- sr-users - kamailio.org

Question about setbflag and isbflagset usage
by Володимир Іванець 07 Jun '19

07 Jun '19

Hello everyone! I'm not sure if I correctly understood the description of functions *isbflagset* and *isbflagset* so once again I'm asking for help :) Here is what I do: 1. Asterisk sends INVITE to Kamailio. Kamailio performs lookup of available contacts and then few checks in branch_route. I'm adding *setbflag("17", $T_branch_idx);* there. 2. Later in route[NATMANAGE] I added *if (isbflagset("17", $T_branch_idx)) {* and additional logic. *isbflagset* results in *true* (for same INVITE). 3. Finally one of UACs responds with 200 OK. Asterisk generates ACK packet that gets in route[NATMANAGE]. *isbflagset* results in *false*. So my question is: shouldn't *isbflagset* be *true* for ACK too since it's the same branch or I interpreted these fenctions incorrectly? Thanks a lot!

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Re: [SR-Users] [EXT] Re: ICSCF mysql error
by Bao, Yin 07 Jun '19

07 Jun '19

Yes I think you are right it’s the mismatch of uri. However we don’t know which configuration resulted that mismatch. Here are a bit more details: ICSCF prepared this query: select * from impu where identity=? And it provided sip:+alice@irisims.org as the parameter for the query However the HSS impu table as well as the SIP INVITE only contains: sip:alice@irisims.org I am suspecting the additional + may be causing this mismatch. But where is the + coming from and why ICSCF add this in front of original INVITE line uri? Thanks! Yin From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of YASIN CANER <caner_yaso(a)hotmail.com> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Date: Friday, June 7, 2019 at 2:47 AM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [EXT] Re: [SR-Users] ICSCF mysql error Hello , It is a mismatch inconming uri and your uri in hss. Yasin Caner Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> şunları yazdı (7 Haz 2019 09:28): Hello, not an expert on this particular module, but this is the internal error path from the module that probably caused this error reply: - case RC_IMS_DIAMETER_ERROR_USER_UNKNOWN, will send message with: - #define MSG_604_USER_UNKNOWN "Does not exist anywhere - HSS User Unknown" The status comes probably from the CDP module: diameter_epc_code_result.h 57: RC_EPC_DIAMETER_ERROR_USER_UNKNOWN = 5001, Maybe this helps. Henning Am 03.06.19 um 06:43 schrieb Trinh, Jenny: Hi, I’m getting the error 604 when ICSCF did a query to mysql and failed with an INVITE. Please see my attached pcap. Would appreciate any feedbacks on what might the issue. We’re using BOGHE client to register and make call with Kamailio IMS 5.2.2 May 22 09:15:10 imskamailio /usr/local/sbin/kamailio[22167]: DEBUG: <core> [core/msg_translator.c:2305]: generate_res_buf_from_sip_res(): copied size: orig:143, new: 56, rest: 630 msg=#012SIP/2.0 604 Does not exist anywhere - HSS User Unknown#015#012Via: SIP/2.0/UDP 0.0.0.0:4060;received=10.0.20.1;branch=z9hG4bK3c92.7402885f6124cf2862bde6f9458080b5.0#015#012Via: SIP/2.0/UDP 10.141.161.85:56962;received=10.141.161.85;branch=z9hG4bK594030036;rport=56962#015#012From: <sip:bob@ims522.irisims.org><mailto:sip:bob@ims522.irisims.org>;tag=594048364#015#012To: <sip:alice@ims522.irisims.org><mailto:sip:alice@ims522.irisims.org>;tag=077cecf844e3c5622044f0dc84818fc7-09fb#015#012Call-ID: 25759faf-6f50-5607-c8f9-162abbbff8cb#015#012CSeq: 19622 INVITE#015#012Server: Kamailio I-CSCF#015#012Content-Length: 0#015#012Warning: 392 10.0.20.1:5060 "Noisy feedback tells: pid=22149 req_src_ip=10.0.20.1 req_src_port=6060 in_uri=sip:alice@ims522.irisims.org<mailto:in_uri=sip:alice@ims522.irisims.org> out_uri=sip:+alice@ims522.ir<mailto:out_uri=sip:+alice@ims522.ir> Thank you _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] Using DNS in Dispatcher for SRV Clarification [SOLVED]
by JR Richardson 07 Jun '19

07 Jun '19

Hi Federico, "use_dns_cache=on" this is on be default. "use_dns_failover=on" this is off by default, changing to "on" did the trick, auto failover to next SRV server is working quickly, as expected now. Thanks for the pointer. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope > > Hi, > if you want failover support, you need to have a couple of core dns > parameters enabled: > https://www.kamailio.org/wiki/cookbooks/5.2.x/core#use_dns_cache > > and > > https://www.kamailio.org/wiki/cookbooks/5.2.x/core#use_dns_failover > > The latter defaults to off. > > Cheers, > > Federico > > On Thu, Jun 6, 2019 at 10:10 PM JR Richardson <jmr.richardson(a)gmail.com> > wrote: > > > Hey All, > > > > I'm in the lab testing kamailio 5.2 with DNS SRV records. The scenario > > is simple, using dispatcher with hard coded IP URIs > > [sip:10.10.10.10:5060] and also a DNS name URI [sip:sip.domain.com]. > > There are 2 SRV servers. > > > > The DNS SRV records are setup and resolving OK, during startup, > > kamailio list the DNS name in the dispatcher list. > > > > During a call scenario, when the dispatcher selects the DNS name URI > > for the next destination, the TM module takes over, resolves the > > _sip._udp.sip.domain.com for both SRV records OK and sends the call to > > the first SRV server in the list. But if I take the first SRV server > > offline, make another call, the TM module does not route to the second > > SRV server until the DNS record timeout is reached, in my lab was set > > for 300 seconds. I decreased the DNS SRV record TTL down to 5 seconds > > with same results above, but 5 seconds lapse when TM routes call to > > next SRV server. > > > > I think this is working as expected but not sure if this is by design > > so my question is, should the TM module route to another SRV server in > > the list in the event of a no response timeout or send the transaction > > back to the dispatcher to select another target? > > > > Should I play around with the TM Module parameter > > "max_noninv_lifetime" to get quicker results or it is better to have > > DNS SRV record TTL set really low? > > > > Thanks. > > > > JR

