sr-users

sr-users@lists.kamailio.org

42 participants
37517 discussions

id column i tables
by Kristian F. Høgh 02 Feb '17

02 Feb '17

Hi list, We provision our kamailio database from an external system. In the future we would like to provision from two different systems. To prevent the two systems from touching each others data, we would like to create views from the two tables. provision system 1 would write to table1 provision system 2 would write to table2 kamailio will read a view over the two tables. The tables we will create views over are: - address - aliases - dispatcher - uri - usr_preferences Now to my question :-) Does kamailio use the id field in any of the tables above? The id's are auto-incremented, and we might get conflicts between table1 and table2 *Kristian Høgh* Telefon: 4422 8822 support(a)uni-tel.dk[1] Gydevang 19 | 3450 Allerød www.uni-tel.dk [2] -------- [1] mailto:support@uni-tel.dk [2] http://www.uni-tel.dk

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Disabling DNS SRV records checks
by Ahmed Fouad 01 Feb '17

01 Feb '17

Hi, I'm running kamailio version 4.1 and having the following DNS related configuration use_dns_failover = off dns_try_naptr = no dns_udp_pref = -1 dns_try_ipv6 = off dns_retr_time=1 dns_retr_no=1 dns_servers_no=1 dns_use_search_list=no But still am seeing SRV record checks for my domains which I like to completely disable these checks and only use A records only. x.x.x.x.57828 > 8.8.8.8.53: 7417+ SRV? _sip._udp.subdomain.com 8.8.8.8.53 > x.x.x.x.57828: 7417 NXDomain 0/1/0 (129) I can't find a DNS option to disable SRV checks. Can you please advise ?

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could not make ul_rm_contact work
by Aqs Younas 01 Feb '17

01 Feb '17

Greetings list, I am trying to remove a specific contact from memory but always resulted in vain. root@debian:/usr/local/kamailio/sbin# ./kamctl ul show Domain:: location table=1024 records=1 max_slot=1 AOR:: 1040(a)192.168.0.101 Contact:: sip:1040@192.168.0.102:36619;rinstance=1dfaed088d873a18;transport=tcp Q= Expires:: 2416 Callid:: _3595KDc1PinhUGFpSf4Dg.. Cseq:: 2 User-agent:: Z 3.9.32144 r32121 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:192.168.0.101:5060 Methods:: 5087 Ruid:: uloc-58911db0-f45-1 Reg-Id:: 0 Last-Keepalive:: 1485905402 Last-Modified:: 1485905402 . /kamctl fifo ul_rm_contact location 1040(a)191.168.0.101 sip:1040@192.168.0.102:36619;rinstance=1dfaed088d873a18;transport=tcp 404 AOR not found Tried another command root@debian:/usr/local/kamailio/sbin# ./kamctl fifo ul_rm location 1040(a)192.168.0.101 root@debian:/usr/local/kamailio/sbin# ./kamctl ul show Domain:: location table=1024 records=1 max_slot=1 AOR:: 1040(a)192.168.0.101 Contact:: sip:1040@192.168.0.102:36619;rinstance=1dfaed088d873a18;transport=tcp Q= Expires:: deleted Callid:: _3595KDc1PinhUGFpSf4Dg.. Cseq:: 2 User-agent:: Z 3.9.32144 r32121 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:192.168.0.101:5060 Methods:: 5087 Ruid:: uloc-58911db0-f45-1 Reg-Id:: 0 Last-Keepalive:: 1485905402 Last-Modified:: 1485905402 And this. . /kamctl ul rm 1040(a)192.168.0.1 sip:1040@192.168.0.102:36619 ;rinstance=1dfaed088d873a18;transport=tcp 404 AOR not found Can someone help me know to remove a contact from memory? Any pointer is much appreciated. Best Regards.

