sr-users September 2014

sr-users@lists.kamailio.org

96 participants
170 discussions

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

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sr-users September 2014