1 0

reading php script result with exec_avp
by Gertjan Wolzak 07 Jun '19

07 Jun '19

Hello Kamailions, I have the following challenge. I want to create a time dependent callerid, based on the time of a outgoing call a callerid will be set. I have a php script, that provides me an id number which relates to an cli. When I run that php script from the console, the id is shown in the console. When I run the script within the kamailio configuration, exec_avp("../script.php", "$avp(s:cliId)"); ,no output commande is saved in the avp variable. I think I am not presenting the result of the php script correctly. I have tried with an echo $variable; and with and fwrite(STDOUT, $variable); but no luck. I'm not a PHP coder so I am not surprised of the missing luck, but maybe somebody has a golden tip for me? Rgds, Gertjan

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Re: [SR-Users] ICSCF mysql error
by Henning Westerholt 07 Jun '19

07 Jun '19

Hello, not an expert on this particular module, but this is the internal error path from the module that probably caused this error reply: - case RC_IMS_DIAMETER_ERROR_USER_UNKNOWN, will send message with: - #define MSG_604_USER_UNKNOWN "Does not exist anywhere - HSS User Unknown" The status comes probably from the CDP module: diameter_epc_code_result.h 57: RC_EPC_DIAMETER_ERROR_USER_UNKNOWN = 5001, Maybe this helps. Henning Am 03.06.19 um 06:43 schrieb Trinh, Jenny: Hi, I’m getting the error 604 when ICSCF did a query to mysql and failed with an INVITE. Please see my attached pcap. Would appreciate any feedbacks on what might the issue. We’re using BOGHE client to register and make call with Kamailio IMS 5.2.2 May 22 09:15:10 imskamailio /usr/local/sbin/kamailio[22167]: DEBUG: <core> [core/msg_translator.c:2305]: generate_res_buf_from_sip_res(): copied size: orig:143, new: 56, rest: 630 msg=#012SIP/2.0 604 Does not exist anywhere - HSS User Unknown#015#012Via: SIP/2.0/UDP 0.0.0.0:4060;received=10.0.20.1;branch=z9hG4bK3c92.7402885f6124cf2862bde6f9458080b5.0#015#012Via: SIP/2.0/UDP 10.141.161.85:56962;received=10.141.161.85;branch=z9hG4bK594030036;rport=56962#015#012From: <sip:bob@ims522.irisims.org><mailto:sip:bob@ims522.irisims.org>;tag=594048364#015#012To: <sip:alice@ims522.irisims.org><mailto:sip:alice@ims522.irisims.org>;tag=077cecf844e3c5622044f0dc84818fc7-09fb#015#012Call-ID: 25759faf-6f50-5607-c8f9-162abbbff8cb#015#012CSeq: 19622 INVITE#015#012Server: Kamailio I-CSCF#015#012Content-Length: 0#015#012Warning: 392 10.0.20.1:5060 "Noisy feedback tells: pid=22149 req_src_ip=10.0.20.1 req_src_port=6060 in_uri=sip:alice@ims522.irisims.org<mailto:in_uri=sip:alice@ims522.irisims.org> out_uri=sip:+alice@ims522.ir<mailto:out_uri=sip:+alice@ims522.ir> Thank you _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

2 1

Using DNS in Dispatcher for SRV Clarification
by JR Richardson 07 Jun '19

07 Jun '19

Hey All, I'm in the lab testing kamailio 5.2 with DNS SRV records. The scenario is simple, using dispatcher with hard coded IP URIs [sip:10.10.10.10:5060] and also a DNS name URI [sip:sip.domain.com]. There are 2 SRV servers. The DNS SRV records are setup and resolving OK, during startup, kamailio list the DNS name in the dispatcher list. During a call scenario, when the dispatcher selects the DNS name URI for the next destination, the TM module takes over, resolves the _sip._udp.sip.domain.com for both SRV records OK and sends the call to the first SRV server in the list. But if I take the first SRV server offline, make another call, the TM module does not route to the second SRV server until the DNS record timeout is reached, in my lab was set for 300 seconds. I decreased the DNS SRV record TTL down to 5 seconds with same results above, but 5 seconds lapse when TM routes call to next SRV server. I think this is working as expected but not sure if this is by design so my question is, should the TM module route to another SRV server in the list in the event of a no response timeout or send the transaction back to the dispatcher to select another target? Should I play around with the TM Module parameter "max_noninv_lifetime" to get quicker results or it is better to have DNS SRV record TTL set really low? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