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sdp_with_transport
by M. Salman 01 Feb '17

01 Feb '17

Hi Guys, Just wanted to clarify the following case: what should be result of sdp_with_transport("RTP/SAVPF") on line: m=audio 10231 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 I am having a weird behavior for two different versions of Kamailio. I hope I am doing something wrong. kamailio 4.3.2 => sdp_with_transport("RTP/SAVPF") = true kamailio 4.4.5 => sdp_with_transport("RTP/SAVPF") = false -- Regards M. Salman VoIP Professional

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Re: [SR-Users] sr-users Digest, Vol 140, Issue 30
by Andrew Miller 01 Feb '17

01 Feb '17

Is it too late to request a place for the meal at FOSDEM? I would love to come. Andy Miller On 30 Jan 2017 11:00, <sr-users-request(a)lists.sip-router.org> wrote: > 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. Diversion Header - relay messed up (Roman Dissauer) > 2. Re: dlg_set_timeout bye to websockets client (Switch168) > 3. Presence Module do not load db_url (Vu. Minh Cao) > 4. Re: FOSDEM 2017 (Giacomo Vacca) > 5. Re: Replace old registration record with new one while > keeping single AOR per contact (Federico Cabiddu) > 6. Re: Replace old registration record with new one while > keeping single AOR per contact (Aqs Younas) > 7. Re: mhomed opened sockets (Diego Nadares) > 8. Re: [sr-dev] FOSDEM 2017 (Daniel-Constantin Mierla) > 9. Re: FOSDEM 2017 (Ivan Todorov) > 10. Re: [Kamailio-Business] [sr-dev] FOSDEM 2017 (Alexandr Dubovikov) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 29 Jan 2017 18:42:23 +0100 > From: Roman Dissauer <roman(a)dissauer.net> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] Diversion Header - relay messed up > Message-ID: <DB3FC45E-ACCE-426E-B662-031909F6BA40(a)dissauer.net> > Content-Type: text/plain; charset=utf-8 > > When I get an INVITE with Diversion Header the Request is forwarded > without Diversion Header and the Request User is taken from Diversion User. > Problem is that on the Destination Host I cannot get original Request User > what is the intended destination! > Is this intended behaviour? How can I change this behaviour? > > I already tried to delete the Diversion Header on request_route but this > didn’t change the behaviour. > > Thanks in andvance for your help! > Roman > > Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as > Carrier IP > > Incoming INVITE: > > INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0. > Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843. > From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. > To: <sip:436761234567@x.x.x.x>. > Call-ID: 906143813_44460603(a)y.y.y.y. > CSeq: 28100 INVITE. > Max-Forwards: 19. > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, > NOTIFY,PRACK,UPDATE,OPTIONS. > Accept: application/sdp, application/isup, application/dtmf, > application/dtmf-relay, multipart/mixed. > Contact: <sip:+43987654321@y.y.y.y:5060>. > P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. > Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no; > reason=unconditional; counter=1. > Supported: timer,100rel. > Session-Expires: 1800. > Min-SE: 90. > Content-Length: 260. > Content-Disposition: session; handling=required. > Content-Type: application/sdp. > ... > > > Relayed INVITE: > > INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0. > Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>. > Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>. > Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582. > 