sr-users May 2014

sr-users@lists.kamailio.org

96 participants
156 discussions

[sr-user] textops module - search_append_body(re, txt) - append new line
by Cockhootec Yahrabee 03 May '14

03 May '14

Hi, I am trying to append new field in SDP and I need to append new line but I don't know how. I tried to google for solution but I didn't find anything. I want to route my RTP and RTCP packet to port 5060 (this is for development of my school project, I know that in the context of correct SIP this has no sense but this is only experimental). My sample kamailio.cfg code: if (is_method("INVITE")) { … [View More]

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Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available
by Alex Villacís Lasso 02 May '14

02 May '14

I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration (attached) was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been heavily modified. Currently asterisk runs on localhost and only listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to come from localhost, from the point of view of asterisk. Currently I have a model on which … [View More]internal SIP phones get identified by the authentication username, and then the contact names at From: and To: get massaged to incorporate the SIP domain, in order to emulate multiple-domain support. The 'sip' table in Asterisk defines all such contacts as SIP accounts of the form name_domain.com, and the SIP phones are configured to use 'name' as authentication username for domain 'domain.com'. However, SIP providers that register on the server with authentication names are left with their original names, since in the model, SIP trunks are available to all domains. Now I have to add support for SIP providers which are to be authorized on the basis of IP only. Apparently, the permissions.so (WITH_IPAUTH) is made for just this purpose, so I enabled it. After authentication, I need to route the INVITE to asterisk, and asterisk must somehow match the account for the SIP trunk from the available information on the INVITE request. A typical INVITE for this scenario looks like this, before being processed by kamailio: INVITE sip:6008010@172.28.161.218:5060;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.25.144.58:5060;branch=z9hG4bK+676ea13f680e853fd847230512a347561+32e3da76+1 Call-ID: FBE75B3A@32e3da76 From: <sip:042294440@200.25.144.58:5060;user=phone>;tag=32e3da76+1+544c000c+52be771c To: <sip:6008010@172.28.161.218:5060;user=phone> CSeq: 975469826 INVITE Expires: 180 Organization: SetelGYE Min-SE: 90 Session-Expires: 18000 Supported: replaces, 100rel, timer Contact: <sip:042294440@200.25.144.58:5060;transport=udp;user=phone> Content-Length: 149 Content-Type: application/sdp Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, PRACK, UPDATE, INFO, REFER v=0 o=- 0 0 IN IP4 201.217.79.3 s=- c=IN IP4 201.217.79.3 t=0 0 m=audio 5388 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Here, 6008010 is the phone number that was dialed at the provider in order to reach my system, and 042294440 is the provider-supplied Caller-ID, which I want to preserve all the way to Asterisk. In particular, 042294440 appears as the value that ends up as $fU (From: username) while being processed in kamailio. If I pass the SIP packet as-is to asterisk, asterisk first tries to match by the value of $fU, which obviously fails to match the trunk name. It then tries to match by incoming IP, which also fails because asterisk received this packet from 127.0.0.1 . Finally, asterisk sort of matches to the first record in the sip table, which is *not* the SIP account for this trunk, but some other random account. I have a partial solution that uses sqlops to make a query to the sip table, using the $si (source IP) and reads the trunk name in order to replace $fU. This works, as now $fU will have the trunk name and asterisk will now recognize the intended SIP account for the trunk. However, this has the unfortunate side effect of throwing out the Caller-ID information. What is the standard/proper way to deal with this situation? Is there a well-known way to make Asterisk match the trunk name, without overwriting the Caller-ID information? Before you ask, requesting the provider to modify its INVITEs is not an option. I believe there is a standard way to deal with this, since this scenario should also arise with a kamailio that faces the internet, and relays INVITEs (after authentication) to an asterisk in a local network. As far as I can tell, the fact that in my case the 'local network' is localhost should be irrelevant. [View Less]

