- sr-users - kamailio.org

Difficulties with with loose_route() when trying to handle non-compliant ACK
by Richard Brady 24 Nov '12

24 Nov '12

Hi guys I have a multihomed Kamailio proxy sitting between two B2BUAs on separate networks and record-routing all dialogs. The problem I have is that when one of these devices receives a 200 OK, it does not populate the RURI of the ACK correctly. Instead of taking it from the Contact header on the 200 OK, it uses the user part from the Contact header and sets the domain to the proxy IP. It then also populates the Route headers. The ACKs are below and the IPs are: 10.152.1.92:5060: UAC on outside 10.200.70.100:5060: proxy outside interface 192.168.242.100: proxy inside interface 192.168.242.102: UAS on inside This the ACK message going into and coming out of the proxy. # U 10.152.1.92:5060 -> 10.200.70.100:5060 ACK sip:natted_ua*9197**192.168.242.102*5080*udp@10.200.70.100 SIP/2.0. Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK-2280-1-5. Route: <sip:10.200.70.100;r2=on;lr=on;nat=yes>,<sip:192.168.242.100;r2=on;lr=on;nat=yes>. From: sipp <sip:sipp@127.0.1.1:5060>;tag=2280SIPpTag001. To: sut <sip:9197@10.200.70.100:5060>;tag=p0FaeQ1QUS9ae. Call-ID: 1-2280(a)127.0.1.1. CSeq: 1 ACK. Contact: sip:sipp@127.0.1.1:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . # U 10.200.70.100:5060 -> 192.168.242.100:5060 ACK sip:192.168.242.100;r2=on;lr=on;nat=yes SIP/2.0. Via: SIP/2.0/UDP 10.200.70.100;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=10.152.1.92;branch=z9hG4bK-2280-1-5. Route: . From: sipp <sip:sipp@127.0.1.1:5060>;tag=2280SIPpTag001. To: sut <sip:9197@10.200.70.100:5060>;tag=p0FaeQ1QUS9ae. Call-ID: 1-2280(a)127.0.1.1. CSeq: 1 ACK. Contact: <sip:sipp@127.0.1.1:5060;alias=10.152.1.92~5060~1>. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. So I am trying to understand why it is trying to relay to itself (I have mhomed=1) and why it is rewriting the RURI as if it is a strict router. Currently decode_contact() is disabled but enabling doesn't seem to help. Unless there is a very specific place where it belongs. Any advice would be hugely appreciated. I can always paste logs / configs / traces. Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady(a)gmail.com

2 3

can not make calls with kamailio 3.3 and asterisk-10.7.0
by James zhu 24 Nov '12

24 Nov '12

hello, guys:i follow the linkhttp://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-a… can registered users, kamctl online shows those status. but i can not make calls. this is debug message when i make a call from 102 to SIP 103--------------------------------------------------------------- 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=486 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=3 n=return 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=489 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=636 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=601 a=25 n=has_totag 8(1286) DEBUG: siputils [checks.c:103]: no totag 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=501 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=26 n=is_method 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=496 a=25 n=t_check_trans 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [tm.c:1015]: lookup_original: t_lookupOriginalT returned: 0xb4d36320 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=497 a=25 n=t_relay 8(1286) DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=29 , global msg id=28 , T on entrance=(nil) 8(1286) DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=60763, isACK=0 8(1286) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 8(1286) DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found 8(1286) DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb4d3a698, callback type 1, id 0 entered 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [t_fwd.c:1243]: DEBUG: e2e_cancel: e2e cancel proceeding 8(1286) DEBUG: <core> [msg_translator.c:206]: check_via_address(192.168.1.103, 192.168.1.103, 0) 8(1286) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 8(1286) DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 8(1286) DEBUG: tm [t_reply.c:703]: DEBUG: reply sent out. buf=0xb721ee38: SIP/2.0 200 cancelin..., shmem=0xb4d38c08: SIP/2.0 200 cancelin 8(1286) DEBUG: tm [t_reply.c:713]: DEBUG: _reply_light: finished 8(1286) DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=3 n=exit 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list=================================anyone can give a hint for that? thanks! Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com

1 0

Using uac_req_send function
by Freddy C 22 Nov '12

22 Nov '12

Hello, I'm using uac_req_send () to send a request generated in Kamailio (eg Example 1.21. UAC module) $ uac_req (method) = "OPTIONS"; $ uac_req (ruri) = "sip: kamailio.org"; $ uac_req (furi) = "sip: kamailio.org"; $ uac_req (turi) = "sip: kamailio.org"; uac_req_send (); However, I need to capture the responses of this request with failure_route TM module. Is there a way to capture these responses when I'm using uac_req_send() function? I appreciate your comments. Freddy

2 1

Next IRC devel meeting
by Daniel-Constantin Mierla 22 Nov '12

22 Nov '12

Hello, I am thinking of having our next IRC devel meeting soon, to plan the next major release and review current stable releases and the environment around the project (e.g., if you can add anything else to the project to make life easier for developers and users). I'm proposing next week, Thursday, at 15:00GMT, on Freenode IRC server, channel #sip-router. Anyone able to join? If it is not a convenient date for you, just propose alternatives and we will select the one that meets the constraints of the most developers and users willing to participate. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 10

How does SIP transfer work?
by Grant Bagdasarian 22 Nov '12

22 Nov '12

Hello, I've been searching the internet to find an explanation on how SIP transfer works using Re-INVITE and/or UPDATE, but I can't seem to find a good source. >From what I understand(and this is the way we do it), the following happens: Bob=Caller Alice=Called John=Transfer party 1) Bob calls Alice. The usual INVITE,Trying,200 OK, ACK. 2) Alice transfers the call to John using Re-INVITE. a. Alice calls John. The usual INVITE,Trying,200 OK, ACK. b. Alice Re-INVITEs Bob using INVITE with adjusted SDP. 3) Bob is connected to John through Alice in some magical way. I'm guessing because the SDP has been changed and for some reason the RTP stream flows between Bob and John through Alice? Is this correct? If not, perhaps someone could explain it to me from scratch. Maybe useful to know that we are using Cisco equipment for call handling (VXML and TCL scripts). Thanks, Grant

3 2

Forward Register Message
by kellon＠sis-grenada.com 21 Nov '12

21 Nov '12

Hi All Can anyone point to documentation tutorial on howto forward statefully a REGISTER message to another SIP Server. I have looked at http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, However I'm getting errors with the uac_req commands even though I have loaded the uac.so module. .

2 1

Shared Call Appearances module
by Andrew Mortensen 21 Nov '12

21 Nov '12

I've been working on a Shared Call Appearances module for the past several months. It implements the Broadsoft SCA feature as laid out in Broadworks SIP Access Side Extensions Interface Specifications document. (Another implementation of the same feature was added to Freeswitch a few years ago, but we didn't want to use Freeswitch.) We've been testing and improving the module over the last month, and I think it's ready to share. The module is available at GitHub here: <https://github.com/fitterhappier/sca> To date, we've only tested the module with Polycom handsets. I'm happy to answer questions. I hope this will prove useful for others. Best, andrew

6 18

cost script
by Ruben Mejia 21 Nov '12

21 Nov '12

IPTEL We needhelp or cost for the development a script for kamailio which can do the following: Register kamailio to another gateway using UAC modules. route all traffic coming towards a private ip 10.x.x.x We will use a postgres database We have installed kamailio 3.1 with postgres support and UAC modules Rubén Mejía Telefonia Cabletamps

1 0

I can not make call from kamailio to asterisk by 3.3
by James zhu 21 Nov '12

21 Nov '12

hello, guys:i follow the linkhttp://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-a… can registered users, kamctl online shows those status. but i can not make calls. this is debug message when i make a call from 102 to SIP 103--------------------------------------------------------------- 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=486 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=3 n=return 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=489 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=636 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=601 a=25 n=has_totag 8(1286) DEBUG: siputils [checks.c:103]: no totag 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=501 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=26 n=is_method 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=496 a=25 n=t_check_trans 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [tm.c:1015]: lookup_original: t_lookupOriginalT returned: 0xb4d36320 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=497 a=25 n=t_relay 8(1286) DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=29 , global msg id=28 , T on entrance=(nil) 8(1286) DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=60763, isACK=0 8(1286) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 8(1286) DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found 8(1286) DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb4d3a698, callback type 1, id 0 entered 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [t_fwd.c:1243]: DEBUG: e2e_cancel: e2e cancel proceeding 8(1286) DEBUG: <core> [msg_translator.c:206]: check_via_address(192.168.1.103, 192.168.1.103, 0) 8(1286) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 8(1286) DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 8(1286) DEBUG: tm [t_reply.c:703]: DEBUG: reply sent out. buf=0xb721ee38: SIP/2.0 200 cancelin..., shmem=0xb4d38c08: SIP/2.0 200 cancelin 8(1286) DEBUG: tm [t_reply.c:713]: DEBUG: _reply_light: finished 8(1286) DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=3 n=exit 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list=================================anyone can give a hint for that? thanks! Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com > From: sr-users-request(a)lists.sip-router.org > Subject: sr-users Digest, Vol 90, Issue 72 > To: sr-users(a)lists.sip-router.org > Date: Wed, 21 Nov 2012 08:58:40 +0100 > > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Shared Call Appearances module (Daniel-Constantin Mierla) > 2. Re: Shared Call Appearances module (Andrew Mortensen) > 3. Re: LCR weight (Juha Heinanen) > 4. problem with $time Pseudo-Variables on 3.3.2 (Uri Shacked) > 5. Re: Dialog module not decrementing profile size > (Grant Bagdasarian) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 20 Nov 2012 22:56:10 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Shared Call Appearances module > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <50ABFC7A.2070602(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > On 11/20/12 10:41 PM, Ovidiu Sas wrote: > > On Tue, Nov 20, 2012 at 4:22 PM, Andrew Mortensen > > <admorten(a)isc.upenn.edu> wrote: > >> On Nov 20, 2012, at 3:16 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > >> > >>> On 11/20/12 8:43 PM, Andrew Mortensen wrote: > >>>> On Nov 20, 2012, at 4:08 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > >>>> > >>>>> Hello, > >>>>> > >>>>> thank for contribution! > >>>>> > >>>>> On 11/20/12 12:16 AM, Andrew Mortensen wrote: > >>>>>> On Nov 19, 2012, at 5:53 PM, David | StyleFlare <david(a)styleflare.com> wrote: > >>>>>> > >>>>>>> Also quick read at the readme, it looks like it does not support multi-domain setups? > >>>>>> Not yet, no. I don't think it would take much work, though. > >>>>> I see it binds to usrloc, what information needs from there? We have two such modules right now, it would be good to make it work with both, kamailio's one has more features in regard to GRUU and partial outbound extensions. > >>>> It's registering for contact expiration and deletion events. The callback terminates subscriptions for the expired/deleted contact. This should probably be configurable. > >>> Ok, so it is about when a contact record in removed from location. > >>> > >>> Just to avoid misunderstandings, do you consider to make configurable this termination of subscriptions based on location? > >> Yes. I can imagine a setup where the location data wouldn't be saved on the host handling SCA. > >> > >>> Also, you need only the AoR and contact address in a callback for location record removal event (un-registration or registration expiration), right? > >> Correct, although Ovidiu makes a good point about (well-behaved) clients unsubscribing at the time of unregistration. In registering for usrloc callbacks, I'm trying to handle a couple problems. First, I'd like to avoid unnecessarily sending and retransmitting NOTIFYs to offline clients. More importantly, I want to be able to hear about expired registrations so I can clear appearance state on the rest of the subscribers. > > For clearing appearance state, you should rely on the dialog module > > and register a callback on the dialog module. When a dialog expires, > > you can clear up the stale appearances. > > Also, by enabling the sst module, you can have a tight control over > > the duration of the dialog. > > This will make the implementation cleaner and more robust. > > > I will not jump to such conclusions. > > While dialog module is designed to track active calls, does not mean > everything has to be on top of it. Just for example, dispatcher module > has an embedded lightweight call tracking system. > > No idea here what the broadsoft extensions require, afaik, there were > some custom specs. It is no problem to have a dedicated dialog tracking > system that is better designed for a particular functionality. We do > have other duplicated systems, for example least cost routing, nat > traversal, a.s.o. > > Also, optimizations and reuse of other parts can be done later if found > to be better ways. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > > > > > ------------------------------ > > Message: 2 > Date: Tue, 20 Nov 2012 17:17:38 -0500 > From: Andrew Mortensen <admorten(a)isc.upenn.edu> > Subject: Re: [SR-Users] Shared Call Appearances module > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <65EAE745-8966-4848-870E-6A50246E0AB7(a)isc.upenn.edu> > Content-Type: text/plain; charset=iso-8859-1 > > > On Nov 20, 2012, at 3:43 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote: > > > Hello Andrew, > > > > First of all, thank you for sharing your work. > > I was following this thread and I have a couple of questions. Why do > > you need to bind to the usrloc module? The subscription itself should > > be sufficient because if a phone will unregister, it will also > > unsubscribe, > > Yes, that's good point. However, if the phone goes off-line instead of unregistering, the contact's registration will expire, and no unsubscribe will take place. As I mentioned in my reply to Daniel, I'm most concerned about catching this so I can release any appearances seized by the expired/deleted contact, and notify other members of the group. I'm certainly open to alternatives. > > > which leads me to the following question: why not add a > > new dedicated module for call-info presence and reuse the existing > > infrastructure from kamailio for handling subscriptions/presence. In > > this case, your module should just push PUBLISH events to the presence > > server and the server will automatically send out notifications for > > subscribers that subscribed to call-info events. > > We've worked extensively with the existing presence & pua modules, albeit primarily dialog;sla using OpenSIPS behind the proxy. (Thanks, Anca!) > > SCA's unusual entanglement with call processing made me hesitate about building on top of the existing presence module. For a first pass, I also felt working with an entirely distinct module gave me more control and transparency during development of a very loosely documented event package, especially as I became more familiar with the available API. > > I don't have any objection to revisiting design decisions, of course. I'm sure the module will continue to evolve, and it would be nice to eliminate redundancy if possible. > > > Based on your README file, you inspect SIP requests/replies and based > > on the presence of the call-info header, you create call-info events. > > This is great for sharing the appearances between phones, but how do > > you perform the retrieval of a call that was put on hold? Are you > > using a dedicated B2BUA behind kamailio? > > No B2BUAs are involved. > > The INVITE retrieving a call held by another member of an SCA group has a particular set of characteristics: RURI, To and From URIs are all the SCA AoR; new unconfirmed dialog (no to-tag); and a Call-Info header referring to the index of the held call. The module detects this type of INVITE, looks up the dialog associated with the information in the Call-Info header, and injects a Replaces header with the dialog of the held call before relaying it to the remote party. The remote party must support RFC3891. I've only worked with Polycoms and Cisco gateways to this point, and both do support that RFC. > > I'm very interested in hearing from users with Cisco (and Snom and Aastra?) SCA setups. I have no doubt they'll find bugs resulting from assumptions I've made in the code because I've only tested with Polycom, not least the dependency on RFC3891 support. > > Thanks for your feedback! > > Best, > andrew > > > > > On Tue, Nov 20, 2012 at 3:16 PM, Daniel-Constantin Mierla > > <miconda(a)gmail.com> wrote: > >> > >> On 11/20/12 8:43 PM, Andrew Mortensen wrote: > >>> > >>> On Nov 20, 2012, at 4:08 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> > >>> wrote: > >>> > >>>> Hello, > >>>> > >>>> thank for contribution! > >>>> > >>>> On 11/20/12 12:16 AM, Andrew Mortensen wrote: > >>>>> > >>>>> On Nov 19, 2012, at 5:53 PM, David | StyleFlare <david(a)styleflare.com> > >>>>> wrote: > >>>>> > >>>>>> Also quick read at the readme, it looks like it does not support > >>>>>> multi-domain setups? > >>>>> > >>>>> Not yet, no. I don't think it would take much work, though. > >>>> > >>>> I see it binds to usrloc, what information needs from there? We have two > >>>> such modules right now, it would be good to make it work with both, > >>>> kamailio's one has more features in regard to GRUU and partial outbound > >>>> extensions. > >>> > >>> It's registering for contact expiration and deletion events. The callback > >>> terminates subscriptions for the expired/deleted contact. This should > >>> probably be configurable. > >> > >> > >> Ok, so it is about when a contact record in removed from location. > >> > >> Just to avoid misunderstandings, do you consider to make configurable this > >> termination of subscriptions based on location? > >> > >> Also, you need only the AoR and contact address in a callback for location > >> record removal event (un-registration or registration expiration), right? > >> > >> > >>> > >>>> Would you consider adding it to main git repository and maintaining it > >>>> there? > >>> > >>> Yes, that would be great. > >> > >> > >> We will prepare write access for you. > >> > >> > >>> Would you prefer to have it in modules? I started work on it in > >>> modules_s, but there's no particular reason it has to stay there. > >> > >> To make it work with both variants of usrloc modules, I think it requires > >> some split of the code. IIRC, both modules still use same names for > >> structures, so including files from both of them will fail. > >> > >> One solutions I am thinking of is to add two small modules, one that binds > >> to usloc(k) and the other to usrloc(s). Each of these two modules will > >> export a function that allow registering callbacks for execution when a > >> location record expires, giving the aor, contact and event type. The > >> existing module will use the intermediate module instead of directly binding > >> to usrloc structures. > >> > >> Usrloc module from kamailio has more enhancements in regards to standard > >> extensions for location services, still it lacks the feature of being able > >> to store variables per contact record. It is the reason that kept back the > >> removal (upon merging) of usrloc module from ser. The solution with > >> intermediary modules should work until we have a merged usrloc module. > >> > >> What do you think about this proposal? > >> > >> Cheers, > >> Daniel > >> > >> > >> -- > >> Daniel-Constantin Mierla - http://www.asipto.com > >> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > >> > > > > > ------------------------------ > > Message: 3 > Date: Wed, 21 Nov 2012 01:04:52 +0200 > From: Juha Heinanen <jh(a)tutpro.com> > Subject: Re: [SR-Users] LCR weight > To: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <20652.3220.931000.738348(a)sip.test.fi> > Content-Type: text/plain; charset=us-ascii > > Camila Troncoso writes: > > > scripts are provided in lcr/utils directory that can be used to check > > the probabilities resulting from a given set of weight values. Same can > > be done with command 'kamctl eval_weights'. > > > > I really don't understand what is the probability finally assign to each > > gateway when they have the same priority. > > All I can see is that it doesn't correspond to percentage of capacity from > > the example above , but how can I determine what value gives the percent > > that I want? > > > > Please give an example like: > > > > Rule_id Gw_id Priority weight > > 36 68 1 60 > > 36 69 1 14 > > 36 70 1 13 > > 36 71 1 13 > > you have to figure that out yourself using the script referenced in > readme. > > -- juha > > > > ------------------------------ > > Message: 4 > Date: Wed, 21 Nov 2012 09:55:17 +0200 > From: Uri Shacked <ushacked(a)gmail.com> > Subject: [SR-Users] problem with $time Pseudo-Variables on 3.3.2 > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAMMbDhQn1r_ke72-u2is8MyRXUV6KkLnZqYwxL5eZG44yYG_2Q(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > I upgraded to 3.3.2 > > On 3.2 i used the variable $time() to get the current time for some process. > > Every 2 minutes a RTIMER process started, checked the $time(hour) and > $time(min) and on a certain time executed something. > > This process stopped working on 3.3.2 > > When i debug, i see that the $time() variable keeps on giving me the same > hour and minute that was given on the first run. > > Meaning the $time() variable does not give the current time. > > I notice also that the $Ts that on 3.2 gave a current time stopped doing it > and $TS does it now.... > > Any help here? > > Is it OK and i need to check time another way? Or something in the $time() > has changed...? > > Thanks, > > Uri >

