sr-users October 2013

sr-users@lists.kamailio.org

78 participants
153 discussions

Re: [SR-Users] Kamailio network edge for registration and rtp pass-through
by JR Richardson 29 Oct '13

29 Oct '13

Date: Fri, 25 Oct 2013 17:48:13 -0400 From: Fred Posner <fred(a)palner.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: SR-Users <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Kamailio network edge for registration and rtp pass-through Message-ID: <6D231DE3-5BED-4601-A097-5A79CEF452F0(a)palner.com> Content-Type: text/plain; charset=us-ascii Do you need the registration be local to the asterisk? I would have all the asterisks send calls to the Kamailio. You can have a lookup on endpoint outbound to decide which asterisk should handle the outbound call for that did. Also a lookup for incoming DIDs, etc. ---Fred > On Oct 25, 2013, at 5:43 PM, Jr Richardson <jmr.richardson(a)gmail.com> wrote: > > Hi All, > > Starting a new project, roll your own SBC, not a full SBC, just need some minor functionality. I'm interested in deploying Kamailio as a edge device on a VSP for single entry point for hosted PBX's, Asterisk based. I had some wonderful and informative conversations at Astricon 2013, several folks assuring me Kamailio w/rtpproxy was the tool for the job, so this is a follow up to delve more into details. > > I've been researching configs, topology, modules needed, ect... Most of the examples I'm reading about for this scenario are spreading the registrations across many PBX's without distinction. One concept I'm struggling with is having a specific phone register to a specific PBX. > > phone-customer-A-x101><internet><kamailio><PBX-customer-A-x101 > phone-customer-B-x101><internet><kamailio><PBX-customer-B-x101 > phone-customer-C-x101><internet><kamailio><PBX-customer-B-x101 > > I could add a unique identifier to some part of the registration of each phone like 'custA-101@kamailio_server', custB-101@kamailio_server, ect. What I'm not clear on is when the request comes to kamailio, where would I identify what PBX the phone should register to and how to re-write the 'custA-101@kamailio_server' to '101@custA-pbx' and forward to the correct PBX and ensure rtp flows through kamailio. > > Could this function be derived using dbaliases or possibly using dispatcher with group number for each customer PBX? > > So assuming I can get the registrations to work properly, would standard invite for calling just work or would I also have to have specific config in place to ensure an invite from customer A phone also reaches the correct customer A PBX? > > A point in the right direction? > > Thanks. > > JR I would like kamailio to be transparent in the process, i.e. phones register locally to their respective PBX. Thanks. JR

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Fwd: Presence Proxy
by Jan Gaida 29 Oct '13

29 Oct '13

Hello, I'd like to explain my scenario and the module I'd like to develop. I am hoping for comments whether there are already modules like that or which modules' API I could use to make the development easier. Scenario: I have two domains (A and B) with a Kamailio server in each of them configured as presence servers. The agents from both domains want to subscribe to few resources from domain A. Idea: In order to reduce traffic between the domains, the presence server from domain B shall handle the subscriptions internally and use an internal virtual presence-user-agent to subscribe to the resources in domain A. SIP Messages: Subscription: 1) Subscribe resource.domain.a, from each user agent ua_n.domain.b 2) 202 OK from kamailio.domain.b 3) If not already subscribed, Subscribe resource.domain.a from kamailio.domain.b with virtual user-agent 4) 202 OK from kamailio.domain.a (Repeat for each subscriber) (Subscription has to be refreshed by an internal timer) Events: 1) Publish resource.domain.a to kamailio.domain.a 2) 200 OK from kamailio.domain.a 3) Notify resource.domain.a to kamailio.domain.b 4) 200 OK from kamailio.domain.b 5) Notify resource.domain.a to each user agent ua_n.domain.b 6) 200 OK from each ua_n.domain.b (Internally, Kamailio of domain B has to forward the incoming Notify to all locally subscribed user-agents) I tried using the PUA module but it does not fit all my needs. Now I think I will have to develop my own module in order to achieve everything. Any ideas? Kind regards Jan Gaida PS: Sorry for mailing to both mail-lists. But I was not sure which list was the correct for this topic. -- *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.gaida(a)grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario.

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Re: [SR-Users] Pickup group implementation in Kamailio
by Daniel-Constantin Mierla 29 Oct '13

29 Oct '13

Hello, for pickup group look at pua_/presence_dialoginfo or sca modules. Cheers, Daniel On 10/23/13 11:52 AM, John Murray wrote: > > Hi, > > I am also looking for a 'group pickup' function with Kamailio. > > Once I have the information on the call I want to get how would I > 'grab' the call. > > My calls are being processed with the tm module. > > I imagine I send an INVITE with the 'replaces' header but what > Kamailio module would process that and what would the config look like? > > Many thanks > > John > > >/ Hello,/ > > >// > > >/this is a custom implementation, only for phones that have programmable/ > > >/buttons. It is not really recommended way to do it./ > > >// > > >/Better rely on standard specs out there and use presence extensions for/ > > >/such feature -- in kamailio you would need to use modules such as > dialog,/ > > >/presence, pua, presence_dialoginfo and pua_dialoginfo./ > > >// > > >/Cheers,/ > > >/Daniel/ > > >// > > >// > > >/On 5/14/12 1:15 AM, M.C. wrote:/ > > >// > > >/Hi all,/ > > >/I want to implement a "group pickup" (asterisk like) with Kamailio and/ > > >/Snom phones./ > > >/I found in the changelog this comment:/ > > >/"- an useful application - call pickup:/ > > >/ - INVITE comes in, callee is in a pickup group/ > > >/ - store callid, cseq, etc in database or notify the other phones in/ > > >// > > >/ the group (easy to do for snom phones and extra programmable/ > > >/ buttons) via uac_req_send()/ > > >/ - send a remote control command telling which call to pickup/ > > >/ (callid, cseq) and where to redirect. In config, call/ > > >/ t_cancel_callid(). in failure route for INVITE catch the > cancelled/ > > >/ transaction (flag is set), get the new destination (e.g., > stored in/ > > >/ htable) and forward the invite there""/ > > >// > > >/regarding new functions in tmx module, but I have found no examples > in kamailio documentation./ > > >// > > >/Is this the way to implement a group pickup in kamailio?/ > > > > _______________________________________________ > 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://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

