the avp. Could this be some race condition or something completely different? Thanks in advance, Charles -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB. [View Less]

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Add websocket header to SIP
by Kamrul Khan 28 Aug '14

28 Aug '14

Hi, I want to append the origin header of my incoming websockets to the SIP INVITES. I tried something like the below but didnt work: if (ws_handle_handshake()) { if(method=="INVITE"){ insert_hf("Origin: $hdr(Origin)"); } exit; } Is there any other way to do it ?

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[Kamailio-4.1.5] [/etc/init.d/kamailio/start] Error
by Djamel Bahamid 28 Aug '14

28 Aug '14

Hello, When I try to execute the /etc/init.d/kamailio start, I've this error : [root@gtk:~][Mon 18/08/2014 16:21:58] # /etc/init.d/kamailio start Not starting Kamailio SIP server: invalid configuration file! failed failed ERROR: loading config file(/usr/local/etc/kamaiilio/kamailio.cfg): No such file or directory failed failed Thank you for your help, Best Regards, Djamel BAHAMID.

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rfc: distributing dialog profiles
by Daniel-Constantin Mierla 28 Aug '14

28 Aug '14

Hello, if you are on sr-dev list, you should have seen the last commits to dialog module, which add support for remote profiles. As most of you know (or at least the users of dialog module), profiles are used to group active calls, their main purposes being counting dialogs belonging to a profile for implementing active call limits (e.g., all calls in the system, call from/to a specif user/address). It was working on a single Kamailio instance, but one could use other meanings to … [View More]interact with other kamailio instance to get the number of calls across multiple instances. The purpose of remote profiles is to make the whole process easier and more transparent for config file, simplifying the maintenance. A remote profile is an item stored in local profile list not bound to a local dialog. At this moment, dialog module in master branch exports a function dlg_remote_profile() that allows to add/remove a remote profile. The next step is to build the notification system between kamailio instances. There are couple of options, as well as questions, that I want to discuss before going to implement the mechanism. There are two major aspects to care of: A) Distribution, I thought of two options for it: 1) using dmq module to publish the operations done with local profiles (add/remove) -- this seems to be the natural choice right now, Charles Chance is also willing to put effort in this direction This will be coded inside the module, so config won't be affected much (eventually some extra parameters of dialog flags). 2) add event routes for adding/removing a local profile. The attributes of the profile can be made available via some PV, then in config use what so ever option (e.g., http query, json rpc, etc) for distributing the information. This will require changes in config file and might not be trivial for new users B) Persistence, with the options: 1) store remote profiles in local database. It is good for restart, but if there is big downtime, then some notifications can be lost and the list of remote profiles might be inaccurate 2) keep in memory only and at startup send a 'request-for-update' to the other instances -- these instances will send the list of their local profiles. It can take a bit between request-for-update and receiving data from the other servers -- during this interval, the checks of overall call limits might not give the desired answer Of course, there can be enhancements/combinations, but it has to be analysed if the benefit worth the complexity. For 1), store also the IP of each remote profile and then interrogate for each profile to see if its still active. For 2), add a mechanism to reject new calls until the request-for-update is completed. Anyone having other ideas or comments to the above? Note that each profile got a unique id, so if you upgrade your dialog module, profiles for existing active calls will not be loaded. Also, the code was barely tested, report here any issue you have with dialog from master branch. 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 [View Less]

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Calls to devices behind NAT not working
by Adnan 28 Aug '14

28 Aug '14

Hi all, I'm trying to configure a Kamailio server, and am having trouble getting devices behind NAT receive calls. Looking at a SIP trace, nothing is being received by the end point, even though Kamailio seems to be sending out a response. I've opened up my firewall rules, so that shouldn't be an issue. If I call another endpoint which is behind the same NAT, it works (although no audio), and looking at the trace, the SIP messaging is coming from the Kamailio server (ie. not the other … [View More]

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Minimal configuration without database support
by Eugene Prokopiev 27 Aug '14

27 Aug '14

Hi, Is it possible to run kamailio without database support (even dbtext)? Where can I find minimal configuration example with minimal modules set only for forwarding any requests/responses from one ip address to another one? -- WBR, Eugene Prokopiev

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Understanding hashtable usage
by aft 27 Aug '14

27 Aug '14

Hi, I've got a piece of config code which should solve my problem, but i'm not getting the code. It involves hashtable module. Here it goes : modparam("htable", "htable", "a=>size=8;autoexpire=400") modparam("htable", "htable", "b=>size=8;autoexpire=31") Ok we have two hashtable , a and b, both has a "size" of 8. Then, else if(status == "503") { if($sht(a=>alarm::503) == $null) $sht(a=>alarm::503)=0; $sht(a=>alarm::$si::503) = $sht(a=>alarm::$… [View More]

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match_dialog not handling BYE
by Satish Patel 26 Aug '14

26 Aug '14

I really want to use top hide using dialog module but having very difficult time.. [UA]------[Opensips]-----[Asterisk] I want to hide UA so Asterisk doesn't know its IP. I have following config, Please correct me if anything wrong in my config, Problem is if callee whoever connected to asterisk hangup call it doesn't not know where to send BYE because of top hiding remove all return routes.. i am surprise how it is working with other folks? something must be wrong in my config then.. if (… [View More]

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Not able to start tls on port 5061
by Debojit Kakoti 26 Aug '14