1 0

dialog ka_timer and ka_interval problem
by Kristofer Signer 21 Nov '12

21 Nov '12

Hello all, first of all I would like to thank you all your help. I've tried to implement dialog keep-alive messages but cant make kamailio send more then just one SIP OPTION request to each end. The OPTION request is sent after ka_interval time but no more OPTION requests are sent during the call. How should this be implemented? and was is the difference and relation between ka_timer and ka_interval. tanks ./Kristofer

2 2

LCR weight
by Camila Troncoso 20 Nov '12

20 Nov '12

Hi, I have a question regarding LCR module and their weight variable. If I have 4 gateways for the same prefix, and with the same priority: Rule_id Gw_id Priority weight 36 68 1 60 36 69 1 14 36 70 1 13 36 71 1 13 The weight variable acts as a percentage of utility of that gateway? If for example I have 100 calls, the gateway with gw_id 68 will receive 60 calls , and the other 14, 13 and 13? Regards, Camila *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 3

Getting started with Kamailio
by Joel Smith | VOZELIA 20 Nov '12

20 Nov '12

Hi, First of all say that I'm new to Kamailio. I've been working with asterisk for a while now and it's time to get started with the sip router project. My idea for a start is to reduce the load of our asterisk servers and once we have good knowledge on how kamailio works, migrate some of our services. I've setup a server in our lab with kamailio 3.3.0 and siremis 3.3.0, created a user (subscriber) and registered with a softphone. So far OK. Now I'm trying to make a call and route it through one of our asterisk servers for testing. To do this I need LCR module (thanks Sébastien, I wasn't subscribed to this list yet) and configure it. After reading http://kamailio.org/docs/modules/stable/modules/lcr.html#id2553110 I've added in kamailio.cfg: #!define WITH_LCR #!ifdef WITH_LCR loadmodule "lcr.so" #!endif #!ifdef WITH_LCR # ----- lcr params ----- modparam("lcr", "db_url", DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "tag_avp", "$avp(lcr_tag)") #!endif And now I see LCR module being loaded at startup. But, I don't understand what gw_uri_avp, ruri_user_avp or tag_avp mean. Where can I find some doc/manual to understand what AVP means and does? (For example the $avp(i:709), etc.) I've added a LCR GW, LCR TARGET and LCR RULE with the following data (I have no idea if it's completely wrong): mysql> select * from lcr_gw; +----+--------+---------+--------------+--------------------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | prefix | tag | flags | defunct | +----+--------+---------+--------------+--------------------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | 1 | 0 | TEST | XX.XXX.XXX.X | test12345678.testing.com | 5060 | NULL | 1 | 1 | NULL | NULL | NULL | 0 | NULL | +----+--------+---------+--------------+--------------------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ 1 row in set (0.00 sec) mysql> select * from lcr_rule; +----+--------+--------+----------+-------------+---------+---------+ | id | lcr_id | prefix | from_uri | request_uri | stopper | enabled | +----+--------+--------+----------+-------------+---------+---------+ | 1 | 1 | .* | | | 0 | 1 | +----+--------+--------+----------+-------------+---------+---------+ 1 row in set (0.00 sec) mysql> select * from lcr_rule_target; +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 1 | 1 | 1 | 1 | 1 | 1 | +----+--------+---------+-------+----------+--------+ 1 row in set (0.00 sec) mysql> Am I still missing basic stuff to be able to place a call ? (To get started I'd like to route all calls to our Asterisk server) I'm glad to read documentation so If all of this is already somewhere please guide me cause I haven't found too much... Thanks in advanced. Regards, Joel.

4 4

t_branch_timeout()
by Alex Balashov 20 Nov '12

20 Nov '12

I am using the latest pull of branch 3.3, and t_branch_timeout() doesn't work if the previous branch resulted in a reply (i.e. a negative final reply). That is to say, t_branch_timeout() == FALSE on a genuine timeout, as long as the preceding branch provided a result (such as 404). t_branch_timeout() == TRUE if this is a timeout, on the first branch. In other words, it appears that if t_any_replied() == TRUE, then t_branch_timeout() == FALSE, even for this branch, and even if the branch really did time out (no response at all from destination). That's not how the documentation says it should behave. Am I doing something wrong? If so, how do I find out if _this particular_ branch timed out, from the failure_route[]? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 3

temporary SIP user
by Bolang 20 Nov '12

20 Nov '12

Hi all, is there any standard mechanism to create a temporary user? i know i can create it and then delete it after some amount of time. but, i'm looking for an established/standard way of doing this. Ideally, the user will be automatically deleted after unregistered. thanks

2 2

Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 20 Nov '12

20 Nov '12

I'd be curious to know if your performance might actually be higher, owing to improved CPUs since the study was done, hyperthreading, more frames per NIC interrupt, etc. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:Thank you, setting ulimits worked! And the performance is the same as stated in the document I mentioned! :) One more thing, I found these errors in syslog: Sep 13 18:38:18 perftest kamailio[5268]: ERROR: <core> [parser/sdp/sdp.c:211]: Invalid payload location Sep 13 18:38:18 perftest kamailio[5268]: ERROR: <core> [parser/sdp/sdp.c:227]: Invalid payload location Sep 13 18:38:18 perftest kamailio[5270]: ERROR: <core> [parser/sdp/sdp.c:227]: Invalid payload location This is possibly something related to my sipp scenario, this is the INVITE sent from sipp <![CDATA[ INVITE sip:800@perftest.vm SIP/2.0 Via: SIP/2.0/[transport] 10.0.2.36:[local_port];branch=[branch] From: "700" <sip:500@10.0.2.36>;tag=[call_number] To: <sip:800@perftest.vm> Call-ID: [call_id] CSeq: 1 INVITE Contact: "700" <sip:700@10.0.2.36:[local_port]> Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] 10.0.2.36 s=- c=IN IP[local_ip_type] 10.0.2.36 t=0 0 m=audio [auto_media_port] RTP/AVP 8 a=rtpmap:8 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 ]]> Do you see something that could cause this error ? Otherwise the call is initiated ok, but I really dont understand what is so strange to kamailio in this INVITE. On Thu, Sep 13, 2012 at 6:37 PM, Peter Lemenkov <lemenkov(a)gmail.com> wrote: > 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: >> You mean on the proxy side? I'm running rtpproxy as root, limits are >> still applied ? ulimit -s unlimited should do the trick ? > > Yes, they usually applied even for superuser, and yes - this should > help (if that's the issue). > > > -- > With best regards, Peter Lemenkov. > > _______________________________________________ > 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 _______________________________________________ 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

6 14

Clarification on is_known_dlg()
by David 20 Nov '12

20 Nov '12

Hello, I am running Kamailio 3.2.3. I am storing dialogs in the database table dialog. loadmodule "dialog.so" modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 1) later I have : if ( is_known_dlg() ) { } I would like a clarification on how the matching is done. Given that I have realtime table setup, does that mean that dialogs that are in the table will return true when I call is_known_dlg() ? Does that mean that if the dialog is NOT in the table, is_known_dlg will return false ? Thanks for the clarification ! David

2 1

UAC and realm
by Johan Wilfer 20 Nov '12

20 Nov '12

Hi, I've done some tests with the UAC module to authenticate to a remote proxy. I've based my config on this example: http://docs.huihoo.com/opensips/tutorials/uac/ar01s06.html (example 9) I have found that if I send a call from a asterisk via kamailio to my remote proxy the realm the remote proxy uses is the ip-address of the asterisk-server (aaa.bbb.ccc.ddd). Like this: WWW-Authenticate: Digest nonce="1353399363:8d2317487ee1521328d0e2237e444e2d",algorithm=MD5,realm="aaa.bbb.ccc.ddd",qop="auth",stale=false. If I have a param like this, the auth works: modparam("uac","credential","username:aaa.bbb.ccc.ddd:password") But as I would like to use more than one remote proxy (=more than one provider) per asterisk server this needs some more work. Is there a way to get UAC to ignore the realm from the remote proxy and send the auth anyway? Alternate paths I've examined: 1. Rewrite from-domain to get the provider to use another realm: modparam("uac","credential","username:example.com:password") uac_replace_from("sip:$fU@example.com"); Works, but I don't think this is a very good solution. 2. Use auth_username_avp / auth_password_avp / auth_realm_avp to feed the uac_auth()-function the right credentials. modparam("uac","auth_username_avp","$avp(s:uac_user)") modparam("uac","auth_password_avp","$avp(s:uac_pass)") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") in the failure route: $avp(s:uac_user) = "username"; $avp(s:uac_pass) = "password"; $avp(s:uac_realm) = "aaa.bbb.ccc.ddd"; This works as well, but I would like to either get the realm the remote proxy sends and use it or make uac ignore the realm. The realm is unimportant as we already know what proxy we are talking to. I've tried $hdr(WWW-Authenticate) but it evaluates to null. Any ideas how to do this? Thanks! -- Johan Wilfer

