- sr-users - kamailio.org

Shared memory
by Morten Tryfoss 04 Sep '14

04 Sep '14

Hi, I wonder about the difference between “used” and “real used” shared memory in Kamailio. Best regards, Morten Tryfoss

2 1

How to determine if a 4xx message came from dispatcher or a client?
by Olli Heiskanen 04 Sep '14

04 Sep '14

Hello, A question on Kamailio variables and using dispatcher: When in failure_route I want to know if the request message was going to a dispatcher ip or a sip client ip (as in any other than dispatcher ip), how do I make an if statement for that? If I use ds_is_from_list(), I get wrong results because in both cases the request was sent by Kamailio. Also, I don't seem to be able to get the source ip of the 4xx message (src_ip shows the source ip of the request message). cheers, Olli

2 1

Can log in from SIP
by Alexis Walhovd 04 Sep '14

04 Sep '14

Hi I just created an account with iptel.org. But I can not log in to my account through my SIP software from OSX. Please see below for details: I type “Alexis82” as password. All the best, Alexis

2 1

need help - Insert_hf when Route: missing
by Satish Patel 04 Sep '14

04 Sep '14

I have post many question on topology hiding any not get any reply back from people and developers now I don't have any option except some goofy hack When I use topology hiding it removes Route: and because of that callee doesn't able to send BYE back to opensips. I want use insert_hf to inject Route: so I get clean BYE message. Do you guys have any suggestion? Sent from my iPhone

2 1

Asterisk cluster behind kamailio natted to pubic IP, presenting internal ip addresses in From tag
by Tim Chubb 04 Sep '14

04 Sep '14

Hi All I'm scratching my head on this one, I have set everything up in the following configuration: Public ip natted to DMZ via Cisco ASA 5510 <= DMZ => Kamailio & RTP Proxy <= Voice DMZ => Asterisk My kamailio configuration is based heavily on the one found here: http://saevolgo.blogspot.co.uk/2013/08/rtpproxy-revisited-kamailio-40.html & http://saevolgo.blogspot.co.uk/2011/11/how-to-increasing-voip-services.html So far everything is working fine, I can register via the public IP address, IM & presence is working, and as does audio, however when I dial an extension the caller id comes up like this 12345@172.16.15.123:5080<mailto:12345@172.16.15.123:5080> which is the ip addresses of the asterisk server that the dispatcher has assigned to the call, I have tried setting the P-Asserted-Identity, P-Preferred-Identity & Remote-Party-ID headers to no effect. What I would like to achieve is that the public IP or domain name comes up when I call an extension. From examining the sip traffic traversing the kamailio box, it seems that the From and Contact headers sent by asterisk are the source of the internal implementation information reaching the end-user. How can I prevent implementation information like this leaking out? I consider this to be a security risk as I don't want anyone to be able to enumerate the number of servers that comprise a given cluster as easily as placing a few calls, ideally I only want data relating to the public side of the network getting out. But I would be happy if it takes digging into the sip traffic to glean any details initially. I'm getting the feeling that this could well be an asterisk configuration issue, so apologies if this is a little off topic, however I have been unable to find a similar configuration to try and achieve my desired results. Tim.

3 3

Support for TLS server_name extension (aka SNI=server name indication)
by Barry Flanagan 04 Sep '14

04 Sep '14

Hi, Back in Kamailio 1.5, the release notes state: "support for TLS server_name extension (aka SNI=server name indication)" However, I cannot find any indication of this in the current TLS docs, and trying to set tls_server_name or server_name in tls.cfg fails with "unsupported option". Is this actually supported? Thanks. -Barry Flanagan

4 6

Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)
by Alex Villacís Lasso 04 Sep '14