26 Aug '14

I am getting error INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... -- Debojit Kakoti Mobile : +91 9650654870 Email : debojkakoti88(a)gmail.com Skype : deboj1988

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Re: [SR-Users] sr-users Digest, Vol 111, Issue 21
by Rakesh Sharma 23 Aug '14

23 Aug '14

How to configure 3 kamailio on three different node, and they work as a single kamailio proxy. which module or method need to implement. On Fri, Aug 22, 2014 at 3:30 PM, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message … [View More]with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. force_send_socket in event route tm:local-request > (Kristian F. H?gh) > 2. Re: SIP Trunk (Satish Patel) > 3. Re: SIP Trunk (Yuriy Gorlichenko) > 4. Re: CNXCC curious case (Carlos Ruiz D?az) > 5. Kamailio script editor (Grant Bagdasarian) > 6. Re: Kamailio script editor (Narsay, Deep) > 7. Re: Kamailio script editor (Grant Bagdasarian) > 8. Re: Kamailio script editor (Alex Balashov) > 9. Re: force_send_socket in event route tm:local-request > (Daniel-Constantin Mierla) > 10. Re: Kamailio script editor (Daniel-Constantin Mierla) > 11. switch kamailio to domain (Mihai Marin) > 12. Re: SIP Trunk (Satish Patel) > 13. Re: Kamailio script editor (M?ns Nilsson) > 14. Re: force_send_socket in event route tm:local-request > (Kristian Frederik H?gh) > 15. How to uniquely identify SIP WS / WSS endpoint (Muhammad Shahzad) > 16. nat_traversal vs nathelper for mediaproxy (Satish Patel) > 17. Re: switch kamailio to domain (Daniel-Constantin Mierla) > 18. Re: switch kamailio to domain (Mihai Marin) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 21 Aug 2014 12:00:49 +0200 > From: Kristian F. H?gh <kfh(a)uni-tel.dk> > To: <sr-users(a)lists.sip-router.org> > Subject: [SR-Users] force_send_socket in event route tm:local-request > Message-ID: <25156162.LDl4apoH3L@kfh-laptop> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > Our kamailio 4.0.6 proxy uses pua/pua_dialoginfo and an external presence > server. > The PUBLISH requests generated by pua* appear in tm:local-request event > route. > > Before sending, we call set_advertised_address and force_send_socket. > Still the requests have the IP address from the first "listen=" as source > and Via address. > The requests are send out using the correct interface and no errors are > logged. > > When forwarding SUBSCRIBE requests received from clients, > force_send_socket before t_relay_to_udp do work as expected. > > mhomed=1 solves the issue, but I prefer force_send_socket. > > Regards, > Kristian H?gh > Uni-tel > > > > > ------------------------------ > > Message: 2 > Date: Thu, 21 Aug 2014 06:43:15 -0400 > From: Satish Patel <satish.txt(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Cc: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] SIP Trunk > Message-ID: <A3BD87DA-AE2A-4E52-B70A-031A875DF111(a)gmail.com> > Content-Type: text/plain; charset=us-ascii > > I will give it a try again today, can you please make sure my t_relay() > syntax is correct? > > So t_relay will rewrite my host past right and send call to trunk. > > While ago I was using rewritehost() function but I think it's not working > with UAC registrant module. > > > Sent from my iPhone > > On Aug 21, 2014, at 3:53 AM, Daniel Tryba <daniel(a)pocos.nl> wrote: > > > On Thursday 21 August 2014 05:56:46 Satish Patel wrote: > >> if ( $rU =~ "sip:1[0-9]@*") { > > > > Try $ru instead, $rU only contains the dialled "number". So > > $ru =~ "sip:1[0-9]@*" > > or > > $rU =~ "1[0-9]" > > > > But note the regexp, that only matches the exact numbers 10 to 19, if > you are > > trying to match prefixes you are doing it wrong (tm). > > > > _______________________________________________ > > 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 > > > > ------------------------------ > > Message: 3 > Date: Thu, 21 Aug 2014 15:37:38 +0400 > From: Yuriy Gorlichenko <ovoshlook(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] SIP Trunk > Message-ID: > <CABSP_VczfN-giH5ghuT2A9Gnqyk_+VNa= > XFtX3tAaWDqM4kZTg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > rewritehost() sucessfully work with UAC. But As I know > 1) It statless function > 2) It read only string argumetns, and do not read variables > > > 2014-08-21 14:43 GMT+04:00 Satish Patel <satish.txt(a)gmail.com>: > > > I will give it a try again today, can you please make sure my t_relay() > > syntax is correct? > > > > So t_relay will rewrite my host past right and send call to trunk. > > > > While ago I was using rewritehost() function but I think it's not working > > with UAC registrant module. > > > > > > Sent from my iPhone > > > > On Aug 21, 2014, at 3:53 AM, Daniel Tryba <daniel(a)pocos.nl> wrote: > > > > > On Thursday 21 August 2014 05:56:46 Satish Patel wrote: > > >> if ( $rU =~ "sip:1[0-9]@*") { > > > > > > Try $ru instead, $rU only contains the dialled "number". So > > > $ru =~ "sip:1[0-9]@*" > > > or > > > $rU =~ "1[0-9]" > > > > > > But note the regexp, that only matches the exact numbers 10 to 19, if > > you are > > > trying to match prefixes you are doing it wrong (tm). > > > > > > _______________________________________________ > > > 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 > > > > _______________________________________________ > > 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 > > > [View Less]

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