2 2

Re: [SR-Users] Returning problem
by Moacir Ferreira 19 Nov '12

19 Nov '12

I wish it were that simple... I have done some "reading" and found that Ubuntu has changed to upstart http://upstart.ubuntu.com/cookbook/. However, I could not find how I can start a non "upstart" style init script like Kamailio has before the upstart styled ones... I guess that I have to either to get the Kamailio community to create an upstart script or find some help from the Ubuntu community how to overcome the problem. Anyway, I think the Kamailio team should address it as it will become a problem for everyone using Kamailio on Ubuntu. Any help? Moacir From: 4lists(a)gmail.com To: sr-users(a)lists.sip-router.org Date: Mon, 12 Nov 2012 14:42:31 -0200 Subject: Re: [SR-Users] Returning problem This is more a system initialization problem, so, take a look in your /etc/rc<run-level>.d directory (or the correspondent one), specially in the S* files and find out the start sequence. There You'll find the answer. Edson. Em 09/11/2012 17:03, Moacir Ferreira escreveu: > My kamailio start script has the following BEGIN INIT INFO: > > # > ### BEGIN INIT INFO > # Provides: kamailio > # Required-Start: $syslog $network $local_fs $time mysql > # Required-Stop: $syslog $network $local_fs mysql > # Default-Start: 2 3 4 5 > # Default-Stop: 0 1 6 > # Should-Start: postgresql mysql radius > # Should-Stop: postgresql mysql radius > # Short-Description: Start the Kamailio SIP proxy server > # Description: Start the Kamailio SIP proxy server > ### END INIT INFO > > However, when I boot my server Kamailio will start before mysql. So, > after boot I can start Kamalio but at boot time it will fail. > > Any hints how to fix it? I am running Ubuntu server 12.04 LTS - 64 bits. > All the code has been compiled at the server. > > Cheers! > > Moacir > > > _______________________________________________ > 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 >

2 5

how to use uac_auth in module UAC
by Freddy C 19 Nov '12

19 Nov '12

Hello, I have a question about UAC and function module "uac_auth" for a REGISTER request. My idea is to use this when I get a 401 unauthorized response. This answer is captured from failure_route. Then generate the header Authorized and forward the request REGISTER to Asterisk. I could help with some examples?. Freddy Condori Ch Ingeniero Investigador Telefonía IP

1 0

Sync nonce between various servers
by Andreas Granig 19 Nov '12

19 Nov '12

Hi, There are lots of parameters controlling the creation of nonce values on a server, and I'm curious if there is a way to kind of "sync" them between servers. The use case would be to have a UA send for example its registration to Proxy1. Proxy1 would challenge it, UA will send the registration again, this time with credentials. Proxy1 would look up the user based on $au/$ar in the subscriber table, and if it's not found, will look up the responsible proxy from another table (with key being $au@$ar), forward it to Proxy2, which then would be able authenticate the user. The reason for this is that the auth credentials are unique across all servers and reliably identify a user, whereas for example From could be something else (e.g. in case of an IP-PBX sending a CLI in the From-userpart). Challenging the user on the second proxy again would theoretically be possible, but if the UA gets a 401 twice (once from Proxy1, once from Proxy2), it'll most likely pop up a password form for soft-clients, so I want to avoid that. Any ideas how to accomplish that? Andreas

4 9

Problem routing to voicemail
by Christophe ROY 19 Nov '12

19 Nov '12

Hi everyone I'm trying to integrate Asterisk with Kamailio for voicemail. I tried to follow this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb BUT: - I had to adapt it because I use LDAP authentication with Kamailio - I had problems with Asterisk 10.7 (problems with chan_sip module crashing) so I've installed Asterisk 11 on another VM - we have high-availability with 2 Kamailio servers, with Kamailio listening on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual IP" (created by keep-alive): this VIP is not visible with ifconfig, but you can see it with the command "ip addr sh eth0" For now, we use Linphone on Windows as SIP clients to test. If I don't define WITH_ASTERISK, calls work, I can call someone(a)domain.tld However, if I define WITH_ASTERISK, calls fail (even with destination registered and available) and I have these errors in the logfile: Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply error Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) It seems to happen on the if (!t_relay()) line in ROUTE[RELAY] 192.168.14.25 is the real IP of the Kamailio server, 192.168.14.24 is the VIP of the Kamailio "cluster" 192.168.14.28 is the IP of the Mysql server 192.168.14.32 is the IP of the Asterisk server I can't find why the relay doesn't work. I've tried to bypass the VIP and have Kamailio listen on the real IP, but it still doesn't work: I don't seem to have the same errors as above, but I don't see any traffic between Kamailio and Asterisk. What could be the problem? Thanks for your help Christophe Below is my kamailio.cfg: #!WITH_DEBUG #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # ###!define WITH_NAT # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ###!define WITH_ASTERISK ###!define WITH_VOICEMAIL #!define WITH_LDAP #!define WITH_AUTH #!define WITH_MYSQL ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openserrw:openserrw@192.168.14.28/openser" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://asterisk:asteriskpwd@192.168.14.28/asterisk" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL6 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="mydomain.corp" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=tcp:192.168.14.24:5060 #listen=tcp:192.168.14.25:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ #port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "192.168.14.32" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.14.32" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "192.168.14.24" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" # loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #chris loadmodule "ldap.so" modparam ("ldap", "config_file", "/etc/kamailio/ldap.cfg") #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("mi_fifo", "fifo_name", "/tmp/kamailio_tmp") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # suppress the check for the CSEQ method # modparam("sanity", "default_checks", 967) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #edit asterisk #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 256) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif #chris commented out this part # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.14.25:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@teopad-toip.corp") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); xlog("L_INFO","apres REQINIT"); # NAT detection route(NATDETECT); xlog("L_INFO","apres NATDETECT"); # handle requests within SIP dialogs route(WITHINDLG); xlog("L_INFO","apres WITHINDLG"); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); xlog("L_INFO","apres t_check_trans"); # authentication route(AUTH); xlog("L_INFO","apres AUTH"); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); xlog("L_INFO","apres RECORD ROUTE"); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } xlog("L_INFO","apres INVITE"); # dispatch requests to foreign domains route(SIPOUT); xlog("L_INFO","apres SIPOUT"); ### requests for my local domains # handle presence related requests route(PRESENCE); xlog("L_INFO","apres PRESENCE"); # handle registrations route(REGISTRAR); xlog("L_INFO","apres REGISTRAR"); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); xlog("L_INFO","apres PSTN"); # user location service route(LOCATION); xlog("L_INFO","apres LOCATION"); route(RELAY); xlog("L_INFO","apres RELAY"); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. xlog("L_INFO","Dans route relay"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","avant manage branch"); t_on_branch("MANAGE_BRANCH"); xlog("L_INFO","avant manage reply"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { xlog("L_INFO","avant manage failure"); t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { xlog("L_INFO","reply error"); sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # if(!sanity_check("1511", "7")) # { # xlog("Malformed SIP message from $si:$sp\n"); # exit; # } } # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Dans WITHINDLG"); if (has_totag()) { xlog("L_INFO","dans has totag"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO","looseroute"); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { xlog("L_INFO","ack"); # ACK is forwarded statelessy route(NATMANAGE); } xlog("L_INFO","relay"); route(RELAY); } else { xlog("L_INFO","else"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests xlog("L_INFO","subscribe avant presence"); route(PRESENCE); xlog ("L_INFO","apres presence"); exit; } if ( is_method("ACK") ) { xlog("L_INFO","else ack"); if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server xlog("L_INFO","else ack avant relay"); t_relay(); xlog("L_INFO","else ack apres relay"); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_INFO","else final"); exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #edit asterisk #!ifdef WITH_ASTERISK xlog ("L_INFO","avant regfwd dans registrar"); route(REGFWD); xlog ("L_INFO","apres regfwd dans registrar"); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #edit asterisk #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { #if new call from out there - send to Asterisk # - non-INVITE requests are routed directly by Kamailio # - traffic from Asterisk is router also directly by Kamailio xlog ("L_INFO", "avant toasterisk dans location"); route(TOASTERISK); xlog ("L_INFO", "apres toasterisk dans location"); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #edit asterisk #!ifdef WITH_ASTERISK #do not auth traffic from Asterisk: trusted! xlog ("L_INFO", "avant if route fromasterisk"); if(route(FROMASTERISK)) return; #!endif if (is_method("REGISTER")) # { # # authenticate the REGISTER requests (uncomment to enable auth) # if (!www_authorize("$td", "subscriber")) # { # www_challenge("$td", "0"); # exit; # } # # if ($au!=$tU) # { # sl_send_reply("403","Forbidden auth ID"); # exit; # } { #edit asterisk ##!ifdef WITH_ASTERISK # xlog ("L_INFO", "dans auth / authcheck sipusers"); # if (!auth_check("$fd","sipusers","1")) ##!else if(is_present_hf("Authorization")) ##!endif { # ldap search if (!ldap_search("ldap://sipaccounts/OU=SIP,OU=Utilisateurs,DC=teopad-toip,DC=corp?teopad-Sip-Username,teopadSipPassword?one?(teopad-Sip-Username=$fU)")) # if (!ldap_search("ldap://sipaccounts/OU=SIP,OU=Utilisateurs,DC=teopad-toip,DC=corp?sAMAccountName,?one?(sAMAccountName=$fU)")) { switch ($retcode) { case -1: # no LDAP entry found sl_send_reply("404", "User Not Found"); xlog("L_INFO", "ldap_search: NO found [$retcode] entries for (uid=$fU)"); exit; case -2: # internal error sl_send_reply("500", "Internal server error"); exit; default: exit; } } ldap_result("teopad-Sip-Username/$avp(s:username)"); ldap_result("teopadSipPassword/$avp(s:password)"); xlog("L_INFO", "ldap_search: found [$retcode] entries for (uid=$fU)"); if(!pv_www_authenticate("$td", "$avp(s:password)", "0")) { xlog ("L_INFO", "ldap pv_authenticate failed") ; www_challenge("$td","1"); exit; } save("location"); sl_send_reply("200", "ok"); xlog ("L_INFO", "ldap pv_authenticate ok") ; exit; } else { www_challenge("$td","1"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # # authenticate if from local subscriber if (from_uri==myself) { # if (!proxy_authorize("$fd", "subscriber")) { # proxy_challenge("$fd", "0"); # exit; # } if (is_method("PUBLISH")) { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; xlog ("L_INFO", "tU = $tU") ; xlog ("L_INFO", "fd = $fd") ; xlog ("L_INFO", "rd = $rd") ; if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID au!=fu ou au!=tu"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; # if ($au!=$fU) { # sl_send_reply("403","Forbidden auth ID au!=fu"); # exit; # } } #consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); #rtpproxy_manage("co","82.127.95.167"); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "tcp:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); xlog("L_INFO","tovoicemail ru: $ru"); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_INFO","dans managereply"); xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]"){ xlog("L_INFO","avant route natmanage"); route(NATMANAGE); } } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #edit asterisk #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { xlog ("L_INFO", "Dans FROMASTERISK? $si / $sp"); if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); xlog ("L_INFO", "Dans TOASTERISK $du") ; xlog ("L_INFO", "Juste avant route relay"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { xlog("L_INFO", "Dans REGFWD"); if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; xlog("L_INFO","avant if dans regfwd"); if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif

2 3

Problem with include_file quotes
by Pirjo Ahvenainen 19 Nov '12

19 Nov '12

Hi All, I've been away from Kamailio for a few months, but recently I started working with it again. I could use a hand in a small problem: I'm using include_file statement to give some structure to my config file, but when checking the config I keep getting error message: 0(4290) : <core> [cfg.lex:1554]: included file name must be between quotes: "defined_modules.cfg" I'm using Kamailio 3.3.2 (version: kamailio 3.3.2 (x86_64/linux) 78d4b1) on Ubuntu. And here's the beginnign of the config file: #!KAMAILIO # include_file "defined_modules.cfg" include_file "defined_values.cfg" include_file "global_parameters.cfg" Those look like quotes to me, but could the problem be something like encoding of the config file itself, something in the included files, or is there something else I can check? I don't have extra whitespaces etc, perhaps a typo I just don't see? Cheers, Pirjo

2 3

How to get dialog variable externally?
by Mino Haluz 19 Nov '12

19 Nov '12

Hi, I'm using dialog module and I set some dlg_var for every dialog. Is it possible to get this variable value with MI command somehow ? I know to get list of dialogs using dlg_list only. Thanks! Mino

2 1

db_sqlite bugs
by Pedro Antonio Vico Solano 19 Nov '12

19 Nov '12

5 11

Planning for memory
by Moacir Ferreira 19 Nov '12

19 Nov '12

Hi, I am trying to figure out how much memory a Kamailio server should have to support an X number of users, considering that I am running Kamailio, MySQL and RTPproxy on the same machine. I am planning to use Kamailio at medium/large enterprises, not at service providers but I got the following questions: - Will Kamailio dynamically allocate the system memory it needs or I have to tune up the memory it takes? I.e.: If I use 8G memory, will it take what it needs or I need to adjust it somewhere? - How much memory is reasonable for 10K, 50K and 100K users in such "enterprise" scenarios? - Can you reference any documentation from where I can get this information? Thanks, Moacir

2 1

Kamailio cfg file doubt
by Kamal Palei 17 Nov '12

17 Nov '12

Dear All When I get INVITE (received in UDP), I send it to another proxy in TLS. For this purpose I have modified kamailio.cfg route section , please refer the lines in red. Just before sending in TLS, I modify the IP address in SDP, so that media packets will pass through RTP proxy running When INVITE is reached the next proxy, I see the IP address is changed . So far so good. Next I am trying to modify the IP address in 200 OK to INVITE. However it is not taking effect. I hope we need to modify in onreply_route function. Something is wrong here. I am receiving 200 OK for INVITE in TLS. Next I am sending 200 OK for INVITE to endpoints using UDP. Not sure if onreply_route is being executed or not. Can somebody give me pointer whats wrong here. *# Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy("r"); } if (isbflagset(FLB_NATB)) { fix_nated_contact(); } #!endif } * * # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); setflag(FLT_NATS); # NAT detection route(NAT); if(is_method("PUBLISH")) { loose_route(); t_relay(); exit; } if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:443"); exit(); } if(is_method("INVITE|BYE|CANCEL|SUBSCRIBE|REFER|NOTIFY")) { xdbg("incoming request\n"); route(RTPPROXY); t_relay_to("tls:115.114.48.75:443"); exit(); } *Thannks Kamal