2 1

Re: [SR-Users] ICSCF mysql error
by Daniel-Constantin Mierla 06 Jun '19

06 Jun '19

Hello, check you kamailio configuration file, the 604 reply is likely to be generated there -- although I haven't written any of the ims modules, typically the reply generated by kamailio are from configuration file, not hardcoded in the source. Cheers, Daniel On 03.06.19 06:43, Trinh, Jenny wrote: > > Hi, > > I’m getting the error 604 when ICSCF did a query to mysql and failed > with an INVITE. Please see my attached pcap. Would appreciate any > feedbacks on what might the issue. > > > > We’re using BOGHE client to register and make call with Kamailio IMS 5.2.2 > > > > May 22 09:15:10 imskamailio /usr/local/sbin/kamailio[22167]: DEBUG: > <core> [core/msg_translator.c:2305]: generate_res_buf_from_sip_res(): > copied size: orig:143, new: 56, rest: 630 msg=#012SIP/2.0 604 Does not > exist anywhere - HSS User Unknown#015#012Via: SIP/2.0/UDP > 0.0.0.0:4060;received=10.0.20.1;branch=z9hG4bK3c92.7402885f6124cf2862bde6f9458080b5.0#015#012Via: > SIP/2.0/UDP > 10.141.161.85:56962;received=10.141.161.85;branch=z9hG4bK594030036;rport=56962#015#012From: > <sip:bob@ims522.irisims.org>;tag=594048364#015#012To: > <sip:alice@ims522.irisims.org>;tag=077cecf844e3c5622044f0dc84818fc7-09fb#015#012Call-ID: > 25759faf-6f50-5607-c8f9-162abbbff8cb#015#012CSeq: 19622 > INVITE#015#012Server: Kamailio I-CSCF#015#012Content-Length: > 0#015#012Warning: 392 10.0.20.1:5060 "Noisy feedback tells: pid=22149 > req_src_ip=10.0.20.1 req_src_port=6060 > in_uri=sip:alice@ims522.irisims.org out_uri=sip:+alice@ims522.ir > > > > Thank you > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Use dmq module with private and public IP addresses
by Markus Bönke 05 Jun '19

05 Jun '19

Hello NG, I have a question regarding the dmq / dmq_userloc module. In my test setup have 2 Kamailio servers in 2 different data centers. The data centers are interconnected by a crosslink. Each Kamailio server has configured an internal (private) and an external (public) IP address. The internal IPs are routed to use the crosslink. The external IPs are routed through the public internet by different ISPs. The requirement is that the replication should primarily be routed via the crosslink, only if the crosslink is down it should be routed via public internet. As I understand the „server_address“ parameter can have only one IP address. If I configure the public IPs on both sides and one ISP connection gets disconnected, packets will NOT be routed automatically via the crosslink. And the other way around, if I configure the private IPs on both sides and the crosslink gets disconnected, packets will NOT be routed automatically via public internet. Is there any way to setup and use dmq for such a scenario ? Thanks and regards Markus

2 3

IMS - change the transport layer protocol for Diameter based interfaces (especially Rx)
by "björn bauer" 05 Jun '19

05 Jun '19

3 2

RTPEngine and dialog moving.
by Igor Olhovskiy 04 Jun '19

04 Jun '19

Hi! Wanted to clarify some info about rtpengine working within HA-environment. Idea is we have multiple rtpengines under same setid. So, when I'm calling rtpengine_offer() I really don't care which is selected. But than might happen situation, where this call(dialog) is moved to other Kamailio instance (via dlg_db_load_callid()) and from there called rtpengine_delete(). So, my question is, how I can be sure, that delete request will go to same rtpengine instance that was selected in rtpengine_offer() on different server. And where this info is stored, within dialog or within TM module (which is not replicated in any way) Regards, Igor

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Info: app_lua_sr module - old Lua sr library
by Daniel-Constantin Mierla 04 Jun '19

04 Jun '19

Hello, several days ago I split the app_lua module, relocating the code related to old Lua sr library to app_lua_sr module, keeping in this way app_lua smaller and focused to the KEMI code. Over the time, app_lua_sr should be deprecated and removed, as we are sure the functions in Lua sr have an equivalent in Lua KSR. Therefore, if you are using in Lua scripts functions from 'sr' library, consider migrating to KSR as soon as possible. The intermediary solution for a quick upgrade to master version (to be release as v5.3.x) is to load app_lua_sr and update some modparams, like: loadmodule "app_lua.so" loadmodule "app_lua_sr.so" modparam("app_lua_sr", "register", "sl") If you find a function from Lua sr not available in Lua KSR, then report it via mailing lists or bug tracker. An article about this topic was published at: * https://www.kamailio.org/w/2019/05/new-module-app_lua_sr/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Multiple definitions with OR or AND in "ifdef" in kamailio script
by Maxim M 04 Jun '19

04 Jun '19

Is it possible to do something like this in kamailio script?: !!ifdef DEF1 || DEF2 .... !!endif

2 1

empty action lists in route block
by Maryam Baghdadi 04 Jun '19

04 Jun '19

Hello everyone! I tried to use from Lua scripting language in kamailio configuration and rewrite all routes in Lua script. and I export many functions in ims_auth in KEMI and recompile ims_auth module. I have a problem with www_ims_challenge function calling in Lua script. when I define and call a function (REG_MAR_ROUTE) in www_ims_challenge as a route black, I get this error " empty action lists in route block" . why this happens and what's the solution ? thanks in advance

3 2

Removal of Dispatcher
by Colin, Andrew 04 Jun '19

04 Jun '19

I am using the API to remove dispatchers with a simple curl command, is there a way i can make kamailio check that there are sufficient dispatchers available before removing them when the command is run? Example: i am removing asterisk1 out of kamailio dispatcher list, but kamailio needs to ensure that there is atleast 2 or 3 other asterisk servers active in the list before it allows it to be removed. any ideas how i can achieve this?

2 3

AAR change subscription ID
by eyas barhouk 03 Jun '19

03 Jun '19

dear community, when i'm trying to deploy kamailio IMS with PCRF i get ims-qos AAR error due to Subscription id. in ims_qos module i found that i can change the subscription_id type and value as following: * subscription_id to hard code subscription ID for AAR. Used for some broken PCRFs. Leave blank to use default * subscription_id_type to hard code subscription ID type for AAR. Only applicable if subscription_id is set. Set to -1 to use default. This is as per RFC 4006: END_USER_E164 0, END_USER_IMSI 1, END_USER_SIP_URI 2, END_USER_NAI 3, END_USER_PRIVATE 4 but when i changed it there was no change on AAR Subscription-id AVP. any advise about how to set or change the subscription id dears ?? B.R Eyas