4fc0216dbecafde29127db502993222c.0. > Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch= > z9hG4bK02B7871714d9345f843. > From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. > To: <sip:436761234567@x.x.x.x>. > Call-ID: 906143813_44460603(a)y.y.y.y. > CSeq: 28100 INVITE. > Max-Forwards: 18. > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, > NOTIFY,PRACK,UPDATE,OPTIONS. > Accept: application/sdp, application/isup, application/dtmf, > application/dtmf-relay, multipart/mixed. > Contact: <sip:+43987654321@y.y.y.y:5060>. > P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. > Supported: timer,100rel. > Session-Expires: 1800. > Min-SE: 90. > Content-Length: 274. > Content-Disposition: session; handling=required. > ... > > > > > > > > ------------------------------ > > Message: 2 > Date: Sun, 29 Jan 2017 18:21:37 +0000 > From: Switch168 <team(a)switch168.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org>, miconda(a)gmail.com > Subject: Re: [SR-Users] dlg_set_timeout bye to websockets client > Message-ID: > <CAJ4KJRjeZT++vYWkfnY10WnkWztpVtpSV700TW0ebh > 7z+jRAEg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi Daniel, > > > For anyone else who might run into this issue I found > https://github.com/kamailio/kamailio/issues/85 to be related to mine. And > its a good a starting point. > > Cheers > Andrew > > > > > On Fri, Jan 27, 2017 at 10:47 AM Switch168 <team(a)switch168.com> wrote: > > > HI Daniel, > > > > modparam("dialog", "dlg_flag", 4) modparam("dialog", "send_bye", 1) > > modparam("dialog", "timeout_noreset", 1) > > > > So regular bye's by useragent i can succesfully relay to this > > <sip:6gjlmali@ec2c66jsa0ei.invalid;transport=ws> > > style of uri by using this snippet below > > > > # Handle requests within SIP dialogs > > route[WITHINDLG] { > > if(is_method("BYE")) { > > xlog("DEBUG: Received BYE"); > > route(NATDETECT); > > loose_route(); > > dlg_manage(); > > t_check_trans(); > > handle_ruri_alias(); > > rtpengine_delete(); > > route(RELAY); > > exit; > > } > > ... > > > > But the byes that are sent out by dlg_set_timeout("$var(timer)") gives > out > > error cannot resolve > > the random.uri and I don't know how to fix the bye before it get sent > out. > > > > Thanks > > Andrew > > > > > > > > > > > > > > > > > > On Fri, Jan 27, 2017 at 12:15 AM, Daniel-Constantin Mierla < > > miconda(a)gmail.com> wrote: > > > > Hello, > > > > isn't the dialog module setting the right value there? What function are > > you using to update the contact? > > > > Cheers, > > Daniel > > > > On 27/01/2017 06:01, Andrew Tan wrote: > > > > Hello, > > > > Just wondering if it's possible to intercept the bye message that > > dlg_set_timeout sends out to do some nat_helper function to fix it so the > > bye can be sent to random.invalid uris. > > > > I know there is the edge proxy outbound module but wondering if I can do > > it without. > > > > Regular bye between 2 clients i was able intercept and fix with nat > helper > > but I dont know how to fix the byes that is sent out from dlg_set_timeout > > function. > > > > Thanks in advance! > > Andrew > > > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > listsr-users@lists.sip-router.orghttp://lists.sip-router. > org/cgi-bin/mailman/listinfo/sr-users > > > > > > -- > > Daniel-Constantin Mierlawww.twitter.com/miconda -- > www.linkedin.com/in/miconda > > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - > www.asipto.com > > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > > > > _______________________________________________ > > 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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Re: [SR-Users] FOSDEM 2017
by Hugh Waite 01 Feb '17