2 2

segfault when phone registers on TLS
by Denis 15 Nov '12

15 Nov '12

Hi, I am using kamailio-3.3.2 on Ubuntu 11.04 x64 and when trying to connect via TLS from my Yealink phone kamailio crashes with the following backtrace: Loaded symbols for /lib/x86_64-linux-gnu/libnss_files.so.2 Core was generated by `/opt/kamailio/sbin/kamailio'. Program terminated with signal 11, Segmentation fault. #0 handle_ser_child (p=0x7fcbaea2eba0, fd_i=-1) at tcp_main.c:3575 3575 if (unlikely(p->unix_sock<=0)){ (gdb) bt #0 handle_ser_child (p=0x7fcbaea2eba0, fd_i=-1) at tcp_main.c:3575 #1 0x000000000051e5bc in send2child (tcpconn=0x7f8c001597d8, ev=<value optimized out>, fd_i=-1) at tcp_main.c:3975 #2 handle_tcpconn_ev (tcpconn=0x7f8c001597d8, ev=<value optimized out>, fd_i=-1) at tcp_main.c:4310 #3 0x0000000000527dbc in io_wait_loop_epoll () at io_wait.h:1092 #4 tcp_main_loop () at tcp_main.c:4656 #5 0x00000000004726f5 in main_loop () at main.c:1727 #6 0x000000000047402e in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2546 (gdb) The manual I am following is http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates Thanks, Den

3 10

Kamailio as SMTP server
by Krishna Kurapati 15 Nov '12

15 Nov '12

Hi, Can Kamailio act as SMTP server to receive emails and forward those to SIP users as MESSAGEs? It would be nice if postfix or some other rich SMTP server could function as module of kamailio using all the services provided by kamailio scripting. Krish Kura

2 1

Problem routing to voicemail
by Christophe ROY 15 Nov '12

15 Nov '12

Hi everyone I'm trying to integrate Asterisk with Kamailio for voicemail. I tried to follow this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb BUT: - I had to adapt it because I use LDAP authentication with Kamailio - I had problems with Asterisk 10.7 (problems with chan_sip module crashing) so I've installed Asterisk 11 on another VM - we have high-availability with 2 Kamailio servers, with Kamailio listening on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual IP" (created by keep-alive): this VIP is not visible with ifconfig, but you can see it with the command "ip addr sh eth0" For now, we use Linphone on Windows as SIP clients to test. If I don't define WITH_ASTERISK, calls work, I can call someone(a)domain.tld However, if I define WITH_ASTERISK, calls fail (even with destination registered and available) and I have these errors in the logfile: Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply error Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) It seems to happen on the if (!t_relay()) line in ROUTE[RELAY] 192.168.14.25 is the real IP of the Kamailio server, 192.168.14.24 is the VIP of the Kamailio "cluster" 192.168.14.28 is the IP of the Mysql server 192.168.14.32 is the IP of the Asterisk server I can't find why the relay doesn't work. I've tried to bypass the VIP and have Kamailio listen on the real IP, but it still doesn't work: I don't seem to have the same errors as above, but I don't see any traffic between Kamailio and Asterisk. My kamailio.cfg is available here as it's too big for the list: http://pastebin.com/zhncJ6j1 What could be the problem? Thank you very much for your help Christophe

1 0

Replicate dialog state
by Kristofer Signer 15 Nov '12

15 Nov '12

In my current setup I've register messages replicated to a second server. Is it possible to also replicate other messages so I can hold dialog state on both servers? I've tried to read the documentation and also google but can't find any examples. Thanks ./Kristofer

2 1

RPC, XMLRPC or other
by Kristofer Signer 14 Nov '12

14 Nov '12

Hello, I'm trying to dig in to the kamailio XMLRPC interfaces and the limitations I read in http://www.kamailio.org/docs/modules/stable/modules/xmlrpc.html#xmlrpc.impl… is really a no go for us. For example, when to retrieve user locations and list of dialogs we get a xml document which is not so well formatted and we can not successfully parse it in an xmlrpc-parser. Is there other options for xmlrpc module? thanks ./Kristofer

4 11

check if the sent reply was local
by Uri Shacked 14 Nov '12

14 Nov '12

Hi, what is the command to check if the reply was recieved or localy generated? Uri

3 3

Re: [SR-Users] sr-users Digest, Vol 90, Issue 39
by Yufei Tao 14 Nov '12

14 Nov '12

What I did to overcome this problem was writing a bash script, that checks if mysql is running, if so, start Kamailio, if not, start mysql first, then start Kamailio, and retries if things fail. And in /etc/rc.local, call this script. Seems to work fine on Ubuntu 10.04. Not tried 12.04 yet. Yufei > > > ------------------------------ > > Message: 2 > Date: Tue, 13 Nov 2012 18:15:05 -0500 > From: Moacir Ferreira <moacirferreira(a)hotmail.com> > Subject: Re: [SR-Users] Returning problem > To: Kamailio Mail List <sr-users(a)lists.sip-router.org> > Message-ID: <COL125-W3986EA29065F0966C8AE6C86C0(a)phx.gbl> > Content-Type: text/plain; charset="iso-8859-1" > > > I wish it were that simple... I have done some "reading" and found that Ubuntu has changed to upstart http://upstart.ubuntu.com/cookbook/. However, I could not find how I can start a non "upstart" style init script like Kamailio has before the upstart styled ones... I guess that I have to either to get the Kamailio community to create an upstart script or find some help from the Ubuntu community how to overcome the problem. Anyway, I think the Kamailio team should address it as it will become a problem for everyone using Kamailio on Ubuntu. Any help? Moacir From: 4lists(a)gmail.com > To: sr-users(a)lists.sip-router.org > Date: Mon, 12 Nov 2012 14:42:31 -0200 > Subject: Re: [SR-Users] Returning problem > > This is more a system initialization problem, so, take a look in your > /etc/rc<run-level>.d directory (or the correspondent one), specially in > the S* files and find out the start sequence. There You'll find the answer. > > Edson. > > Em 09/11/2012 17:03, Moacir Ferreira escreveu: >> My kamailio start script has the following BEGIN INIT INFO: >> >> # >> ### BEGIN INIT INFO >> # Provides: kamailio >> # Required-Start: $syslog $network $local_fs $time mysql >> # Required-Stop: $syslog $network $local_fs mysql >> # Default-Start: 2 3 4 5 >> # Default-Stop: 0 1 6 >> # Should-Start: postgresql mysql radius >> # Should-Stop: postgresql mysql radius >> # Short-Description: Start the Kamailio SIP proxy server >> # Description: Start the Kamailio SIP proxy server >> ### END INIT INFO >> >> However, when I boot my server Kamailio will start before mysql. So, >> after boot I can start Kamalio but at boot time it will fail. >> >> Any hints how to fix it? I am running Ubuntu server 12.04 LTS - 64 bits. >> All the code has been compiled at the server. >> >> Cheers! >> >> Moacir >> >> >> _______________________________________________ >> 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 >> > > > >

1 0

update the to_tag on a local generated reply
by Uri Shacked 14 Nov '12

14 Nov '12

Hi, When i set a timer to "request time out" kamilio generates a 408 with now to_tag. I saved the to_tag from the 183 and try to update the acc table with the to_tag (i need that for my CDR to work) My question is (understanding there is no option to set the to_tag on the 408) is: How to i get the right timing to update using sqlops? i must update after the acc inserted the row to the acc table.... Any ideas? Thanks, Uri

2 3

Presence Dialog Info synchronisation issue
by David 12 Nov '12

12 Nov '12

Hello, I have two Kamailio boxes running. First one routes calls between my phones and asterisk. Here are the extra lines compared to the default config that came with my installation on the first server. Let's call this server kamailio : loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_dialoginfo.so" modparam("pua_dialoginfo", "include_localremote", 0) modparam("pua_dialoginfo", "include_tags", 0) modparam("pua_dialoginfo", "include_callid", 0) modparam("pua_dialoginfo", "caller_confirmed", 1) modparam("pua", "outbound_proxy", "sip:kamailio_presence_ip:5060") if ( is_method("SUBSCRIBE") ) { t_relay("kamailio-presence-ip", "5060"); } The second one handles presence stuff ( SUBSCRIBE and PUBLISH ), let's call it presence : After the standard sanity checks : if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; exit; The problem is that I have two phones connected to the same username on the server. Here is the sequence of events of a correctly functionning call : 1. Kamailio receives an INVITE 2. It sends the INVITE to each of the two phones 3. The first phone replies 180 Ringing 4. Kamailio sends a PUBLISH to presence 5. Presence replies with an OK, which includes a SIP-ETag header 6. Kamailio recieves the OK 7. Kamailio receives the 180 Ringing from the second phone 8. Kamailio sends a new PUBLISH with the SIP-If-Match using the value from the SIP-ETag header 9. Presence receives it, updates the presence table and sends an OK. This is ok, the trouble is when the second 180 Ringing reaches Kamailio before the OK to the publish with the SIP-ETag. 1. Kamailio receives an INVITE 2. It sends the INVITE to each of the two phones 3. The first phone replies 180 Ringing 4. Kamailio sends a PUBLISH to presence 5. Kamailio receives the 180 Ringing from the second phone 6. Kamailio sends a PUBLISH to presence 7. Presence replies with an OK, which includes a SIP-ETag header 8. Presence replies with an OK, which includes a second SIP-ETag header 9. Kamailio recieves the OK 10. Kamailio recieves the OK The presence table now has two seperate dialogs for the call because Kamailio didn't send the SIP-If-Match header as it did not yet receive the reply from presence. When the two phones start ringing at the same time, the time between the two packets arriving at the server is shorter than a round trip to the presence server. In the second scenario, Kamailio never updates the presence information using the first SIP-ETag, so the light on my phone stays stuck at "ringing". Where do I go from here ? Thanks, David

1 0

Returning problem
by Moacir Ferreira 12 Nov '12

12 Nov '12

My kamailio start script has the following BEGIN INIT INFO: # ### BEGIN INIT INFO # Provides: kamailio # Required-Start: $syslog $network $local_fs $time mysql # Required-Stop: $syslog $network $local_fs mysql # Default-Start: 2 3 4 5 # Default-Stop: 0 1 6 # Should-Start: postgresql mysql radius # Should-Stop: postgresql mysql radius # Short-Description: Start the Kamailio SIP proxy server # Description: Start the Kamailio SIP proxy server ### END INIT INFO However, when I boot my server Kamailio will start before mysql. So, after boot I can start Kamalio but at boot time it will fail. Any hints how to fix it? I am running Ubuntu server 12.04 LTS - 64 bits. All the code has been compiled at the server. Cheers! Moacir

2 1

LUA 5.2 - Support
by David | StyleFlare 12 Nov '12

12 Nov '12

Curious to know if there is any plans to support LUA 5.2. Seems like this is a newer release, I tried to build kamailio lua modules using 5.2 but it was not supported. Seems like some minor changes in the way LUA handles some things. Any ideas? Thanks.

2 1

xcap: no more space for spec value
by Juha Heinanen 12 Nov '12

12 Nov '12

i got the these error messages when my xcap server tried to serve 'GET xcap-caps' request: Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: INFO: XCAP service GET <xcap-caps> Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: <core> [pvapi.c:1305]: no more space for spec value Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: <core> [pvapi.c:1314]: buffer overflow -- increase the buffer size... Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: <core> [sr_module.c:1665]: Could not convert the PV-formated string to str Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: xhttp [xhttp_mod.c:461]: unable to get body config code where this comes is this: xlog("L_INFO", "XCAP service $rm <$xcapuri(u=>auid)>\n"); $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> <auid>org.openmobilealliance.user-profile</auid> <auid>org.openmobilealliance.pres-content</auid> <auid>org.openmobilealliance.search</auid> <auid>org.openmobilealliance.xcap-directory</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> <namespace>urn:oma:xml:xdm:user-profile</namespace> <namespace>urn:oma:xml:prs:pres-content</namespace> <namespace>urn:oma:xml:xdm:search</namespace> <namespace>urn:oma:xml:xdm:xcap-directory</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "OK", "application/xcap-caps+xml", "$var(xbody)"); i have not checked yet, which buffer should be increased, but would it be a good idea to make the default bigger? -- juha

2 2

Use different flags in rtpproxy_manage() based on from which route its called
by Aft nix 12 Nov '12

12 Nov '12

Hi, rtpproxy_manage() combines both rtpproxy_offer() and rtpproxy_answer(). Now if i want different flags set for rtpproxy_offer() and rtpproxy_answer() then how should i use rtpproxy_manage()? To specific, what i want : route[REQUEST]{ ..... rtpproxy_offer("z20"); ... } onreply_route[]{ .... rtpproxy_answer("z120"); .... } Now in kamailio.cfg there is one call rtpproxy_manage() in route[NATMANAGE]. route[NATMANAGE] is called from both route[REQUEST] and onreply_route[]. So i can't emulate the functionality described above. My question is using only rtpproxy_manage() is it possible to emulate the above functionality? Thanks in advance. -- -aft

3 3

Homer multiple domains
by Reda Aouad 11 Nov '12

11 Nov '12

Hi I am using homer and managing multiple SIP domains. My problem is that in the sip_capture table, the FROM and TO domains are not logged in the FROM and TO fields. The MSG field contains the domains, but the queries involving domains become tricky and heavy to execute. Is there a way I can have the domains in their respective FROM/TO columns? Thanks Reda

1 0

how to use "t_drop_replies" but keep the last reply for acc
by Uri Shacked 11 Nov '12

11 Nov '12

Hi, I am trying to make an option of "route when no answer" or " route when busy". What I am doing is checking the reply and if "busy", for example, I use "t_drop_replies". Then, I set the new number and route[relay] again. On the accdb table, I get the first invite with 183 and after that the second invite with 183 and with 200. I would like to do exactly what i do, but would like to see on the accdb the 486 reply from the first invite. how do i do it? BR, Uri

4 11

Kamailio language definitions- dictionary
by kellon＠sis-grenada.com 11 Nov '12

11 Nov '12

Hi All Can anyone direct me to some go literature/documentation on the language/functions used in the kamailio config file. E.g cfg_get(), strempty(), proxy_authorize(), sl_send_reply() etc.... Thanks Kellon

2 1

No Debug Log message
by kellon＠sis-grenada.com 10 Nov '12

10 Nov '12

Hi all I am currently having to modify the kamailio config file without the benefit of log messages in /var/log/message. I am using the default configuration parameter settings of: #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 Has anyone experience this issue before or have any suggestions on howto fix. Making changes to the config file line by line and restarting the daemon is killing me!!!