1 0

Releasing a new version for Siremis
by Daniel-Constantin Mierla 03 Jun '19

03 Jun '19

Hello, after quite a bit of time, we are considering to release a new version for Siremis - the web management interface for Kamailio. A lot of effort has been pushed to address the issues with PHPv7.x, which deprecated a lot of PHPv5.x API as well as made more strict rules in regards to classes and inheritances. At this stage it should be in good stage, but it would help if others can give it a ride to Siremis master branch: * https://github.com/asipto/siremis Installation guidelines: * https://kb.asipto.com/siremis:install51x:main Testing with both PHP v5 and v7 is appreciated in order to be sure things were not broken for old systems. Any issue found should be reported on github project issue tracker. If all goes fine, in soon we will make the release. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

invalid payload location
by Kelvin Chua 31 May '19

31 May '19

is there a plan to create an sdp sanity functionality? ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location opensips has something similar sipmsg_validate() *'s'* - checks the integrity of the SDP body, if it exists Kelvin Chua

2 2

Table creation error when enabling ims_usrloc_scscf module.
by Ali Shirvani 30 May '19

30 May '19

Hi all, I installed kamailio v5.2.3 from source and IMS modules included. Then I tried to create required database and tables, I removed "dialog" from STANDARD_MODULES and added "ims_dialog ims_charging ims_usrloc_scscf ims_usrloc_pcscf presence" to the EXTRA_MODULES in kamctlrc file. When I issue `<kamailio_home>/sbin/kamctl create` I got following error: ERROR 1062 (23000) at line 39: Duplicate entry 'subscriber' for key 'table_name_idx' ERROR: Creating extra tables failed at ims_usrloc_scscf! Should I exclude an other module from STANDARD_MODUELS? Regards, Ali

2 1

turn on DEBUG level debugging for ICSCF module
by Trinh, Jenny 30 May '19

30 May '19

Hi, How do I turn on DEBUG level debugging for ICSCF module? Thanks Jenny

2 1

app_java.so: undefined symbol: JNI_CreateJavaVM please help
by Renato Gallo 30 May '19

30 May '19

gcj is not a requirement since 2018 [ https://www.codesd.com/item/undefined-symbol-jni-createjavavm-on-linux.html | https://www.codesd.com/item/undefined-symbol-jni-createjavavm-on-linux.html ] all details are in the bug ticket at the link: [ https://github.com/kamailio/kamailio/issues/1969 | https://github.com/kamailio/kamailio/issues/1969 ] please help Renato Gallo System Engineer sede legale e operativa: Via Privata Cefalonia, 14 - 20156 - Milano (MI) Tel. +39 02 - 87049490 Fax +39 02 - 48677349 Mobile. +39 342 - 6350524 Wi | FreeNumbers: https://freenumbers.way-interactive.com Wi | SMS: https://sms.way-interactive.com Wi | Voip: https://voip.way-interactive.com Asterweb: http://www.asterweb.org Le informazioni contenute in questo messaggio e negli eventuali allegati sono riservate e per uso esclusivo del destinatario. Persone diverse dallo stesso non possono copiare o distribuire il messaggio a terzi. Chiunque riceva questo messaggio per errore è pregato di distruggerlo e di informare immediatamente [ mailto:info@sigmaware.it | info@ ] asterweb.org

1 0

Fwd: Stack smash error when using auth_ephemeral with KEMI
by Grant Bagdasarian 30 May '19

30 May '19

Hi Daniel, See pastebin url below for the backtrace. https://pastebin.com/tTUUk4YH Hopefully I did it the right way. Please let me know if you need anything else. FYI, Kamailio is running inside of a docker container. version: kamailio 5.2.1 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 6.3.0 I can easily replicate this, but to be sure I've saved the core file locally on my machine. Regards, Grant On Thu, May 2, 2019 at 10:05 AM Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > Hello, > > can you get a core file for this case? > > Set `ulimit -c unlimited` before starting Kamailio as root and run again. > > Once you have the core file, get the backtrace with gdb. > > Cheers, > Daniel > > On 01.05.19 20:26, Grant Bagdasarian wrote: > > Hello, > > > > I was trying out the auth_ephemeral module inside the xhttp > > event_route in Python KEMI, and discovered a possible bug. > > > > https://pastebin.com/4Ghumfis > > > > The following will result in the stack smash error. Whenever I leave > > out the unixtimestamp the > > module will complain about not being able to convert to an int, which > > make sense, but doesn't lead to a stack smash error and a complete > > crash of Kamailio. > > > > KSR.auth_ephemeral.autheph_authenticate("1576734089:grant", "1234"). > > > > According to the documentation this module can only be used in > > REQUEST_ROUTE, but I found a presentation online which also used this > > module and that particular function in event_route, which is what I > > need for WebSocket authentication. > > > > Hope this helps. Please let me know if you need more information. > > > > Regards, > > > > Grant > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -- www.asipto.com > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com >

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Kamailio Load Balancing
by Gaurav Bmotra 30 May '19

30 May '19

HI i am using kamailio 5.1.2 With Ubuntu 18.04.2 LTS and using Freeswitch as Media server and kamailio as Register server i wan to implement load balance on this plz help thank you Gaurav -- *Regards:* Gaurav Kumar

3 7

usrloc extra attributes and db_mode=0
by Patrick Wakano 29 May '19

29 May '19

Hello list, Hope you all doing well! I am trying to use the extra attributes (xavp_contact) of the usrloc module to save some additional info about the user. I am setting the value before the save() and doing a call to registered() (not lookup()) before trying to access these extra attributes. This works fine with db_mode=3 but does not work in case of db_mode=0. Does anyone knows if it should also work with mode 0? I was expecting it to work but a call to kamcmd ul.dump show nothing related..... I don't want to use the DB for the location purposes because I prefer to have multiple servers using the dmq_usrloc (that works like a charm!) and it conflicts with DB persistence of the location table (when the DMQ replication happens, all servers (sharing the same DB) try to save the same user info in the location table) Thank you, Kind regards, Patrick Wakano

