sr-users September 2013

sr-users@lists.kamailio.org

89 participants
128 discussions

onreply_route check_route_param?
by Gareth Rylance 25 Sep '13

25 Sep '13

The check_route_param docs say that you must call loose_route() first. loose_route() cane called from not within onreply_route. How do I check form route params within the onreply_route? Thanks Gareth

2 1

OT - Consultation
by Gilbert T. Gutierrez, Jr. 25 Sep '13

25 Sep '13

I looking to hire a consultant to build a new multi-tenant solution for my organization. I do not have the time to do it myself nor the staff. If you are a consultant interested in more details, please contact me off list. Sorry to post an off topic email to this list, I could not find a Kamailio list for this type of post. Gilbert T. Gutierrez, Jr. Operations Manager Phoenix Internet Gilbert.G(a)phoenixinternet.net

2 1

Decoding HTTP URLs in event_route[xhttp:request]
by Peter Dunkley 25 Sep '13

25 Sep '13

Hello, I was wondering if there was an easy way to decode HTTP URLs in event_route[xhttp:request]? For example, it would be good to be able to breakdown a URL like: /sip?apiKey=abcdefg&username=1234567890:alice@example.com into "path/on/server" ("/sip" in this case) and a set of parameters. For the parameters something like the {param.value,name} transformation for SIP header parameters would be ideal (which works perfectly for picking values out of HTTP Cookie: headers). I noticed that there is already an {s.urldecode.param} transformation in the PV module but I couldn't find any documentation for it in the wiki and looking at the code it doesn't appear to do this anyway. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

2 5

Record routing transport=tcp and examples/websocket.cfg
by Mikko Lehto 24 Sep '13

24 Sep '13

Hi websocketeers, examples/websocket.cfg starts with this check: --- request_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) || $Rp == MY_MSRP_PORT) { xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } --- My in-dialog SIP over TCP requests got 403 treatment because of this. I believe reason was MY_WS_ADDR in route set, added by record_route(). Above sanity check seems sensible tough, so I fixed advertised route set with call to force_send_socket() as seen on attached patch. I wonder if my fix is the best approach. Should the remaining other two t_relay() calls also be prepared with force_send_socket()? -- Mikko

2 1

unregister user when kamailio looses TCP connection.
by Vitaliy Aleksandrov 24 Sep '13

24 Sep '13

Hello, I've made a patch to kamailio-4.0.3 which removes stale registration when kamailio looses an incoming tcp connection. Of course this patch needs more work. Since the are no direct references between user location contacts and tcp connections callback function uses linear search through the whole location table using received field as a key. Can anybody more experienced in kamailio internals check if I chose the right place to get information about lost tcp connections ? Another thing I wanted to ask is maybe somebody can suggest a better way to tie a tcp connection to the user location information without complicating "usrloc" module by any heavy data structures. If anybody else except me need this It would be great to fix known problems and add it to kamailio.

7 28

unsubscribe
by Andres Paglayan 23 Sep '13

23 Sep '13

2 1

new module: gzcompress
by Daniel-Constantin Mierla 23 Sep '13

23 Sep '13

Hello, following the Kristian's email and blog about Apple Facetime using compressions of the body (http://lists.sip-router.org/pipermail/sr-users/2013-September/079816.html), I did a quick coding Friday evening and put together a new module that uses zlib to compress/decompress the body for SIP messages - I named it gzcompress, its readme is available at: - http://kamailio.org/docs/modules/devel/modules/gzcompress.html Might be useful for those that want to save bandwidth when peering two kamailios or target to get under MTU size. It is now enabled for SIP traffic only, but the plan is to get it for http as well, where it can be more relevant for xcap traffic. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013 - more details about Kamailio trainings at http://www.asipto.com -

4 5

Problem with to: header
by julian arsanches 23 Sep '13

23 Sep '13

Can someone advise me on how to change the to header to show the host that we are sending the call to an not the servers ip. I am using dispatcher on my setup . i am getting this U 2013/09/23 12:57:54.576312 10.0.1.206:5060 -> 2.2.2.2:5060 INVITE sip:+42123333235@2.2.2.2:5060 SIP/2.0. Record-Route: <sip:1.1.1.1;lr=on;ftag=as1f3df8b3>. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bKb29d.7399c2b3.0. Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK42a1ecaf;rport=5060. Max-Forwards: 16. From:<sip:unavailable@1.1.1.1>. To: <sip:+4212333323@1.1.1.1>. Contact: <sip:anonymous@1.1.1.5:5060>. Call-ID: 0df8db614d45bae27035443c35166ba6@1.1.1.5:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Mon, 23 Sep 2013 16:58:25 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Cisco-Guid: 7128745-3588944267-852064@msc1 Content-Type: application/sdp. Content-Length: 288. . v=0. o=root 1760548326 1760548326 IN IP4 54.236.97.30. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.5. t=0 0. m=audio 39794 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. I need to sent header to a carrier like this To: <sip:+4212333323@2.2.2.2> instead of 1.1.1.1. i am proxing calls from asterisk to a main carrier. Please help. here is my config. if (starts_with("$var(o)","anonymous")) { ds_select_domain("$var(z)", "4");#carrier dynamic xlog("here is anonymous call <$var(o)>77777\n"); $var(n)=$(tU{s.substr,3,0}); remove_hf("From"); remove_hf("P-Asserted-Identity"); remove_hf("Privacy"); insert_hf("From:<sip:unavailable@1.1.1.1.1>\r\n", "From"); $tU=$var(n); xlog("out header CHECK ANONYMOUS BEFORE to $tu--$td - contact pai+++ <<$ct>>++ from_uri=$fu;<$tU---=$var(n)> to_uri=$tu; }pai<$ai>intid=$fU; type_call=$si; dst_ip=$ru; carriercode=$var(z);callmode=$var(out)"); if(!t_relay()){; sl_reply_error(); exit; }; ##ENDANONYMOUS exit; } Again thanks a lot for any help.

2 1

Caller ID using From header
by Daniel W. Graham 21 Sep '13

21 Sep '13

Wondering what others are doing / looking for any tips for the best handling of caller ID in this scenario. Call flow - UA >> Proxy >> Asterisk >> Proxy >> Carrier Carrier only reads caller ID by 'From header' Asterisk is only used for media functions / b2bua and not user aware. UA username is not a DID but rather some other string, such as 'dan(a)domain.com'. Goal is to use DID for caller ID, but I like to avoid modifying the 'From header'. Any thoughts? [cid:image001.gif@01CEB53F.854769A0]

3 3

Stupid NAT tricks to learn from Apple
by Kristian Kielhofner 20 Sep '13

20 Sep '13

I've been spending some time looking at some of the significant changes Apple has made to Facetime in iOS 7. I'm far from an Apple fanboy but some of them are pretty interesting: - multiplexing everything over a single UDP port - deflate compression with SIP - various /slight/ protocol violations ;) More here: http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html As SDP bodies swell more and more can we hope to build significant support for multiplexing and deflate compression in the SIP-focused open source ecosystem? -- Kristian Kielhofner

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sr-users September 2013