2 1

Kamailio CDR logging to RADIUS
by Owen Lynch 10 Nov '12

10 Nov '12

Hi, I have transaction accounting to RADIUS setup and get the start and stop events logged to file when a call is made. I want to extend this to also get CDR logs - call duration etc. To do this I understand I need to configure the acc module and load and configure the dialog module: #!define FLD_DLG 4 modparam("acc", "cdr_enable", 1) modparam("dialog", "dlg_flag", FLD_DLG) I was expecting extra entries in the RADIUS file but they don't appear. I think this is because I need to set and clear the dialog flag at appropriate places in the route script (e.g. setflag(FLD_DLG)) but I'm not sure where. Should this be done in the INVITE and BYE method handling as is currently done for the transaction accounting, e.g.: # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } Thanks, Owen Lynch

3 4

"branch" tag in the "Via" header of the ACK message for the re-INVITE
by Alex Solt 10 Nov '12

10 Nov '12

Hi, We have a problem with the "branch" tag in the "Via" header of the ACK message for the re-INVITE. The "branch" tag value is not a unique number so the end point thinks the ACK is belong to the previous call. This causes the active calls fail after 15 min (right after the ACK message for re-INVITE passed by kamailio proxy). Is there a way to set the "branch" tag in the "Via" header of the ACK message for the re-INVITE to be a unique number? PS1: We are running kamailio v3.3.PS2: we did not have this issue with the old version of openser. Thanks,AS

2 3

Dispatcher question
by Mino Haluz 10 Nov '12

10 Nov '12

Hi, I have simple dispatcher that is dispatching initial invites to the core router. The question is, how can I tell kamailio to not add the Via header in the INVITE. I mean, I want that this dispatcher will be completely transparent to the core router, so any responses will go directly to the original sender, not to the dispatcher. This is my request route: request_route { if (is_method("INVITE")) { if (!ds_select_domain(1, 0)) { xlog("L_INFO", "[MAIN] ERROR: Proxy1 failed"); if (!ds_select_domain(2, 0)) { xlog("L_INFO", "[MAIN] FATAL: Proxy2 failed"); exit; } } xlog("L_INFO","[MAIN] Dispatching INVITE to $rd"); forward(); } } 1 and 2 are defined in dispatcher.list. If it can be done in kamailio, please how. Thank you! Mino

3 5

presence module makes excessive cleanups when subs_db_mode=3
by Andrew Pogrebennyk 09 Nov '12

09 Nov '12

Hi, when presence module is running with subs_db_mode=3 it makes an excessive number of SQL select queries, litelly dozens per second: > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=64 AND event<>'presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=116 AND event<>'presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=116 AND event='presence.winfo' > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=64 AND event='presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=14 AND event<>'presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=14 AND event='presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=118 AND event<>'presence.winfo' > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=66 AND event<>'presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=118 AND event='presence.winfo' > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=66 AND event='presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=16 AND event<>'presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=16 AND event='presence.winfo' Steps to reproduce: registers two subscribers in jitsi and add them to contact lists of each other. The kamailio version is 3.3.2. I'm not doing anything special configuration-wise: loadmodule "presence.so" modparam("presence", "db_url", "mysql://kamailio:snbF93@localhost/kamailio") # in 3.3 the fallback2db change to subs_db_mode modparam("presence", "subs_db_mode", 3) modparam("presence", "notifier_processes", 3) loadmodule "presence_xml.so" modparam("presence_xml", "db_url", "mysql://kamailio:snbF93@localhost/kamailio") modparam("presence_xml", "force_active", 0) modparam("presence_xml", "integrated_xcap_server", 1) # retry-after 5 minutes modparam("presence_xml", "xcapauth_userdel_reason", "probation;retry-after=300") Q: it is a bug of a feature? :) Thanks. Andrew

2 2

xmlrpc dlg.end_dlg problem
by Kristofer Signer 09 Nov '12

09 Nov '12

Hi all, I have an kamailio 3.3.2 server deployed with dialog awareness. When call dlg.end_dlg over xmlrpc I get an 200 OK response but the dialog is not ended. What I can do; call dlg.list (xmlrpc) and get the active dialog list, xmlrpc interface working. call dlg.dlg_list (xmlrpc) and get description of one dialog, xmlrpx method call with parameters working. call dlg_end_dlg (kamctl) and successfully end the dialog. so my questions are... Is the xmlrpc method dlg.end_dlg known to work? and if, what have I missed? thanks ./Kristofer Signer $ /usr/local/kamailio-3.3/sbin/kamailio -V version: kamailio 3.3.2 (x86_64/freebsd) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, select, kqueue. id: unknown compiled on 09:53:26 Nov 2 2012 with gcc 4.2.1

2 3

dispatcher: What's "first destination" in a database
by Olle E. Johansson 08 Nov '12

08 Nov '12

Friends, Dispatcher has a mode 8 - "use first destination" which makes sense for a textfile driven dispatcher configuration. Now if I use a database - will dispatcher first sort on priority, then take the first? It seems so from the documentation of the text file: priority: sets the priority in destination list (based on it is done the initial ordering inside the set) Which one is top priority - lowest or highest value? I can update the docs to clarify if someone helps me with the answer :-) /O

2 1

Another NAT question
by Moacir Ferreira 08 Nov '12

08 Nov '12

Hi, I have a scenario where I got a firewall with 3 interfaces: internal, DMZ and external. All the traffic from internal going to external is NATed. However, the traffic between internal and DMZ is NOT NATed. The external and DMZ are using public IP addresses. On the DMZ I have a Kamailio server running with RTPProxy + NAT Helper. Everything works fine when public (from the internet) users (UAs) are behind NAT as Kamailio will force the RTP to go via RTPProxy. However, when the public user has a public IP, it will fail to establish the RTP to a user who registered on Kamailio coming from the internal firewall interface. The reason is that, as I don't do NAT between internal and DMZ firewall interfaces, Kamailio will not RTPRroxy in between the UAs because, from the way Kamailio detects NAT, they are not behind NAT. So the public user UA tries to reach a private IP address. If the "inside" user tries to call a public user (a user on the Internet with a public IP), it works. Yes, I could do NAT in between the internal and DMZ firewall interfaces. However, this would force all RTP traffic of my UAs at the LAN go to Kamailio RTPProxy, an bad effect that I would like to avoid. So my question is: what would be the best approach to solve this issue using Kamailio and RTPProxy in such scenario? Cheers! Moacir

2 4

siptrace buffer
by Martin Formanko 08 Nov '12

08 Nov '12

Hi, do you know what happens if the database for siptrace is not reachable ? I assume siptrace is non-blocking in new versions of kamailio, but is there any buffer? Or insert queries are discarded when timed out ? Mino

2 1

Gateway failover on slow call setup
by Spencer Thomason 08 Nov '12

08 Nov '12

Hello, I'm trying to address a problem where sometimes our upstream providers have slow call setup times. We use drouting with multiple providers for each route to handle failover in the event of an outage but in this case, we get a 100 Trying but sometimes a 18[03] response takes 4-5 seconds or we never get it because the caller cancels the request. What would be a "safe" amount of time to allow a PSTN gateway to reply with a 180/183? What I'm trying to avoid is a scenario where the PSTN legitimately takes a long time to setup (i.e. an international call to some cell phone providers) and we needlessly failover which will actually increase the post dial delay for the user. My thought is this for example: route[PSTN] { t_set_fr(1000); # set fr timer to 1 second so we failover quickly if no provisional response t_on_reply("PSTN_REPLY"); # Routing logic } onreply_route[PSTN_REPLY] { if (status=~"1[0-9][0-9]") { t_set_fr(4000); # set fr timer to 4 seconds } if (status=~"18[03]") { t_reset_fr(); # reset timer if we get a 180 or 183 } } Also, if we failover because of slow call setup for a particular route, I'd like to temporarily avoid that gateway for that route (but only that route). I'm curious if anyone has any pointers or how they solved this. Thanks, Spencer

2 1

Adding Users not excepting email option
by kellon＠sis-grenada.com 08 Nov '12

08 Nov '12

I have attempted to add users as per documentation. kamctl add <username> <password> <email> However, Kamailio responds with the following: -- command 'add|passwd|rm' - manage subscribers add <username> <password> .......... add a new subscriber (*) passwd <username> <passwd> ......... change user's password (*) rm <username> ...................... delete a user (*) Is the option to add user with email address etc still available in the current release of 3.3 Thanks Kellon

2 1

Kamailio not responding
by kellon＠sis-grenada.com 07 Nov '12

07 Nov '12

Thanks Anton, Fred, Jeremy thanks for the suggestions. The problem was the firewall. I don't know why I didn't think of it. I learnt one other thing also the use of kamctl monitor. Just two other problems at the moment. I'll start a new thread for them

1 0

TLS Certificate Verification Issue
by Kamal Palei 07 Nov '12

07 Nov '12

Dear All I have modified kamailio,cfg and compiled all the modules with TLS enabled, and able to bring up the kamailio proxy properly. Kamailio proxy will receive the REGISTER message from endpoints in UDP , and want to send this REGISTER message to another intermediate proxy in TLS. For this purpose, I have added few lines in kamailio.cfg file as below. I have created the certificates, private keys as explained by README file in kamailio-3.1.5/modules/tls/ path. if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:443"); exit(); } Looks like this is taking effect. When Kamailio receives REGISTER message it is trying to do handshake with intermediate proxy. I used wireshark to see the handshake messages. 1. From Kamailio proxy, a TCP SYNC message is going to intermediate proxy. 2. intermediate proxy sends SYNC + ACK 3. Kamailio sends CLIENT HELLO 4. intermediate proxy sends SERVER HELLO, CERTIFICATE and SERVER HELLO DONE 5. The Kamailio sends ALERT (Level: Fatal, Description: Unknown CA) ---> IS something going wrong here.............. 6. Then Kamailio sends FIN + ACK Can somebody please let me know why the certificate verification fails (I get this log in console). How can I put a work around to avoid certification verification failure. Best Regards kamal

2 10

Kamailio not responding
by kellon＠sis-grenada.com 07 Nov '12

07 Nov '12

I have recently installed kamailio on CentOS 6.3 and configured a couple of SIP phone Linksys and Grandstream to test inercommunication. The SIP-Server/ Kamailio server ha started and is listen on port 5060. The UA are sending registered message to the SIP Server but there is no response from the SIP server. I left the syslog at default but not seeing any messages on the /var/log/messages. Can someone tell me if I have missed some configuration information. Note: I registered two users 101 and 202 using the command kamctl add 101 101. Is it possible to check whether the Users added have been added to the system??. and does anyone have any idea how to to troubleshoot a response from the Server. I am using Wireshark to monitor communication between between UA and SIP Server

4 3

Problem with my kamailio installation
by Ramazan Yilmaz 06 Nov '12

06 Nov '12

Hi, I installed kamailio 3.2.x a few times before and they were really fine. I decided to install a fresh one and followed the instructions on http://www.kamailio.org/wiki/install/3.3.x/git. However there is a problem: clients cannot register to kamailio. Each registration attempt produces the following error in the syslog: Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [tcp_main.c:3578]: BUG: handle_ser_child: fd -1 for 0 (pid 32414) Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [pass_fd.c:103]: ERROR: recv_all: 1st recv on 56 failed: Bad file descriptor Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [tcp_main.c:3363]: ERROR: handle_tcp_child: read from tcp child 0 (pid 0, no 0) Bad file descriptor [9] Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 17 on 56: Bad file descriptor (9) Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: ERROR: <core> [tcp_main.c:4006]: ERROR: send2child: send_fd failed for 0x7f510bc41588 (flags 0x4018), fd 17 What could be the problem? Any suggestion? In case you need, the server is ubuntu server 11.10, and I installed including mysql and tls modules.

2 15

ERROR: presence not working correctly with Kamailio 3.3.2-1.1 and Postgresql 9.1
by Robert Verspuy 05 Nov '12

05 Nov '12

Hi All, I've got a problem with Kamailio 3.3.2-1.1 on CentOS 6.3 with locally postgresql 8.4.13-1.el6_3 libraries, connected to a remote postgresql91-9.1.6-1PGDP.rhel6 server The presence module tries to insert a record in the presentity table, with some xml in the body column. But the body value in the postgresql table is not represented as a string '<?xml ...etc' But encodes as '\x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462...etc' This gives problems when the presense module is trying to send out a notify based on the value in the database. With debugging, this gives the error: Entity: line 1: parser error : Start tag expected, '<' not found x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462d38273f3e3c The \x in front of the hex encodes string is not properly interpreted by postgresql or kamailio. The body column of the table presentity is a bytea column, and I believe postgresql 9 outputs these bytea columns a bit differently then postgresql 8. When I try to run the sql statement: insert into presentity (domain,username,event,etag,sender,body,received_time,expires ) values ('newsip.lifexs.nl','00086','presence','a.1352107949.18632.27.0','','<?xml version=''1.0'' encoding=''UTF-',1343534532,1345213723) On the postgresql 9.1 server locally, I still get the '\x3c3f786d6c... etc' value in the database. The bytea_output setting in postgresql 9 is now standard set to 'hex' in stead of escape. I've changed the bytea_output setting in postgresql.conf to 'escape', and then the presence is working without any issues. So maybe the database module of postgresql has to be changed to pick up these bytea encodings properly? Because i think kamailio should work correctly with the default postgresql settings. With kind regards, Robert Verspuy -- *Exa-Omicron* Eenspan 8-K 3897 AL Zeewolde http://exa.nl

2 1

Kamailio sending re invite to the next proxy
by Maurice Vistan 05 Nov '12

05 Nov '12

Hi All, I am using kamailio as dispatcher and set up kamailio just to send 200 okay back for the method register, I would like check if its normal for the kamailio send another invite on the next proxy listed on my dispatcher.lst after the first proxy process the invite and auth. Below is how the call is being process. user kamailio 1proxy 2proxy user2 invite |----------> | 100 | <---------- | invite | | ---------> | 100 |<--------- | auth |<-----------| auth | <------------------------| ack |------------->| ---------->| Invite |-------------->| --------------------------> | then process Is there any I can limit this? or it is realy the behavior? Thanks a lot! Maurice

2 2

ERROR: presence not working correctly with Kamailio 3.3.2-1.1 and Postgresql 9.1
by Robert Verspuy 05 Nov '12