2 6

Statistics help
by Duarte Rocha 29 May '19

29 May '19

Greetings, I'm doing some monitoring of my Kamailio in order to quickly identify system failures and overloads. In order to do that i have scripts that read the output of "kamcmd sl.stats" and identify the rising of "500_replies", "5xx_replies" and "6xx_replies". However, i have rate limiting on my Kamailio, which generates "503" replies when limit is reached, which generates an unnecessary alarm. Is there a way to get more specific stats about the repleis generated by Kamailio? If that isn't possible, will i be safe only monitoring "500" and "6xx" replies? In my experience, when the server has reached a processing or memory limit, the reply is always 500, but i'm not really sure about this. Thank you for your help. Best Regards, Duarte Rocha

2 1

Push proxy and load balancer for asterisk?
by Gerrit Jacobsen 29 May '19

29 May '19

Good morning, I want to realise a push notification proxy to signal incoming calls from asterisk to softphones. Is Kamailio the preferred solution for this? I looked also at Flexisip and Asterisk scripts and was told by a seasoned professional that Kamailio is the preferable solution as its scalable. True? I would be glad if someone could share his experiences. Also can one Kamailio instance be used as an asterisk load balancer and also push notification proxy ? I have been a heavy openser user a long time ago. I used first openser as my core routing engine but then switched to yate. Please excuse my rather broad questions. From my experience with Openser the devil is in the detail. So any hints on where this setup may cause problems would be great. Thanks a lot for your input. Gerry

3 2

Kamailio Lord Balance
by Gaurav Bmotra 29 May '19

29 May '19

hi i m using kamailio 5.1 with Ubuntu 18.04.2 LTS and freeswitch as Media server -- *Regards:* Gaurav Kumar

4 4

RFC: split out old sr api from app_lua
by Daniel-Constantin Mierla 29 May '19

29 May '19

Hello, I am considering to split the app_lua module in order to move the older style API available under Lua 'sr' module in a dedicated module (like app_lua_sr). Later this module can be obsoleted and removed, once we know that everything there is covered by KEMI KSR Lua module. In this way, we make it more clear that KSR is the preferred API to use as well have the app_lua smaller, for those using only KSR and not needing 'sr' API. Therefore I am starting the discussion to see if anyone has something against or other suggestions. I cc-ed sr-dev to make developers aware, but I think the best place for discussion is sr-users, being related to how Kamailio is used, therefore reply only to sr-users. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

RTPEngine / Transcoding question
by Carsten Bock 29 May '19

29 May '19

Hi, I want to implement selective transcoding, e.g. avoiding Transcoding between a HD codec (G722) and a non HD Codec (G711). In case the caller does offer HD codecs, this is fine, e.g. if(sdp_with_codecs_by_name("G722,OPUS")) { rtpengine_offer("codec-transcode=OPUS codec-transcode=G722 codec-transcode=PCMA"); } else { rtpengine_manage("codec-transcode=PCMA"); } Now I have the issue, if the callee only sends me G711, I don't want to offer G722 or Opus to the caller. However, rtpengine_answer() does not seem to accept "codec-strip=G722 codec-strip=OPUS", e.g.: onreply_route() { if(!sdp_with_codecs_by_name("G722,OPUS")) { rtpengine_ manage("codec-strip=OPUS codec-strip=G722"); } } As a result, the SDP always contains the transcoding options, e.g. G722 and OPUS. I always end up in transcoding G722 to G711, which is meaningless in that case. Any ideas on how to solve that? (above examples very simplified) Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

2 5

RTPEngine: rtpengine_manage(), Subsequent requests, Transcoding
by Carsten Bock 29 May '19

29 May '19

Hi, quick question: If understood it correctly in the past (and it worked for me quite well this way), I can simply call rtpengine_manage() for subsequent requests without adding any additional information (e.g. used interfaces, RTP/SRTP translation or anything like that), as this information is stored in the internal hash table of the RTPEngine module during call setup and initial offer/answer. However, this does not seem to apply to the transcoding options. When I just call "rtpengine_manage()" for a Re-INVITE of a transcoded call (in my case AMR-WB to G722), the transcoding options are no longer applied. The forwarded offer does not get the transcoded codecs added (e.g. in my case G722 is not added to the list of AMR/AMR-WB from the phone). Did I misunderstand something or is it a bug? My RTPEngine is still on 7.2, but I will double check with more recent versions today or tomorrow. However, I also found no commit messages relating to this. Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

3 2

Call Routing at IMS
by Iman Mohammadi 29 May '19

29 May '19

Hi Everyone When P-CSCF receives INVITE from client it sends INVITE to S-CSCF then S-CSCF without any look up(checking the user is registered or not)sends INVITE to I-CSCF then I-CSCF sends LIR and receives LIA and sends INVITE to same S-CSCF (in case of one S-CSCF exists).My Question is here why without any look up S-CSCF sends INVITE to I-CSCF??this procedure affects system performance and increases call set up time

2 1

WebRTC bad message (offset: 22)
by Ilie Soltanici 28 May '19

28 May '19

Hello, Can someone help me to discard those messages from kamailio log files? ERROR: <core> [core/parser/parse_fline.c:262]: parse_first_line(): parse_first_line: bad message (offset: 22) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia: SIP/2.0/TLS 212.2.161.34:60477#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: VeiXbJClWneIrIyYgM5v3XKYHLA=#015#012Server: WebRTC GW#015#012Content-Length: 0#015#012#015#012> As I understand this is just an HTTP response to the first request, and shouldn't go through config file execution, but I don't understand why it is getting through it. Thank you

1 0

kamailio webrtc 183 early media.
by Zhan Bazarov 28 May '19

28 May '19

Hello! I have setup kamailio+rtpengine+asterisk.. And I have issue with early media for WEBrtc endpoints... As I understood right, for this case I need to use rel100(PRACK) for sending early media... But I didn't find any example for solve this issue... Can you help me with it? It would be greatly appreciated!