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Kamailio v4.0.5 crash - transformations related ?
by Dragos Oancea 02 May '14

02 May '14

Hi We experimented a crash with kamailio 4.0.5 , it looks like a memory corruption. After an analyse of the core file, it appears that it crashed while doing a str2int transformation (trying to convert the value of myvar to int): if (($(dlg_var("myvar"){s.int}) == 0) && some_other_condition ) { do_something(); } gdb output: (gdb) frame 4 #4 0x00000000004bcb77 in rval_get_btype (h=0x7fffd69713d0, msg=0x7ffc8a58a4d0, rv=0x7ffc8a3dfc18, … [View More]

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Frequent Crash in Kamailio 4.1.3 in the file data_lump.c see gdb backtrace in mail
by varun pratapsingh 02 May '14

02 May '14

HI All/ Daniel, According to the recommendation I have migrated my Kamailio 4.1.2 to the Kamailio 4.1.3. But very frequent crashes are coming in Kamailio 4.1.3 which was not coming on the 4.1.2. The gdb backtrace of the crash is below: I have seen many crashes in couple of days but the "data_lump.c: No such file or directory" this line is coming in every crash. I think every time it crashes in the same file and same function somewhere else. GDB Backtrace: Program terminated with signal 11, … [View More]

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Kamalio
by Akash Ashok 02 May '14

02 May '14

Hi, I have just started using Kamalio and We are using it for a VOIP app that I am trying to build. Here are the things that i have done 1. I have setup kamalio server 2. I could get it running using linphone app 3. I could add users using kamctl command Since I am building a smartphone app I want to register users remotely from the phone. I did check up the XML RPC documentation and it doesn't talk about apis to add users and a lot of other functionalities that are present on kamctl … [View More]

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What does this script do?.
by VOIP Tests 02 May '14

02 May '14

Can someone let me know what this script does? avp_db_load("$ru/domain","i:/domain_preferences"); Thank you, Arun

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Limit the number of requests per second based on the user
by Sebastian Damm 02 May '14

02 May '14

Hi, I'm looking for a solution to limit the number of call initiations or OPTIONS packets or REGISTER packets per user. Neither pike, nor pipelimit or ratelimit give me the possibility to do this. Limiting it based on the source IP is not an option in our case. Is there a module I can use? Or is this generally a bad idea? Best Regards, Sebastian <http://www.sipgate.co.uk>

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Dispatcher ping source ip:port?
by Daniel Tryba 02 May '14

02 May '14

Is it possible to manipulate the pings (OPTIONS in my case) to force a source ip:port combo per destination? dispartcher.list: 1 sip:192.168.0.1:5060 2 sip:192.168.1.1:5070 3 sip:192.168.2.1:5050 The kamailio (4.1) is listening on 192.168.3.1 5060 and 5070. dispatcher 2 cares about the source port, which should be 5070. 1 and 3 don't care. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

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Re: [SR-Users] Voice is breaking when using Kamailio 4.1.3
by varun pratapsingh 02 May '14

02 May '14

Hi Daniel, Yes I agree with you that kamailio is not involved in the rtp handling and forwarding but RTP Proxy does this. So I have mentioned that when my call is established then my RTP passes through the RTPProxy. The issue is that RTP is coming proper at UAC Side so no Voice Problem but may be RTP is not coming good at Phone Side that is why voice is bad at Phone SIde. Please see again my network. SIP UAC--------->Kamailio+RTPProxy------>SIP Server------->Phone And Kamailio and … [View More]

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How can I implement Asterisk as Media server?
by Patrik Kristel 02 May '14

02 May '14

Hello, I have Kamailio with websocket module and I want to connect Asterisk as media server. I'm trying to route calls from web JsSIP users to non-web users I would like to ask how can I implement it? Can i use the rtpproxy-ng Module and here setup IP of Asterisk? Or is there any other way to do it? Thank you for your help! Regards, Patrik Kristel

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sr-users May 2014