05 Nov '12

Hi All, I've got a problem with Kamailio 3.3.2-1.1 on CentOS 6.3 with locally postgresql 8.4.13-1.el6_3 libraries, connected to a remote postgresql91-9.1.6-1PGDP.rhel6 server The presence module tries to insert a record in the presentity table, with some xml in the body column. But the body value in the postgresql table is not represented as a string '<?xml ...etc' But encodes as '\x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462...etc' This gives problems when the presense module is trying to send out a notify based on the value in the database. With debugging, this gives the error: Entity: line 1: parser error : Start tag expected, '<' not found x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462d38273f3e3c The \x in front of the hex encodes string is not properly interpreted by postgresql or kamailio. The body column of the table presentity is a bytea column, and I believe postgresql 9 outputs these bytea columns a bit differently then postgresql 8. When I try to run the sql statement: insert into presentity (domain,username,event,etag,sender,body,received_time,expires ) values ('newsip.lifexs.nl','00086','presence','a.1352107949.18632.27.0','','<?xml version=''1.0'' encoding=''UTF-',1343534532,1345213723) On the postgresql 9.1 server locally, I still get the '\x3c3f786d6c... etc' value in the database. The bytea_output setting in postgresql 9 is now standard set to 'hex' in stead of escape. I've changed the bytea_output setting in postgresql.conf to 'escape', and then the presence is working without any issues. So maybe the database module of postgresql has to be changed to pick up these bytea encodings properly? Because i think kamailio should work correctly with the default postgresql settings. With kind regards, Robert Verspuy -- *Exa-Omicron* Eenspan 8-K 3897 AL Zeewolde http://exa.nl

1 0

Entries in PUA List and RLS Watchers List
by Sangeeta Shah 03 Nov '12

03 Nov '12

Hello All, I am hoping the authors of the RLS/PUA modules can answer this question since I am not sure what's going on. I have the rls module enabled, with the following config spec: # ------rls module params ------ modparam("rls", "db_url", DBURL) modparam("rls", "db_mode", 2) modparam("rls", "integrated_xcap_server", 1) modparam("rls", "to_presence_code" ,10) modparam("rls", "server_address", "sip:rls@ip:5060") #!endif and modparam("pua_reginfo", "default_domain", "ip") modparam("pua_reginfo", "server_address", "sip:reginfo@ip") modparam("pua", "db_mode", 2) So for both I have DB_MODE set to DB ONLY. For whatever reason, even though all I am doing is bringing up my client I see over 250+ entries in the rls_watchers table for the same watcher and over 100 entries in the pua list table for the same presentity? Anyone have any idea what would be triggering this. There is definitely not anymore more than the normal messaging going on between the client and the server. Only error I see in the syslog: Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11745]: ERROR: db_mysql [km_dbase.c:122]: driver error on query: Duplicate entry '1350937406' for key 'expires_idx' Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11744]: DEBUG: presence [subscribe.c:1216]: subs->contact= sip:rls@ip:5060 - len = 25 Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11761]: DEBUG: <core> [db_val.c:117]: converting STRING [8475551001] Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11746]: DEBUG: <core> [parser/msg_parser.c:626]: method: <SUBSCRIBE> Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11745]: ERROR: <core> [db_query.c:210]: error while submitting query Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11744]: DEBUG: rls [rls.h:241]: did= 41826031568472f9-11752@127.0.0.1;533cb9e91f4b999cf76861cbb9ed54ed-93a3;a6a1c5f60faecf035a1ae5b6e96e979a-464d Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11761]: DEBUG: <core> [db_val.c:117]: converting STRING [10.50.251.12] Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11746]: DEBUG: <core> [parser/msg_parser.c:628]: uri: <sip:8475551001@ip> Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11745]: ERROR: pua [pua_db.c:1149]: DB insert failed -- Sangeeta Shah

5 18

Error while "Make All"
by Techie Sup 02 Nov '12

02 Nov '12

Hello, I followed the instruction mentioned in Kamailio website. When I do a make all, I get the error. Please let me know how to fix this. I am new to Kamailio. ------------------- Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found CC (gcc) [M app_mono.so] app_mono_api.o In file included from app_mono_api.c:37: app_mono_api.h:27:26: warning: mono/jit/jit.h: No such file or directory app_mono_api.h:28:39: warning: mono/metadata/environment.h: No such file or directory In file included from app_mono_api.c:37: app_mono_api.h:35: error: expected specifier-qualifier-list before ‘MonoDomain’ app_mono_api.c:39:39: warning: mono/metadata/mono-config.h: No such file or directory app_mono_api.c:49: error: expected specifier-qualifier-list before ‘MonoDomain’ app_mono_api.c: In function ‘sr_mono_load_script’: app_mono_api.c:93: error: ‘sr_mono_load_t’ has no member named ‘next’ app_mono_api.c: In function ‘mono_sr_init_load’: app_mono_api.c:139: warning: implicit declaration of function ‘mono_config_parse’ app_mono_api.c:141: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:148: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:148: warning: implicit declaration of function ‘mono_jit_init’ app_mono_api.c:149: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:157: error: ‘sr_mono_load_t’ has no member named ‘assembly’ app_mono_api.c:157: warning: implicit declaration of function ‘mono_domain_assembly_open’ app_mono_api.c:157: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:158: error: ‘sr_mono_load_t’ has no member named ‘assembly’ app_mono_api.c:162: error: ‘sr_mono_load_t’ has no member named ‘next’ app_mono_api.c: In function ‘mono_sr_initialized’: app_mono_api.c:208: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c: In function ‘app_mono_exec’: app_mono_api.c:230: error: ‘sr_mono_env_t’ has no member named ‘msg’ app_mono_api.c:237: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:246: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:246: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:247: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:256: warning: implicit declaration of function ‘mono_jit_exec’ app_mono_api.c:256: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:256: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:257: warning: implicit declaration of function ‘mono_environment_exitcode_get’ app_mono_api.c:261: warning: implicit declaration of function ‘mono_jit_cleanup’ app_mono_api.c:261: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c: In function ‘app_mono_run’: app_mono_api.c:282: error: ‘sr_mono_env_t’ has no member named ‘msg’ app_mono_api.c:284: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:284: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:285: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:285: error: ‘sr_mono_load_t’ has no member named ‘assembly’ app_mono_api.c:286: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:291: warning: implicit declaration of function ‘mono_domain_set’ app_mono_api.c:291: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:298: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:298: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c: At top level: app_mono_api.c:310: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘*’ token app_mono_api.c:314: error: expected declaration specifiers or ‘...’ before ‘MonoString’ app_mono_api.c: In function ‘sr_mono_log’: app_mono_api.c:316: warning: implicit declaration of function ‘mono_string_to_utf8’ app_mono_api.c:316: error: ‘text’ undeclared (first use in this function) app_mono_api.c:316: error: (Each undeclared identifier is reported only once app_mono_api.c:316: error: for each function it appears in.) app_mono_api.c:316: warning: assignment makes pointer from integer without a cast app_mono_api.c:318: warning: implicit declaration of function ‘mono_free’ app_mono_api.c: At top level: app_mono_api.c:321: error: expected ‘)’ before ‘*’ token app_mono_api.c:328: error: expected ‘)’ before ‘*’ token app_mono_api.c:338: error: expected ‘)’ before ‘*’ token app_mono_api.c:408: error: ‘sr_mono_api_version’ undeclared here (not in a function) app_mono_api.c:410: error: ‘sr_mono_err’ undeclared here (not in a function) app_mono_api.c:411: error: ‘sr_mono_dbg’ undeclared here (not in a function) app_mono_api.c:412: error: ‘sr_mono_modf’ undeclared here (not in a function) app_mono_api.c: In function ‘sr_mono_load_class_core’: app_mono_api.c:420: warning: implicit declaration of function ‘mono_add_internal_call’ app_mono_api.c: At top level: app_mono_api.c:427: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘*’ token app_mono_api.c:478: error: expected ‘)’ before ‘*’ token app_mono_api.c:530: error: expected ‘)’ before ‘*’ token app_mono_api.c:579: error: expected ‘)’ before ‘*’ token app_mono_api.c:632: error: expected ‘)’ before ‘*’ token app_mono_api.c:679: error: expected ‘)’ before ‘*’ token app_mono_api.c:729: error: ‘sr_mono_pv_gets’ undeclared here (not in a function) app_mono_api.c:729: error: initializer element is not constant app_mono_api.c:729: error: (near initialization for ‘_sr_M_export_pv[0].method’) app_mono_api.c:730: error: ‘sr_mono_pv_geti’ undeclared here (not in a function) app_mono_api.c:730: error: initializer element is not constant app_mono_api.c:730: error: (near initialization for ‘_sr_M_export_pv[1].method’) app_mono_api.c:731: error: ‘sr_mono_pv_seti’ undeclared here (not in a function) app_mono_api.c:731: error: initializer element is not constant app_mono_api.c:731: error: (near initialization for ‘_sr_M_export_pv[2].method’) app_mono_api.c:732: error: ‘sr_mono_pv_sets’ undeclared here (not in a function) app_mono_api.c:732: error: initializer element is not constant app_mono_api.c:732: error: (near initialization for ‘_sr_M_export_pv[3].method’) app_mono_api.c:733: error: ‘sr_mono_pv_unset’ undeclared here (not in a function) app_mono_api.c:733: error: initializer element is not constant app_mono_api.c:733: error: (near initialization for ‘_sr_M_export_pv[4].method’) app_mono_api.c:734: error: ‘sr_mono_pv_is_null’ undeclared here (not in a function) app_mono_api.c:734: error: initializer element is not constant app_mono_api.c:734: error: (near initialization for ‘_sr_M_export_pv[5].method’) app_mono_api.c:750: error: expected ‘)’ before ‘*’ token app_mono_api.c:801: error: expected ‘)’ before ‘*’ token app_mono_api.c:850: error: expected ‘)’ before ‘*’ token app_mono_api.c:895: error: expected ‘)’ before ‘*’ token app_mono_api.c:930: error: ‘sr_mono_hdr_append’ undeclared here (not in a function) app_mono_api.c:930: error: initializer element is not constant app_mono_api.c:930: error: (near initialization for ‘_sr_M_export_hdr[0].method’) app_mono_api.c:931: error: ‘sr_mono_hdr_remove’ undeclared here (not in a function) app_mono_api.c:931: error: initializer element is not constant app_mono_api.c:931: error: (near initialization for ‘_sr_M_export_hdr[1].method’) app_mono_api.c:932: error: ‘sr_mono_hdr_insert’ undeclared here (not in a function) app_mono_api.c:932: error: initializer element is not constant app_mono_api.c:932: error: (near initialization for ‘_sr_M_export_hdr[2].method’) app_mono_api.c:933: error: ‘sr_mono_hdr_append_to_reply’ undeclared here (not in a function) app_mono_api.c:933: error: initializer element is not constant app_mono_api.c:933: error: (near initialization for ‘_sr_M_export_hdr[3].method’) make[1]: *** [app_mono_api.o] Error 1 make: *** [modules] Error 1 Regards Selvam

3 4

Kamailio WebSocket support
by Owen Lynch 02 Nov '12

02 Nov '12

Hi, I see Peter Dunkley's websocket work is not in the latest 3.3 release (3.3.2). What is the road map for this? Thanks, Owen Lynch

2 1

Re: [SR-Users] Kamailio and sylkserver integration
by Vijay Thakur 01 Nov '12

01 Nov '12

Thanks for your reply. Will you please explain that how can i forward calls from kamailio to sylkserver. Regards, VJ++ Message: 7 Date: Wed, 31 Oct 2012 08:26:50 +0100 From: "Olle E. Johansson"<oej(a)edvina.net> Subject: Re: [SR-Users] Kamailio and sylkserver integration To: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing List"<sr-users(a)lists.sip-router.org> Message-ID:<D7F883EA-4231-4D21-A79F-BB3449F972B1(a)edvina.net> Content-Type: text/plain; charset=us-ascii 31 okt 2012 kl. 06:49 skrev Vijay Thakur<vijay.thakur(a)loopmethods.com>: > Dear Sir, > > Is there any one in the list who have set up (integrated) Kamailio with sylkserver for conference. There is very less available information on the internet on this topic. All the required changes are made in sylkserver config files. > Kindly advise for required changes in Kamailio. As far as I understand, Sylkserver is just a SIP server. Forward calls from kamailio to sylkserver using any of the forwarding commands. /O -- * Olle E. Johansson -oej(a)edvina.net * Kamailio & SIP Masterclass Miami FL December 2012 *http://edvina.net/training/

2 1

Two kamailio servers using same DB but different version table
by Uri Shacked 31 Oct '12

31 Oct '12

Hi, can two kamailio servers, one running 3.2 and the other 3.3 run using the same DB? one will turn to version table X and the other to version table Y (i am not using the modules that the DB table changed... BR, Uri

2 1

statefull proxy Q
by Alex Solt 31 Oct '12

31 Oct '12

Hi, Is there any function in kamailio that could differentiates between a new incoming INVITE message and a re-INVITE message for an existing active call? Thanks, AS

3 6

2600Hz selects Kamailio
by Olle E. Johansson 31 Oct '12

31 Oct '12

Friends, The 2600Hz project that builds an API-driven telephony service have selected to switch to Kamailio as a SIP proxy for their SBC. They blog about the switch, comparing Kamailio with their previous choice and coming to a conclusion: "To our minds, a more active development community means that other users are likely to help solve bottlenecks for us, so we spend less time worrying about our SBC, and more time writing fun new applications! Conclusion: We’re switching to Kamailio because: • Many of our Major Clients already use it • The Naming Conventions are more sensible • The Pace of Innovation on Kamailio is faster" Join me in welcoming 2600Hz to the Kamailio project! /O Links: * http://www.kamailio.org/w/2012/10/2600hz-selects-kamailio/ * http://blog.2600hz.com/post/34639934434/our-other-border-brother-kamailio

1 0

Kamailio and sylkserver integration
by Vijay Thakur 31 Oct '12

31 Oct '12

Dear Sir, Is there any one in the list who have set up (integrated) Kamailio with sylkserver for conference. There is very less available information on the internet on this topic. All the required changes are made in sylkserver config files. Kindly advise for required changes in Kamailio. With Regards, Vijay Thakur

2 1

additional user data for SIP gateway usage?
by Jeremy Ardley 31 Oct '12

31 Oct '12

Hi, I'm after advice. I'm developing a script to provide a degree of PBX functionality for a small to medium office using Kamailio & SEMS. This includes mapping internal extensions to external PSTN numbers via the SIP trunk provider. The incoming mapping is easy using the db_alias functionality. The outgoing mapping is more difficult as the SIP trunk provider requires both the Full National Number and an account number matching the FNN in the outgoing From header. I'd appreciate suggestions on the best way to store, retrieve, and manage the additional information required for outgoing PSTN calls. I think avpops will work but is there some other way to do this? Thanks,