1 0

mem leak on 5.1.2?
by Andrew Chen 27 May '19

27 May '19

Hey guys, Need some help with possible run away memory issue on my 5.1.2 Kamailio. I noticed I'm getting this memory error from kam logs: May 24 13:47:52 ashintgtpsg51 /usr/sbin/kamailio[1857]: WARNING: http_client [functions.c:298]: curL_query_url(): Curl library out of memory (url:https://<some url>) my -m on kamailio startup is currently set to 64 but there is only 1 call going through. In our check_mk, I noticed the memory usage is ramping up: [image: image.png] What's the cause of this? -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

4 8

Getting the following error when sending INVITE request for calling
by Mohit Chawla 27 May '19

27 May '19

Hi , I am getting the following error in trying to send INVITE message from SIP client. {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} NOTICE: acc [acc.c:281]: acc_log_request(): ACC: transaction answered: timestamp=1558902843;method=INVITE;from_tag=69588468;to_tag=ebd29a60;call_id=4_Do2VIMWLR6Wf5ubiBxgg..;code=200;reason=OK;src_user=1016;src_domain=172.25.22.23;src_ip=172.25.22.40;dst_ouser=1015;dst_user=1015;dst_domain=172.25.22.40 May 27 02:04:03 epc-srv-03 /usr/sbin/kamailio[17238]: {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Unknown column 'src_user' in 'field list' (1054) May 27 02:04:03 epc-srv-03 /usr/sbin/kamailio[17238]: {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} ERROR: <core> [db_query.c:243]: db_do_insert_cmd(): error while submitting query May 27 02:04:03 epc-srv-03 /usr/sbin/kamailio[17238]: {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} ERROR: acc [acc.c:477]: acc_db_request(): failed to insert into database May 27 02:04:10 epc-srv-03 /usr/sbin/kamailio[17229]: {2 3 BYE 4_Do2VIMWLR6Wf5ubiBxgg..} NOTICE: acc [acc.c:281]: acc_log_request(): ACC: transaction answered: *Can anyone help me understand the reason for this error.* *I am attaching the used kamailio.cfg file in this message.* *Thanks and Regards,* *Mohit* -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 6

remove_hf() leaves junk in the headers?
by Oded Arbel 27 May '19

27 May '19

I'm trying to rewrite the `From` header in an invite going out to an external system (*), by calling remove_hf("From"); followed by append_hf("From: <some value>\r\n"); but the result is an invalid SIP message, because apparently after `remove_hf()` finishes, it leaves the content of $fU in place of the `From` header - just the raw text, without a header name or even a "\r\n" - and that eats the next header line and breaks the INVITE. Here is how the INVITE originally looks (without either `remove_hf()` or `append_hf()`): ---8<--- INVITE sip:972522353283@10.20.30.40:5060 SIP/2.0 Record-Route: <sip:10.90.80.70;lr;did=649.f0b> Via: SIP/2.0/UDP 10.90.80.70:5060;branch=z9hG4bK6188.ff5c4fd855bd2970fdf32d4807b114c2.0 Via: SIP/2.0/UDP 10.90.80.71:5066;received=10.90.80.71;branch=z9hG4bK581897a4;rport=5066 Max-Forwards: 69 From: <sip:972515340014@example.com:5066>;tag=as0bcee217 To: <sip:972522353283@10.20.30.40:5060> Contact: <sip:972515340014@10.90.80.71:5066> Call-ID: 722299dc15aad53d28172d433e8665f4@10.90.80.71:5066 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.7.2 Date: Sun, 26 May 2019 14:42:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces X-CID: ad199638-4df3-4aef-81f1-1b50581032a7 Content-Type: application/sdp Content-Length: 280 v=0 o=root 817636675 817636675 IN IP4 10.50.60.70 s=Asterisk PBX 14.7.2 c=IN IP4 10.50.60.70 t=0 0 m=audio 39064 RTP/AVP 8 0 18 9 a=maxptime:60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:39065 a=ptime:20 ---8<--- and here is another call with the same setup, with just `remove_hf("From");` being invoked (without calling `append_hf()` - this is just for reproduction): ---8<--- INVITE sip:972522353283@10.20.30.40:5060 SIP/2.0 Record-Route: <sip:10.90.80.70;lr;did=f45.91b1> Via: SIP/2.0/UDP 10.90.80.70:5060;branch=z9hG4bK024d.2690ebf7a809edca2208e6b2561ccf9e.0 Via: SIP/2.0/UDP 10.90.80.71:5066;received=10.90.80.71;branch=z9hG4bK79ae3073;rport=5066 Max-Forwards: 69 972515340014To: <sip:972522353283@10.20.30.40:5060> Contact: <sip:972515340014@10.90.80.71:5066> Call-ID: 7b20ba5a549f6f886c22985146765736@10.90.80.71:5066 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.7.2 Date: Sun, 26 May 2019 14:44:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces X-CID: ffbb4ac1-3a0f-4dd6-9ce9-7ad24361081a Content-Type: application/sdp Content-Length: 280 v=0 o=root 509740923 509740923 IN IP4 10.50.60.70 s=Asterisk PBX 14.7.2 c=IN IP4 10.50.60.70 t=0 0 m=audio 39102 RTP/AVP 8 0 18 9 a=maxptime:60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:39103 a=ptime:20 ---8<--- As you can see, the `From` header is indeed gone, but it left the $fU just before the `To` header. WTF? I'm running Kamailio 5.2.2 built locally. Notes: *) the problem I'm trying to solve is that I can't get my Asterisk installation to generate the `From` header I want, the reason is out of scope of this question, so I'm sending custom SIP headers over to Kamaiio and I want to rewrite the header in Kamailio to the "correct" value. -- Oded Arbel