01 Feb '17

Hi all, Please count me and one other in for the meal on Saturday. See you in Brussels! Hugh Waite On 30 Jan 2017 11:00 a.m., <sr-users-request(a)lists.sip-router.org> wrote: 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. Diversion Header - relay messed up (Roman Dissauer) 2. Re: dlg_set_timeout bye to websockets client (Switch168) 3. Presence Module do not load db_url (Vu. Minh Cao) 4. Re: FOSDEM 2017 (Giacomo Vacca) 5. Re: Replace old registration record with new one while keeping single AOR per contact (Federico Cabiddu) 6. Re: Replace old registration record with new one while keeping single AOR per contact (Aqs Younas) 7. Re: mhomed opened sockets (Diego Nadares) 8. Re: [sr-dev] FOSDEM 2017 (Daniel-Constantin Mierla) 9. Re: FOSDEM 2017 (Ivan Todorov) 10. Re: [Kamailio-Business] [sr-dev] FOSDEM 2017 (Alexandr Dubovikov) ---------------------------------------------------------------------- Message: 1 Date: Sun, 29 Jan 2017 18:42:23 +0100 From: Roman Dissauer <roman(a)dissauer.net> To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Diversion Header - relay messed up Message-ID: <DB3FC45E-ACCE-426E-B662-031909F6BA40(a)dissauer.net> Content-Type: text/plain; charset=utf-8 When I get an INVITE with Diversion Header the Request is forwarded without Diversion Header and the Request User is taken from Diversion User. Problem is that on the Destination Host I cannot get original Request User what is the intended destination! Is this intended behaviour? How can I change this behaviour? I already tried to delete the Diversion Header on request_route but this didn’t change the behaviour. Thanks in andvance for your help! Roman Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as Carrier IP Incoming INVITE: INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0. Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 19. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no; reason=unconditional; counter=1. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 260. Content-Disposition: session; handling=required. Content-Type: application/sdp. ... Relayed INVITE: INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0. Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>. Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>. Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582. 4fc0216dbecafde29127db502993222c.0. Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 18. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 274. Content-Disposition: session; handling=required. ... ------------------------------ Message: 2 Date: Sun, 29 Jan 2017 18:21:37 +0000 From: Switch168 <team(a)switch168.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org>, miconda(a)gmail.com Subject: Re: [SR-Users] dlg_set_timeout bye to websockets client Message-ID: <CAJ4KJRjeZT++vYWkfnY10WnkWztpVtpSV700TW0ebh7z+jRAEg(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi Daniel, For anyone else who might run into this issue I found https://github.com/kamailio/kamailio/issues/85 to be related to mine. And its a good a starting point. Cheers Andrew On Fri, Jan 27, 2017 at 10:47 AM Switch168 <team(a)switch168.com> wrote: > HI Daniel, > > modparam("dialog", "dlg_flag", 4) modparam("dialog", "send_bye", 1) > modparam("dialog", "timeout_noreset", 1) > > So regular bye's by useragent i can succesfully relay to this > <sip:6gjlmali@ec2c66jsa0ei.invalid;transport=ws> > style of uri by using this snippet below > > # Handle requests within SIP dialogs > route[WITHINDLG] { > if(is_method("BYE")) { > xlog("DEBUG: Received BYE"); > route(NATDETECT); > loose_route(); > dlg_manage(); > t_check_trans(); > handle_ruri_alias(); > rtpengine_delete(); > route(RELAY); > exit; > } > ... > > But the byes that are sent out by dlg_set_timeout("$var(timer)") gives out > error cannot resolve > the random.uri and I don't know how to fix the bye before it get sent out. > > Thanks > Andrew > > > > > > > > > On Fri, Jan 27, 2017 at 12:15 AM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > > Hello, > > isn't the dialog module setting the right value there? What function are > you using to update the contact? > > Cheers, > Daniel > > On 27/01/2017 06:01, Andrew Tan wrote: > > Hello, > > Just wondering if it's possible to intercept the bye message that > dlg_set_timeout sends out to do some nat_helper function to fix it so the > bye can be sent to random.invalid uris. > > I know there is the edge proxy outbound module but wondering if I can do > it without. > > Regular bye between 2 clients i was able intercept and fix with nat helper > but I dont know how to fix the byes that is sent out from dlg_set_timeout > function. > > Thanks in advance! > Andrew > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router. org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > _______________________________________________ > 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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any public databases or schemas for dial plans?
by Daniel Pocock 31 Jan '17

31 Jan '17

Hi, Are there any public databases of dial plan information? Are there any schemas (e.g. XML / XSD) useful for describing arbitrary dial plans from different carriers around the world? The type of things that I'm interested in: - being able to translate any local number to E.164 (if possible) - knowing the expected length (or range of lengths) for different numbers - being able to recognize numbers that can't be translated to E.164 (e.g. the emergency numbers) - being able to classify specific types of number that usually have a carrier-specific or country-specific meaning (e.g. classifying numbers as directory assistance, customer service, emergency) I'm familiar with Google's libphonenumber, although it is not a database, rather, it is a library that has dial plan logic in Java. It supports many of the things I want for numbers that can be translated to E.164 and country-level dial plans, but not local dial plans (e.g. dialing a London number without the 020 area code). Regards, Daniel

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Using DNS for Service Discovery with Kamailio
by Casco, Alessio 31 Jan '17

31 Jan '17

Hello Guys! I'd like to be able to use SRV records inside dispatcher groups, so I can automatise the discovery of new backends on my network using DNS I've tried with the following formats into the dispatcher.list file: $group _sip._udp.sip-voice_backends $group sip-voice_backends and none looks to be working, running the kamailio_ctl dispatcher.list won't show me the group set with SRV records. ( I also try NAPTR records, with no luck as well.) * Does the dispatcher module support SRV or NAPTR records? (maybe my format is not correct) * If not, do you have any other idea on what I could do to achieve what I'd like to do? Thanks Alessio