2 1

SIP connection tracking
by Daniil Baturin 31 Oct '12

31 Oct '12

Hi all, We have two softswitch hosts that don't support TLS. We have a peer that requires TLS, they terminate all our connections on a single address. We want to use kamailio as a proxy for it. Could you point me to the documentation and/or working examples that describe how to implement connection tracking to ensure replies to connections originated from host A are relayed back to host A, and connections from host B get back to host B? Thanks in advance. -- #!/usr/bin/env perl @a=split(//, "daniil @ baturin . org" );# Daniil Baturin @b=split(//,q/Px%!+o0Q6lh*7dp$.@8#%|y{/);while($i<24){$_.= chr((ord(@b[$i])-ord(@a[$i])+62)%94+32);$i++};print"$_\n"#

2 1

Re: [SR-Users] Blacklist initialization
by Javi Gallart 30 Oct '12

30 Oct '12

Hi Olle What about doing the init stuff in event-route[htable:mod-init] ? Regards Javi ------Original Message------ From: Olle E. Johansson Sender: sr-users-bounces(a)lists.sip-router.org To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List ReplyTo: SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing List Subject: [SR-Users] Blacklist initialization Sent: Oct 30, 2012 21:58 Hi! I am looking for a way to add addresses to the IP blacklist in Kamailio at startup. Is that possible? I know I can do it with sercmd, but I fail to find a documented way to do it in the configuration... Must be missing something simple. Thank you for your help! /O _______________________________________________ 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

1 0

Blacklist initialization
by Olle E. Johansson 30 Oct '12

30 Oct '12

Hi! I am looking for a way to add addresses to the IP blacklist in Kamailio at startup. Is that possible? I know I can do it with sercmd, but I fail to find a documented way to do it in the configuration... Must be missing something simple. Thank you for your help! /O

1 0

Default database named now kamailio
by Daniel-Constantin Mierla 30 Oct '12

30 Oct '12

Hello, just short note to let everyone know that the default database for Kamailio is now named 'kamailio' - the change happened over the weekend as part of migrating from last bits from openser to kamailio namespace. Also, the default usernames and passwords to access the database are now: - kamailio with password kamailiorw for read-write operations - kamailioro with password kamailioro for read-only operations If you set db_url parameters in the config file, there should be no impact on existing configurations. You also have to make sure you set the right value in kamctlrc file if you update kamctl. This change impacts development version (to be released as 3.4.0). Two other components are in the to-do list for this change: perl and snmpstats Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Question about acc module
by Thorsten Willand 30 Oct '12

30 Oct '12

3 2

Unknown status in presence_dialog
by David 29 Oct '12

29 Oct '12

Hello, I am using the presence_dialoginfo module on a stand alone presence server. I have this : if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; If the server does not have any status information, it will send out a blank notify with a 0 content length. Is there anyway to tell Kamailio to send an xml doc with status "terminated" if the server does not know the status that was requested by the SUBSCRIBE? Thanks, David Server : version: kamailio 3.2.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 14:19:23 Apr 19 2012 with gcc 4.4.3

1 0

siptrace vs. sipcapture
by Mino Haluz 29 Oct '12

29 Oct '12

Hi, we are thinking about switching from siptrace to sipcapture + HEP encapsulation. Do you know some reasons why we should switch to sipcapture or why we should not ? ;) We have multiple SIP proxies and Cisco gateways. In earlier versions of kamailio was siptrace blocking, that's why we started to looking for some other capturing mechanism. I think now siptrace is buffered, so we are hesitating .. Mino

2 1

rtpproxy ipv4/ipv6 bridging question
by Juha Heinanen 29 Oct '12

29 Oct '12

is it possible to make rtpproxy to include both ipv4 and ipv6 address in sdp? if not, i don't see how ipv4/ipv6 bridging can be made to work when $rd is domain name. -- juha

1 0

NEW: new module: xhttp_pi
by Ovidiu Sas 28 Oct '12

28 Oct '12

Hello all, A new module is available in trunk. It provides a web provisioning interface: http://kamailio.org/docs/modules/devel/modules/xhttp_pi.html A sample config example is provided in the source tree: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ex… An additional xml file is required for web config framework (what to configure, where and how). A sample xml file is provided in the source tree for dispatcher and dialplan tables: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ex… When kamailio is up and running, a web provisioning interface for managing db records will be available. No supplementary web server needs to be installed in order to manage the db records. Also, it provides a tight control over what can be modified and how. Testing and feedback is appreciated. Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

2 1

web homer not displaying data
by Christopher Holland 26 Oct '12

26 Oct '12

I have sipcapture and siptrace setup on the same server using kamailio 3.4.0-dev4 and webhomer 3.3.0 setup to capture sip traffic from a mirriored port on the LAN. Interface ETH1 is to capture and ETH0 is for managment. The sipcapture database is getting data put into it. I can log into webhomer, but see no data in the graphes nor can I search. I have double checked my database permissions and they seem to be set up correctly. webserver is set up with: Apache/2.2.16 (Debian) PHP5 MYSQL Json phpmyadmin Here is a copy of my kamailio.cfg and configuration.php rev_dns=no auto_aliases=no debug=1 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=5 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* IP and port for HEP capturing) */ listen=udp:127.0.0.1:9060 mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" loadmodule "pv.so" loadmodule "db_mysql.so" loadmodule "sipcapture.so" loadmodule "textops.so" loadmodule "siptrace.so" # ----- mi_fifo params ----- ####### Routing Logic ######## modparam("sipcapture", "db_url", "mysql://openser:******@localhost/homerdb") modparam("sipcapture", "capture_on", 1) /* activate HEP capturing */ modparam("sipcapture", "hep_capture_on", 1) /* IP to listen. Port/Portrange apply only on mirroring port capturing */ modparam("sipcapture", "raw_socket_listen", "10.0.0.0:5060-5080") /* Name of interface to bind on raw socket */ modparam("sipcapture", "raw_interface", "eth1") /* activate IPIP capturing */ #modparam("sipcapture", "raw_ipip_capture_on", 1) /* My table name*/ modparam("sipcapture", "table_name", "sip_capture") /* children for raw socket */ modparam("sipcapture", "raw_sock_children", 4) /* insert delayed */ #modparam("sipcapture", "db_insert_mode", 1) /* activate monitoring/mirroring port capturing. Linux only */ modparam("sipcapture", "raw_moni_capture_on", 1) /* Promiscious mode RAW socket. Mirroring port. Linux only */ modparam("sipcapture", "promiscious_on", 1) /* activate Linux Socket Filter (LSF/BPF) on mirroring interface. Linux only */ #modparam("sipcapture", "raw_moni_bpf_on", 1) /* sipcapture node */ modparam("sipcapture", "capture_node", "homer01") # check IP and port of your capture node modparam("siptrace", "duplicate_uri","sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag",22) modparam("siptrace", "trace_on", 1) # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { sip_capture(); drop; } onreply_route { sip_capture(); drop; } ################################## configuration.php <?php /*********************************************************************************/ /* Access db of homer */ define('HOST', "localhost"); define('PORT', 3306); define('USER', "webhomer"); define('PW', "******"); define('DB', "web_homer"); /* Homer connection * this user must have the same password for all Homer nodes * please define all your nodes in homer_nodes table */ define('HOMER_HOST', "localhost"); /* DEFAULT. Don't forget insert this host to your DB nodes table */ define('HOMER_PORT', 3306); define('HOMER_USER', "openser"); define('HOMER_PW', "******"); define('HOMER_DB', "homerdb"); define('HOMER_TABLE', "sip_capture"); /*********************************************************************************/ /* webHomer Settings * Adjust to reflect your system preferences */ define('PCAPDIR',"/var/www/webhomer/tmp/"); define('WEBPCAPLOC',"/webhomer/tmp/"); define('APIURL',"http://localhost"); define('APILOC',"/webhomer/api/"); /* INCLUDE preferences */ include("preferences.php"); ?>

2 2

Message Routing To/From Gateway
by Ed Brady 26 Oct '12

26 Oct '12

Hi, I am trying to route SIP messages to a gateway. I am using SIP MESSAGE over UDP as well as the MSILO module for storing messages for users that aren't currently logged in. I have read through the Dynamic Routing module info but am not able to determine the best way to accomplish the routing I require. Our scenario is outlined below. User A sends a message addressed to "sip:12345678900@mydomain.com" Kamailio at mydomain.com checks if 12345678900 is a registered user in the system. If not, the message header is translated to "sip:12345678900@gatewaydomain..com" and forwarded. Incoming messages need to be accepted from their IP as well. It would also be useful if the address came as "sip:2345678900@mydomain.com" could be translated to "sip:*1*2345678900@gatewaydomain.com". Is there any built in functionality I can configure for this, or will this need to be scripted? Thanks, Ed

2 9

[OT] the role of SBCs
by Daniel-Constantin Mierla 26 Oct '12

26 Oct '12

Hello, based on the outcome of the discussion carried in the thread: http://lists.sip-router.org/pipermail/sr-users/2012-August/074480.html I am looking again the clarify some aspects out there in the VoIP world. So, as mainly dealing with proxy/sip signaling deployments, it's very often to be the first one hit by support issues claiming things don't work. Then you investigate and end up in conclusions like in the above thread: "The problem was at the SBC, __where I did not expect it__." The underlined part heats me up a bit, because I never understood from where it comes this perception that SBC is a MUST-TO-HAVE and the PERFECT (never mistaken or breaking things) node in a VoIP networks. To some extent, the SBC is just a very costly SIP ALG, and a SIP ALG is there to break the things. I don't want to start like a flame war, but is it something that I am obviously missing in regards to what benefits a SBC can bring? I see only inconveniences: - another point of failure - it is a b2bua, therefore very unlikely to offer the same performances of a proxy - if transcoding is needed, a media server can be used behind the proxy, properly protected of attacks by the proxy and eventually deployed as a farm load balanced by the proxy - if topology hinding is wanted in the b2bua fashion (not the proxy fashion with encoding headers), then the b2bua can be behind the proxy, properly protected of attacks by the proxy and eventually deployed as a farm load balanced by the proxy - nat traversal was solved long time ago in proxy environment, being scalable by deploying a farm of rtp proxy I don't want to go to other features, including the transport layer, it's a clear win of the proxy in my experience (ok, being deep involved in this project). Then, what makes the SBC so desirable in many companies/voip deployments? If any SBC user here that can share, what was the reason to buy such a device? Any conceptual functionality that cannot be achieved with the proxy as the first hop in front of the (wild) clients? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

10 14

MSRP: Generation of REPORTs on the relay
by Peter Dunkley 26 Oct '12

26 Oct '12

Hello, One of my colleagues at Crocodile has been doing some testing with the msrp module. With the auth change that was made yesterday, and the updated configuration I put in the README example it seems to mostly work. However, there is one set of scenarios relating to failure REPORTs that don't seem to be possible with the current module. RFC 4976 section 6.4.1 paragraph 3 states: If the Failure-Report header is "yes" or "partial", and if there is a problem processing the SEND request or if an error response is received for that SEND request, then the relay MUST respond with an appropriate error response in a REPORT back to the original source of the message. This means that when a SEND request that contains a Failure-Report: header with value of "yes" or "partial" is relayed by Kamailio we need to maintain some state so that when the TCP send fails, or we receive a failure response from the next hop, Kamailio can generate and send a REPORT. At the moment I don't think the msrp module has these capabilities. If the module is getting some re-work before the next release it would be good to get this in there. I am quite happy to help out in any way I can, but I am not sure of the best way to proceed with this. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Kamailio on Twitter: @kamailioproject
by Olle E. Johansson 26 Oct '12

26 Oct '12

The marketing arm of the Kamailio project has without further discussion opened up a Twitter account for Kamailio, to support our web site news, G+ and Facebook pages. Feel free to contribute with case stories and news we can publish on the web site and follow up in the social media flow. The twitter account is @kamailioproject - please follow it. And yes, Kamailio is THE best thing since the invention of sliced bread! This week's EXTRAORDINARY offer: Download TWO licenses of Kamailio for the same licensing cost as we've been offering for ONE license! And get an unlimited number of user licenses for your Kamailio SIP accounts! Kamailio greetings! Have a great SIP weekend! /Marketing

1 0

MSRP relay authentication
by Peter Dunkley 25 Oct '12

25 Oct '12

Hi, MSRP AUTH requests must be authenticated using HTTP Digest authentication. Part of the concatenated data when doing this authentication is the method name. Because of the way MSRP requests are formatted, the part of the request-line that would contain the method name in HTTP or SIP requests is always MSRP. The MSRP method name (AUTH in this case) is at the end of the request-line. When Kamailio converts an MSRP request to SIP it has a method name of MSRP. This means that when Kamailio authenticates an MSRP request it uses "MSRP" as the method name, not the actual MSRP method name (AUTH). Is this correct? Should MSRP requests be authenticated with a method name of "MSRP" (which kind-of makes sense as this is the string at the start of the MSRP request line - which is where the method names are in HTTP and SIP) or should the MSRP method name of "AUTH" be used? If it is the later, does anyone know of an easy way to add this to Kamailio? Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

2 4

dns queries on ipv6 addresses
by Juha Heinanen 25 Oct '12

25 Oct '12

from the mailing list archives i have learned that dns_try_ipv6=yes in config file is supposed to prevent dns query on ipv6 address. if i have it "no", i get to syslog: Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: INFO: Request URI is <sip:+35832345679@[2002:c062:670a::10]:4050;transport=udp> Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: ERROR: tm [ut.h:314]: ERROR: uri2dst: failed to resolve "[2002:c062:670a::10]" :unresolvable A or AAAA request (-7) Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches and sip proxy naturally does not send anything out. if i have it "yes", request is sent out properly, but still sip proxy makes dns queries on the ipv6 address: Oct 24 14:08:35 siika /usr/sbin/sip-proxy[18196]: INFO: Request URI is <sip:+35832345679@[2002:c062:670a::10]:4050;transport=udp> Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail why is that? how to completely prevent dns queries on ipv6 addresses? -- juha

