sr-users November 2012

sr-users@lists.kamailio.org

85 participants
126 discussions

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

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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 >

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sr-users November 2012