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RFC: Kamailio tutorials/cookbooks/docs - wiki or markdown
by Daniel-Constantin Mierla 31 Jan '17

31 Jan '17

Hello, I want to start a discussion about some of the kamailio tutorials from the wiki, mainly those related to core cookbook, variables and transformations. They are now in dokuwiki at: - https://www.kamailio.org/wiki/#cookbooks The wiki served us pretty well so far, however the contributions there are not that active. From time to time we still get spammers, even we added restriction that only registered and authenticated users can write. I am thinking that maybe people don't like creating yet another account just add some example or rephrase for clarity. On the other hand, there are over 400 registered users, many of them I expect to be either made because if misunderstanding the purpose or by spammers that guessed the (sip) captcha (plenty of random usernames). Besides that I think the wiki adds overhead when releasing a new version as each time we have to copy and paste old version content to a new set of pages and update the version number. With the above in mind, I wonder if won't be better to store those tutorials in markdown format and use mkdocs.org or gitbook (https://github.com/GitbookIO/gitbook) tools to generate the html and host it in kamailio.org. The wiki will still be used to index them. Among the benefits I see: * store in github.com (a new repo, like: kamailio-docs) and all devs can contribute directly, other users can make pull requests * create branches for each major version of kamailio and backport across them with git whenever is applicable * the edit can be done directly via github.com website (similar to the wiki right now) and even read it directly from github in html transformed from markdown by github * one can get them offline by cloning the git repo, use them or enhance when offline Many if the tutorials will stay in the wiki, but I think those that are related to documenting per kamailio version should be migrated to gihub storage + markdown. If there is a strong pro opinion for this change, I can try to convert one of the cookbooks very soon so people can feel better the difference. If we get to a consensus, then I think we can have those tutorials in the new format for v5.0. Should anyone have comments, other suggestions or improvements, let's put them on debate. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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serial forking calls using TSILO
by Tomas Zanet 30 Jan '17

30 Jan '17

Hello, I'm using Kamailio 4.4.4 with TSILO module in order to support Push Notifications used by our voip app on Apple ios10. So far, everything works fine: Kamailio can get an incoming call, suspend it, then send push notification with an external script, receive a new app registration and then call it after resuming the invite. To do so, thanks to Mr, Cabiddu, I used all the functions described here http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-… Right now, I would like to support serial forking calls using push notification because it was already supported with legacy voip. To do so, I used failure_route function where, after getting the call destinations and setting them into an avp, I set the new SIP request uri $ru. Here is my failure_route: failure_route[MANAGE_FAILURE] route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) t_reply("404","Not found"); exit; #!endif if (is_avp_set("$avp(group_members_db)")) $ru ="sip:"+$avp(group_members_db)+"@"+$fd; $avp(group_members_db) = $null; route(LOCATION); exit; I am troubleshooting this scenario and this is what I see: 1) Kamailio receives incoming call 2) Suspend it 3) Send Push notification to account1 4) Kamailio receives account1 registration -> INVRESUME route is now called 5) Then call account1 6) Nobody answers the call 7) failure_route[MANAGE_FAILURE] is now called -> set $ru with next sipaccount: account2 8) suspend invite 9) send push notification to account2 10) account2 sends its sip registration -> INVRESUME route is now called 11) Kamailio calls account1 instead of account2 -> this is my issue On step 10: despite account2 is registered, I checked it using "kamctl ul show", the second call is forwarded all'account1 and not to account2. I also checked the TSILO logs and everything seems ok. Here is the logs: First call: suspended transaction [53945:1648394094] asterisk => account1 htable key value [53945:1648394094] resuming trasaction [53945:1648394094] account1 53945:1648394094) second call: suspended transaction [53945:1648394094] asterisk => account2 htable key value [53945:1648394094] resuming trasaction [53945:1648394094] account2 53945:1648394094) In order to find out the issue I put some xlogs, printing $ru value: what I see is the $ru value is set correctly on failure_route but as soon as the t_continue is called, the ru overwritten back to account1 Can anyone address me to find out the solution? Thanks in advance T. Tomas Zanet Software Design Department tzanet(a)came.com

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