6 16

rtpproxy delete fails
by Juha Heinanen 25 Oct '12

25 Oct '12

i made rtpproxy test in setup where two sip phones have registered the same AoR test(a)test.fi. one is behind nat and the other is not. when i call this AoR, my sip proxy executes rtpproxy_manage("FROW3"); in branch route of the branch that is behind nat. syslog shows: Oct 19 15:49:59 siika /usr/sbin/sip-proxy[18596]: INFO: Calling rtpproxy_manage on INVITE <sip:test@192.168.0.31:5074;transport=tcp> Oct 19 15:49:59 siika mediaproxy-ng[12832]: Got valid command from udp:127.0.0.1:48714: 18596_16 US ptonjivixvatrhz(a)siika.tutpro.com;z9hG4bKqxklorkm 192.98.103.10 8000 wnzdf;1 Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com] Creating new call Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Opened ports 50104/50105 for RTP Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Opened ports 50106/50107 for RTP Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Returning to SIP proxy: 18596_16 50104 192.98.103.10 4 then the uas hehind nat replies with 480 and sip proxy executes in onreply route rtpproxy_manage("FROW3"); that fails and i get to syslog: Oct 19 15:50:15 siika mediaproxy-ng[12832]: Got valid command from udp:127.0.0.1:56183: 18594_12 D ptonjivixvatrhz(a)siika.tutpro.com;z9hG4bKqxklorkm wnzdf shsqn Oct 19 15:50:15 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com] Tags didn't match for delete message, ignoring looks like this mismatch is due to the missing ";1" in from tag param that was there when rtpproxy_manage was executed on invite. if my theory is correct, why is ";1" missing? if not correct, what did i do wrong? -- juha

4 17

udp transport in rtpproxy sockets
by Juha Heinanen 25 Oct '12

25 Oct '12

rtpproxy module readme gives impression that udp transport is the only possibility: 4.1. rtpproxy_sock (string) Definition of socket(s) used to connect to (a set) RTPProxy. It may specify a UNIX socket or an IPv4/IPv6 UDP socket. if that is the case, what is the justification for it, i.e, why is tcp (and better yet tls) transport not supported? -- juha

3 3

rtpproxy module flags 1 and 2 question
by Juha Heinanen 25 Oct '12

25 Oct '12

i'm new to rtpproxy (have been using mediaproxy for years) and now try to understand if it would make sense to switch over. rtpproxy readme has this on flags value 1: + 1 - append first Via branch to Call-ID when sending command to rtpproxy. ... This is especially useful if you have serially forked call scenarios where rtpproxy gets an "update" command for a new branch, and then a "delete" command for the previous branch, which would otherwise delete the full call, breaking the subsequent "lookup" for the new branch. how it would be possible in SERIAL forked scenario to get anything for a new branch before previous branch has first failed and gone? -- juha

4 7

Manually add Via Header
by Vassilis Radis 25 Oct '12

25 Oct '12

I have a situation where a far end SIP provider doesn't behave properly when sending 487 replies. The scenario is this: I have a registered user calling into my kamailio which ,using lcr module, routes the call to a SIP provider. When the caller Cancels the call, my kamailio forwards the cancel message to the provider (along with a 200 Cancelling message to the caller). The problem is with the 487 sent by the provider to my kamailio: It does not contain all the two Via headers. It only contains the via header that names my kamailio. So when kamailio gets the 487, it removes its Via header and sends the 487 back to the caller, which now doesnt find any via header. Note that the far end provider, correctly sends all the Via Headers in other replies (183 etc). So now I am trying to "intercept" the 487 from this specific provider in the onreply route, and patch it with the header that the provider should have included. How can I do it? The problem is twofold: 1.I need to detect which header I should manually add (maybe store the Via Header set from the initial invites or the 183 replies, and get it from there? .I dont know how to do that) 2.I need to add it. (How can I manually add a Via header? .This seems easier Can anyone help me, maybe with another solution?

4 5

ip address fields
by Juha Heinanen 25 Oct '12

25 Oct '12

lcr_gw table has: ip_addr varchar(47) trusted table has: src_ip varchar(50) address table has: ip_addr varchar(48) why is it that they are not all the same? also, why the common length is not 8 * 4 + 7 = 39 (http://en.wikipedia.org/wiki/IPv6#Address_format)? -- juha

4 19

Kamailio & OMA Presence Compliance
by Thomas Hoffman 24 Oct '12

24 Oct '12

My colleague and I have been trying to test Kamailio configured as a Presence Server using OMA standards/messages. It appears that Kamailio does not support the OMA standards as it pertains to Presence. This is primarily in the area of the XCAP messages. Can someone please confirm if this is the case? If this is the case, it is on the roadmap to support it? If not, we'd like to know that too and we'll continue trying to get it to work. Thank you, Tom ______________________________________________________________________________________________________________________________ This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify so to the sender by e-mail and delete the original message. In such cases, please notify us immediately at sanchar-sadhan(a)infinite.com . Further, you are not to copy, disclose, or distribute this e-mail or its contents to any unauthorized person(s) .Any such actions are considered unlawful. This e-mail may contain viruses. Infinite has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachments. Infinite reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infinite e-mail system. ***INFINITE******** End of Disclaimer********INFINITE********

1 0

RFC5626 SIP Outbound Support of Flow-Timer Header
by Jonathan Hunter 24 Oct '12

24 Oct '12

Hi, Testing with RFC5626, can you confirm that the Flow-Timer header field is supported and configurable with Kamailio 3.3 ? Thanks Jon

2 3

sercmd compiles and installs as kamcmd for kamailio
by Daniel-Constantin Mierla 24 Oct '12

24 Oct '12

Hello, short note to let everyone know that sercmd tool compiles and installs as kamcmd for flavour kamailio as of today in the master git branch. There was a conflict when trying to install both flavours from packages, because of overlapping sercmd file. I added a basic manpage for the tool, a recommendation for each binary app in many linux distros. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Registering Phones
by Re Alvarez 24 Oct '12

24 Oct '12

Hi I am setting Kamailio as SBC in my organization using Freeswitch as PBX. The integration between the two systems is working perfectly and now I am trying to add some CISCO 7970 phones to the mix. Here's a summary of the setup 1. Kamailio has 2 NICs (1 public, 1 private), RTPproxy is running 2. Freeswitch is on private network. 3. All the CISCO phones are also on the private network. We will never have a SIP phone outside our network, My question, in this case. Should I register my phones to Kamailio or Freeswitch. I am new to Kamailio and still getting down with the SIP concepts. I would prefer to register the phones with Freeswitch and only want Kamailio to get involved when calls are leaving/entering the network. If two users, registered to the same freeswitch server call each other, Kamailio should not have to do anything (ideally, not sure if its possible with SBC setup). Thanks Re

1 0

getting siptrace error on start up
by Christopher Holland 23 Oct '12

23 Oct '12

getting this error: kamailio: ERROR: <core> [modparam.c:151]: set_mod_param_regex: parameter <trace_to_database> not found in module <siptrace> parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 63, column 46: Can't set module parameter I have set this to modparam("siptrace", "trace_to_database", "0") in my .cfg /I didn't compile siptrace when I compiled kamailio. I don't know how to add a module after compiling and installing kamailio. I haven't been able to find any documentaion on how to add a module after installing only compiling before installing. Thanks all for your help Christopher /

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Kamailio Hosted Solutions
by Satyajeet Thokchom 23 Oct '12

23 Oct '12

I am looking for hosted Kamailio Solutions for my VOIP calling app in android and iphone. I have the app ready and have even setup kamailio server in our rackspace cloud but there is many problems like one way calling, noisy calls, no call connection etc. (application VOIP library , we are using IMSdroid and Idoubs , based on doubango SIP) I am looking for a hosted solutions affordable for me - with a support for 300 concurrent calls at a time. Best Regards, Satyajeet Thokchom Business Manager, NGBM Solutions Ltd Phone: India:+91-11-25541320 Mobile: 91 7428565264 Website: www.ngbm.net Web Support: www.ngbmsolutions.com Note: "This e-mail is confidential and may also be privileged. If you are not the intended recipient, please notify us immediately; you should not copy or use it for any purpose, nor disclose its contents to any other person"

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turning off rev. dns
by Christopher Holland 22 Oct '12

22 Oct '12

I'm setting up Homer with kamailio for sip capturing. I've got it all set up and am getting this error. I have one interface for packet capture on a mirroired port and one interface for managment. I'm thinking that I have to only have kamailio listen on the packet capture interface and not the managment interface. Would that resolve this issue? loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(1832) WARNING: <core> [socket_info.c:1406]: WARNING: fix_hostname: could not rev. resolve 192.168.111.88 Listening on udp: 127.0.0.1:5060 udp: 192.168.111.88:5060 Aliases: udp: localhost:5060 config file ok, exiting... sense it's a local address it's not possible to do a rev. resolve of that address. Is there a way to turn off the rev. resolve. I've looked at this in the trouble shooting faq. How do I avoid Reverse-DNS-Lookups? A: Simply not use src_ip = "11.22.33.44" and use instead src_ip = 11.22.33.44! We had in our installation some serious trouble with requests coming from systems, where the hostname could not be reverse resolved (the DNS-Server responsible for this IP-Range was poorly configured and would not answer at all).

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Remote Commands
by Ed Brady 22 Oct '12

22 Oct '12

Hi, I am trying to get a company who handles our billing and account provisioning connected to our Kamailio instance. I understand that XMLRPC is likely the best method for doing this, but I am unable to find direct commands equivalent to "kamctl add" or "kamctl rm". Is there a resource that lists the MI or XMLRPC commands, or do I need to create even simple commands? Thanks, Ed

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Hangup call on leg B side
by pars3c 22 Oct '12

22 Oct '12

Hi , i don't know how to realize this: A --- INVITE---->proxy ----INVITE----> B <--- 200 OK ---BYE----> ---INVITE ----->C <--- 200 OK proxy <--- 200 OK -------ACK ---> -------ACK ---> When B answer the call, the proxy should immediately hang up the call and make a new invite. For user A is always the same call. I tried this : onreply_route { ################################ connect if (is_method("INVITE") && status == "200") { dlg_bye("callee"); drop(); } } But it doesn't work. When arrive 200 , the proxy drop the message to the user a , wait for a while, and then send a cancel to b (it should send a BYE, instead send a CANCEL, probably because it not have a ACK). How can I try this? Thanks

2 1

Kamailio 3.1.5 crashing during TLS connection setup
by Kamal Palei 22 Oct '12

22 Oct '12

Dear All I have setup Kamailio 3.1.5 with TLS enabled. Whenever I receive the REGISTER request from endpoint, trying to forward to next proxy. For that I have added below code in kamailio.cfg * if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:5061"); exit(); } *I see once REGISTER received at Kamailio proxy, it is sending TCP synch packet to next proxy, next proxy sends back a synch+reset packet. Then kamailio proxy is crashing with log as below. *root@B2BUA:/usr/local/src/scripts# 9(2347) ERROR: <core> [tcp_main.c:4139]: connect 115.114.48.75:5061 failed 9(2347) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting * The complete log is given as below. Please let me know how to get out of this issue. I used -m 64 option while starting proxy, but same result. Thanks, Kamal, NECS, Bangalore * 0(2336) INFO: <core> [tcp_main.c:4730]: init_tcp: using epoll_et io watch method (config) 0(2338) INFO: usrloc [hslot.c:53]: locks array size 512 0(2338) INFO: tls [tls_init.c:511]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1c 10 May 2012" (0x1000103f), kerberos support: off, compression: on 0(2338) INFO: tls [tls_init.c:519]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1c 10 May 2012" (0x1000103f), kerberos support: off, zlib compression: off compiler: gcc -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -Wa,--noexecstack -DL_ENDIAN -DTERMIO -O3 -fomit-frame-pointer -Wall -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM 0(2338) INFO: tls [tls_init.c:373]: tls: init_tls: disabling compression... 0(2338) WARNING: tls [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 5242880 and 2621440 bytes 0(2338) INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 5242880 0(2338) INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 2621440 0(2338) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2338) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 0(2338) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2338) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 0(2338) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 0(2338) INFO: tls [tls_domain.c:176]: TLSs<default>: tls_method=9 0(2338) INFO: tls [tls_domain.c:188]: TLSs<default>: certificate='/etc/certs/pocserver.com/cert.pem' 0(2338) INFO: tls [tls_domain.c:195]: TLSs<default>: ca_list='(null)' 0(2338) INFO: tls [tls_domain.c:202]: TLSs<default>: crl='(null)' 0(2338) INFO: tls [tls_domain.c:206]: TLSs<default>: require_certificate=0 0(2338) INFO: tls [tls_domain.c:213]: TLSs<default>: cipher_list='(null)' 0(2338) INFO: tls [tls_domain.c:220]: TLSs<default>: private_key='/etc/certs/pocserver.com/key.pem' 0(2338) INFO: tls [tls_domain.c:224]: TLSs<default>: verify_certificate=0 0(2338) INFO: tls [tls_domain.c:227]: TLSs<default>: verify_depth=9 0(2338) INFO: tls [tls_domain.c:544]: TLSs<default>: No client certificate required and no checks performed 0(2338) INFO: tls [tls_domain.c:176]: TLSc<default>: tls_method=9 0(2338) INFO: tls [tls_domain.c:188]: TLSc<default>: certificate='(null)' 0(2338) INFO: tls [tls_domain.c:195]: TLSc<default>: ca_list='(null)' 0(2338) INFO: tls [tls_domain.c:202]: TLSc<default>: crl='(null)' 0(2338) INFO: tls [tls_domain.c:206]: TLSc<default>: require_certificate=1 0(2338) INFO: tls [tls_domain.c:213]: TLSc<default>: cipher_list='(null)' 0(2338) INFO: tls [tls_domain.c:220]: TLSc<default>: private_key='(null)' 0(2338) INFO: tls [tls_domain.c:224]: TLSc<default>: verify_certificate=1 0(2338) INFO: tls [tls_domain.c:227]: TLSc<default>: verify_depth=9 0(2338) INFO: tls [tls_domain.c:529]: TLSc<default>: Server MUST present valid certificate 2(2340) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 1(2339) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 4(2342) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 0(2338) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 6(2344) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 3(2341) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 7(2345) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 6(2344) INFO: ctl [io_listener.c:224]: io_listen_loop: using epoll_et io watch method (config) root@B2BUA:/usr/local/src/scripts# 9(2347) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 8(2346) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled root@B2BUA:/usr/local/src/scripts# 9(2347) ERROR: <core> [tcp_main.c:4139]: connect 115.114.48.75:5061 failed 9(2347) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting 0(2338) ALERT: <core> [main.c:742]: child process 2347 exited by a signal 6 *

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