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ALIAS_DB module resolving to multiple user@domain does not work when use_domain is set to 1
by Vassilis Radis 29 Oct '13

29 Oct '13

The documentation of the append_brances(int) function in ALIAS _DB module says: "If the alias resolves to many SIP IDs, the first is replacing the R-URI, the rest are added as branches." However this only works if not using the domain for resolving, i.e. use_domain is set to 0. This happens because there is a unique constraint on the dbaliases table for the columns (alias_username, alias_domain). Would it hurt to drop the constraint so that I can have multiple branches when resolving username AND domain? Any other suggestions/comments? Thanks.

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How to add debugger module in kamailio.cfg
by Owais ul Haq 29 Oct '13

29 Oct '13

Hello, I want to add "dubugger module" in kamailio. As I want to know which part of kamailio.cfg is responsible for reply of some of incoming sip messages. Can some one please help me in this regard. Regards, Kamailio kid.

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Help with 200k responses to the contact header not the record-route
by anfecora 29 Oct '13

29 Oct '13

Hi all, can anyone help me to find out what is wrong with my setup, i have an asterisk behind a kamailio, kamailio is proxying all packages to the outside. when the call is bridge it gets disconnected after a few seconds, it seems that our voip carrier is sending a bye because we didn't answer to their 200 ok properly, but as the trace shows we did only that kamailio is answering to the contact header ip not the ip that is sending the ok. I am sorry i sent a too long message before i will try skim it a bit. any help is appreciated . thanks. my setup request_route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); } if(method=="BYE") { #Account BYE transactions }; if (method=="CANCEL") { if (t_check_trans()) t_relay(); exit; }; if (loose_route()) { t_relay(); exit; } if (is_method("INVITE")) { record_route(); } f (!t_relay_to_udp("3.1.1.1", "5060")) { sl_reply_error(); exit; }; exit }; here is a trace to a call made to a hotel. i had changed the real ips for obvious reasons. thanks. asterisk ip 1.1.1.1 kamailio internal 1.1.1.2 kamailio external 2.0.0.1 Voip Carrier 3.1.1.1 voip contact ip 3.1.1.2 U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0.

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Regarding Exporting a function
by premchandiran.marimuthu 29 Oct '13

29 Oct '13

Hi All, I was able to export a function to cfg file with int return type but when i try to export a function with char/char-ptr as return type , in cfg file i was not able to print/use the characters which i returned in the exported function. can i get a clue/tip to overcome above said issue?. Regards, Prem Chandiran M -- Open WebMail Project (http://openwebmail.org)

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Presence issues
by Wingsravi R 29 Oct '13

29 Oct '13

Hi kamailio'ns I am working on Kamailio V 4.0.3, installed from Ubuntu 12.04(precise) repository. My SIP clients(jitsi) are registering to kamailio sever successfully but their presence is not sharing between the clients. I am frequently logging following presence related errors in syslog while trying to register SIP clients to kamailio server: Oct 25 20:01:05 kamailio /usr/sbin/kamailio[15274]: INFO: pua_usrloc [ul_publish.c:221]: ul_publish(): not marked for publish Oct 25 20:01:05 kamailio /usr/sbin/kamailio[15273]: WARNING: presence [publish.c:481]: handle_publish(): Missing or unsupported event header field value Oct 25 20:01:05 kamailio /usr/sbin/kamailio[15273]: ERROR: presence [publish.c:484]: handle_publish(): event=[reg] what could be the wrong ? How can i reslove this issue ? please find in the attachment is my kamailio config file. (suggest me if any changes to be made to this configuration) Any help will greatly appreciate. Regards, Ravi

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kamailio 3.3.2 db_mysql.so problem after mysql upgrade to 5.6
by Uri Shacked 28 Oct '13

28 Oct '13

Hi, I upgraded mysql from 5.5 to 5.6. After the upgrade kamailio fails to start and throws the following error: "..... /usr/lib64/libmysqlclient.so.18: version `libmysqlclient_16` not found....." Any ideas what to do? Thanks, Uri

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media between 2 clients behind separate NAT
by Vassilis Radis 28 Oct '13

28 Oct '13

Hello, When I have 2 clients using a kamailio proxy, and both of the clients are behind their own NAT, then my only options for relaying media between them is using some kind of intermediate rtp proxy or STUN etc? Proxy is on a public IP and essentially I am asking if I can avoid setting up stuff to proxy the media or mess with STUN, by using some kind of request response mangling. i.e. have both clients's sdp address/port changed to their public facing ip/port and then each client can send an initial packet to the other end. Those initial packets will be blocked by the other receiver router, but they will open a nat hole so that next package from the peer will pass through. Is that feasible? Thanks.

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sr-users October 2013