6 8

PRE-PAY update timer
by Steve Bucklin 25 May '19

25 May '19

Hello all (again), I am looking at a 'good' method to start a call with "chunks" of time allocated. I wish the initial call to check a credit, and if credit is good, allow a small amount of time (say 30 seconds). I have enabled DIALOG and have looked at CNXCC, but am confused! When the initial 30 seconds runs out, I want to authorise another 30 seconds, etc, etc, etc. I have started to play and set a dialog timer that expires to a route, and then can 'return' allowing the call to continue - BUT was not able to reset the dialog timer to trip again after another 30 seconds. I tried to 'set_max_time' on cnxcc, but this did not seem to trip the event route upon timeout? Does anyone have a pointer to any examples? I also am unsure of how to update the timer and keep a call in play! Steve

5 4

No Private memory left using usrloc with db_redis in mode 3
by José Seabra 24 May '19

24 May '19

Hello there, i have stored ~200 000 registrations contacts in redis database and when kamailio starts looking for expired contacts in redis then it starts giving these errors: WARNING: PR_LOG: 123: db_redis [redis_dbase.c:1099]: db_redis_perform_query(): performing full table scan on table 'location' while performing query WARNING: PR_LOG: 123: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 0 is 'expires' *WARNING: PR_LOG: 123: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 1 is 'expires'* *ERROR: PR_LOG: 123: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7ff80922b010, 3055920); Free fragment not found!* *ERROR: PR_LOG: 123: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7ff80922b010, 3055920) called from core: db_res.c: db_allocate_rows(183), module: core; Free fragment not found!* *ERROR: PR_LOG: 123: <core> [db_res.c:185]: db_allocate_rows(): no private memory left* *ERROR: PR_LOG: 123: db_redis [redis_dbase.c:1115]: db_redis_perform_query(): Failed to allocate memory for rows * I have configured the private memory to 128 MB but seems that isn't enough to handle with all these contacts. Is the only option here increase the private memory or there is any other kind of configuration(in redis or kamailio) that i can make to avoid this? Best Regards José Seabra -- Cumprimentos José Seabra

2 1

Re: [SR-Users] Kamailio IMS installation from pacakges
by Mohit Chawla 24 May '19

24 May '19

Hi, My name is Mohit Chawla working on LTE Network integration with Kamailio IMS server. We are facing issues in installation & Configuring of kamailio IMS on CentOS 7 due to no documentation is available on web (or at least I am not able to find one). Primarily I am interested in installing packages (not building from source code as of now) and use it for further testing. I can go with building the package from source if its more easier and quick. Can you guys help me share any document/wiki which is helpful for my cause ? I will be really grateful if someone can reply me by today. Thanks and Regards, Mohit From: Mohit Chawla [mailto:mohit.chawla@azcom.in] Sent: Wednesday, May 22, 2019 5:57 PM To: 'sr-users(a)lists.kamailio.org' Subject: Kamailio IMS installation from pacakges Hi, My name is Mohit Chawla working on LTE Network integration with Kamailio IMS server. We are facing issues in installation & Configuring of kamailio IMS on CentOS 7 due to no documentation is available on web (or at least I am not able to find one). Primarily I am interested in installing packages (not building from source code as of now) and use it for further testing. I can go with building the package from source if its more easier and quick. Can you guys help me share any document/wiki which is helpful for my cause ? I will be really grateful if someone can reply me by today. Thanks and Regards, Mohit -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 3

Use of kamailio as IMS server to integrate with LTE Core Network
by Mohit Chawla 24 May '19

24 May '19

Hi , We have successfully installed the kamailio IMS server on centOS 7 using the following link (http://www.kamailio.org/wiki/tutorials/ims/installation-howto) with appropriate changes for command in CentOS 7. We are currently trying to integrate the PCSCF module with LTE Core network , but I don't have any idea how to do this. Neither I am able to find any tutorials which I can use to understand how kamailio works. Can anyone share any reference document to help me understand the kamailio working in my use case. Thanks and Regards, Mohit -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

1 0

listen (bind) to variable ip and port
by Roiss-Weiss, Roman 23 May '19

23 May '19

Hi guys, is it somehow possible to bind to a variable ip-address and port? e.g. getting the ip and port from a database? Kind regards, Roman

3 2

Kamailio CDRs for calls interrupted without BYE
by Ildar Mukhametzhanov 23 May '19

23 May '19

Hello! I am using dialog+acc modules to write CDRs in kamailio. Everything works great except one test. During active call I switched off both UACs preventing them to send BYE. Rtpengine reported about call interruption, but kamailio thinks that this call is in-progress for a long time. What is the right way to make kamailio know about calls interrupted without BYE. Thanks. -- Sincerely, *Ildar Mukhametzhanov* Head of software development VoiceON Telecom Services GmbH im(a)voice-on.net <mailto:im@voice-on.net> t. +49 30 398 20 500-77

3 8

warning logs - DB_Redis and usrloc in mode 3
by José Seabra 22 May '19

22 May '19

Hello there, I'm testing the db_redis with usrloc module in mode 3 and i have been noticing that every time that kamailio checks if there is contacts expired then it logs the following warning messages: *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1099]: db_redis_perform_query(): performing full table scan on table 'location' while performing query* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 0 is 'expires'* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 1 is 'expires'* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1274]: db_redis_perform_delete(): performing full table scan on table 'location' while performing delete* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1277]: db_redis_perform_delete(): scan key 0 is 'expires'* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1277]: db_redis_perform_delete(): scan key 1 is 'expires'* I'm not sure what is the rational behind of this, but having these logs being written when we have log level equal 2 in production is something unnecessary, so i'm asking if it shouldn't be better we have it as debug instead of warning? or do you forecast any performance issues using usrloc module in mode 3 with db_redis? Thank you for your support Cheers José -- Cumprimentos José Seabra

2 1

Re: [SR-Users] Modern SIP Trunking, Route on To: or INVITE Headers?
by JR Richardson 22 May '19