04 Sep '14

Please consider the following SIP packet exchange, as seen by a tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170: IP 198.58.101.75.5060 > 201.234.196.170.5060 INVITE sip:*43@201.234.196.170:5060 SIP/2.0 Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport Max-Forwards: 70 From: "9002" <sip:9002@198.58.101.75>;tag=as0bc522a9 To: <sip:*43@201.234.196.170:5060> Contact: <sip:9002@198.58.101.75:5060> Call-ID: 2c14c21f5052a74a78ca4ab736657b00@198.58.101.75:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.20.0) Date: Fri, 29 Aug 2014 18:23:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 299 v=0 o=root 521741684 521741684 IN IP4 198.58.101.75 s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 c=IN IP4 198.58.101.75 t=0 0 m=audio 16426 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv IP 201.234.196.170.5060 > 198.58.101.75.5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060 From: "9002" <sip:9002@198.58.101.75>;tag=as0bc522a9 To: <sip:*43@201.234.196.170:5060> Call-ID: 2c14c21f5052a74a78ca4ab736657b00@198.58.101.75:5060 CSeq: 102 INVITE Server: kamailio (4.1.5 (x86_64/linux)) Content-Length: 0 IP 201.234.196.170.5060 > 198.58.101.75.5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes> From: "9002" <sip:9002@198.58.101.75>;tag=as0bc522a9 To: <sip:*43@201.234.196.170:5060>;tag=as2798a3b9 Call-ID: 2c14c21f5052a74a78ca4ab736657b00@198.58.101.75:5060 CSeq: 102 INVITE Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:*43@127.0.0.1:5080;alias=127.0.0.1~5080~1> Content-Type: application/sdp Require: timer Content-Length: 305 v=0 o=root 159029581 159029581 IN IP4 201.234.196.170 s=Asterisk PBX 11.12.0 c=IN IP4 201.234.196.170 t=0 0 m=audio 18446 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes According to a strict interpretation of the SIP RFC, which address should the machine at 198.58.101.75 use to send the subsequent ACK? Which field(s) are to be used to extract said address? I am trying to understand an issue of a missing ACK between 201.234.196.17x and a different public IP, with the only difference that the other IP is not running Asterisk. For the exchange shown above, 201.234.196.170 receives an ACK, but I want to know whether the packets correctly indicate the address for the ACK, or whether the Asterisk at 198.58.101.75 is compensating for a malformed packet.

3 7

want to know about best book or best way to learn kamailio.
by Talha Bukhari 04 Sep '14

04 Sep '14

please guide me, how i learn kamalio as i am new to kamailio. suggest me some book or resources. waiting.

2 1

Re: [SR-Users] How to add route header (Senthil K)
by Surendra 03 Sep '14

03 Sep '14

Hi Senthil, You can use record_route function to add Record_route header in INVITE message( use rr.so module). This record_route header will become Route header while sending BYE message from client. Suppose if you want add directly Route header, then you can use Insert_hf function. Please go thru the textops modules for the same. Regards surendra

1 0

How to uniquely identify SIP WS / WSS endpoint
by Muhammad Shahzad 03 Sep '14

03 Sep '14

Sorry for putting this question on both dev and user mailing lists, as it is a rather theoretical question and i hope some SIP guru on either mail list will answer. For non-WS endpoints which use TCP or UDP for SIP transport, each upstream request has top most VIA header pointing to the previous hop which forwarded the request to current hop while each downstream request has top most VIA header pointing to next hop to which it will be forwarded from current hop. But for WS endpoints, the top most VIA has dummy static value, so there is no way to identify who sent this request or to whom the reply is going to. Please note that i am not specifically interested in network address of remote endpoint (though VIA header is suppose to provide it), i only need to match requests and responses from / to a specific device using SIP v2.0 standard. Any help is highly appreciated. Thank you.

3 7

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-SIP Now I want Kamailio to connect to local fritz.box (by SIP) with one fritzbox-user to handle all invites and calls. ------------------------------- Scenario (fictional): Fritzbox (Router and VOIP-Server) Internal-IP: 192.168.178.1 or fritz.box External-IP: dynip.de VOIP-User: 610 VOIP-Passwort: test Kamailio (running on RaspberryPI and Siremis) Internal-IP: 192.168.178.2 Network Gateway: 192.168.178.1 Connection from Outside (NAT): dynip.de:5061 User: euser Pass: epass ------------------------------- Questions: Is there a Module i could use to do this Job? (configurable through Siremis e.g. Dispatcher List?) How can i ensure that my user "euser" from subsriber-list connects with 610:test to fritz.box? Is there a simple configuration to solve this problem? I'm sorry to ask you these my non-skilled questions, since it is not my business to use SIP, but this scenario could be a useful tutorial for many many fritz.box users over the world. Roaming prices for mobile-calls could be safed with sitting into next McDonalds and do a SIP-Call through free Wifi. Thank you very much! I would be very grateful to get some help. Sincerly yours Lukatz

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 packet first) is able to register successfully on the server and all the other users after the first one are not able to register on the server. Am I missing something in the configuration file such that this is happening. Please help in this regard. Waiting in anticipation Thanks & Best Regards Saurabh Srivastava

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 proposal and we can decide together which one fits most of us. This one gives about 2 weeks and a half for new features development. Therefore, if you plan to include a new module (many were announced at the IRC devel meeting, hopefully some are ready), hurry up. It is also a good time to start filling the wiki page with what is new in upcoming release: - http://www.kamailio.org/wiki/features/new-in-devel 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 1

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

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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 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?

2 5

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 pseudovariable, both within quotes or not, or even a static string. But when I look at those messages sent to the RTP proxy, there's always the IP from where the SIP packet was received (the loadbalancer IP) in the request. Previously we used the "r" parameter, sending the original IP from the SDP to the proxy. But due to strange behavior of some UACs, we want to get rid of that. Do I misunderstand the second parameter? This is what the documentation says: * ip_address* - new SDP IP address. Unfortunately, I couldn't find any example using this. Best Regards, Sebastian

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