22 May '19

Benoit, Alex, I appreciate the feedback, confirming my assumptions. Good news is the Vendor finally responded positively and agreed to add in configuration option to route on INVITE R-URI in next SDK release, expected later this year. In the meantime I added 'uac_replace_to' function to my proxy to re-write To: header with R-URI, so far so good. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope > > The To header is a cosmetic, purely logical commentary on the intended ultimate destination of the call, and no routing should ever be done on it. The RURI of the INVITE is the appropriate vehicle for that. > > When dealing with proxies, however, one will encounter from time to time the grim reality that there are user agents which expect the To URI and the INVITE RURI to be in alignment. For that scenario, you can use ‘uac’ module’s stateful rewriting functionally (uac_replace_to()) to modify To with Kamailio in a way that displays fidelity and esteem for protocol formalities. > > — > Sent from mobile, with due apologies for brevity and errors. > > > On May 21, 2019, at 9:32 AM, JR Richardson <jmr.richardson(a)gmail.com> wrote: > > > > Hey Folks, > > > > I could use some feedback to see if my mind is right on this topic.I'm > > in a discussion with a software Vendor (respectfully unnamed) with > > terminal software routing on To: header info. It's my contention any > > modern SIP software should use INVITE header to retrieve DID info and > > route from there. > > > > My reasoning is due to the To: header should contain the original > > intended DID, Extension, Name, AoR or whatever else the original SIP > > transaction wanted to contact, but being in a multi-carrier, > > multi-hop, call forwarding environment, the INVITE header contains the > > hop-to-hop true intention of where the call should route at any given > > transaction. So in the case of terminal software the To: header could > > likely be irrelevant where as the INVITE header is accurate. > > > > My most recent response from the Vendor suggest I put a SIP Proxy in > > front of the terminal software and transform the To: header to > > whatever I need for this application. Although I can do this, I'm > > concerned with breaking RFC for CANCEL and BYE transaction sent back > > from the terminal software that may not be recognized by upstream > > origination carriers. > > > > Plus modifying To: header is widely frowned upon by most folks. > > > > Thanks. > > > > JR

1 0

Kamailio v5.2.3 Released
by Daniel-Constantin Mierla 22 May '19

22 May '19

Hello, Kamailio SIP Server v5.2.3 stable release is out. This is a maintenance release of the latest stable branch, 5.2, that includes fixes since the release of v5.2.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.3. For more details about version 5.2.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/05/kamailio-v5-2-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Kamailio IMS installation from pacakges
by Mohit Chawla 22 May '19

22 May '19

Hi, My name is Mohit Chawla working on LTE Network integration with Kamailio IMS server. We are facing issues in installation & Configuring of kamailio IMS on CentOS 7 due to no documentation is available on web (or at least I am not able to find one). Primarily I am interested in installing packages (not building from source code as of now) and use it for further testing. I can go with building the package from source if its more easier and quick. Can you guys help me share any document/wiki which is helpful for my cause ? I will be really grateful if someone can reply me by today. Thanks and Regards, Mohit -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

1 0

Releasing Kamailio v5.2.3
by Daniel-Constantin Mierla 22 May '19

22 May '19

Hello, I am considering to release Kamailo v5.2.3 out of latest git branch 5.2 next week, likely on Wednesday, May 22. As usual, if you are aware of issues not reported to the github bug tracker, fill a report there as soon as possible to give a chance to be analyzed and fixed. Also, if there was a fix for an issue that you were reporting, check to see if it was backported to be sure it is not forgotten. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

remove_hf inside of onreply_route
by Володимир Іванець 21 May '19

21 May '19

Hello! I was wondering if someone is using remove_hf in onreply_route and if yes on which Kamailio version? I'm using Kamailio v5.2.1 and as soon as I add remove_hf function to onreply_route Kamailio will fail to start with following error messages: ERROR: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7fc1dc044010, 27120); Free fragment not found! ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fc1dc044010, 27120) called from core: core/io_wait.c: init_io_wait(524), module: core; Free fragment not found! Thank you for any suggestions!

2 2

Modern SIP Trunking, Route on To: or INVITE Headers?
by JR Richardson 21 May '19

21 May '19

Hey Folks, I could use some feedback to see if my mind is right on this topic.I'm in a discussion with a software Vendor (respectfully unnamed) with terminal software routing on To: header info. It's my contention any modern SIP software should use INVITE header to retrieve DID info and route from there. My reasoning is due to the To: header should contain the original intended DID, Extension, Name, AoR or whatever else the original SIP transaction wanted to contact, but being in a multi-carrier, multi-hop, call forwarding environment, the INVITE header contains the hop-to-hop true intention of where the call should route at any given transaction. So in the case of terminal software the To: header could likely be irrelevant where as the INVITE header is accurate. My most recent response from the Vendor suggest I put a SIP Proxy in front of the terminal software and transform the To: header to whatever I need for this application. Although I can do this, I'm concerned with breaking RFC for CANCEL and BYE transaction sent back from the terminal software that may not be recognized by upstream origination carriers. Plus modifying To: header is widely frowned upon by most folks. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

4 3

$snd(proto) mapping?
by Andrew Chen 21 May '19

21 May '19

Hey guys, Is there a numeric mapping for $snd(proto)? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 1

How to get started (configuring an interconnection)
by Benoit Panizzon 21 May '19

21 May '19

Dear Kamailio Users I might need a bit help with starting. We are a TSP and at the moment we use a commercial carrier grade voice switch, which is very VERY inflexible and where bugs we find either don't get fixed, or only with a large delay. As everywhere else we embrace opensource and enjoy the possibility to quickly fix bugs ourselves (we have a software development department) we are evaluating Kamailio. Basically, if we have a centralized SIP routing engine, which can record CDRS for Billing and do some Number Translations (lookup on external service or database) for (location based emergency numbers, call forwarding services, ported numbers, call spam blacklisting) we could do way more than our actual $$$ carrier switch is capable of. Kamailio sounds like perfect for this task. First step: Get a machine up and running with Kamailio and Siremis, done, that was easy. Next: Connect his SIP wise via what I would call a 'SIP carrier interconnecton', ip based authenticated trunk between our carrier switch and Kamailio so we can route some number ranges from our carrier switch to Kamailio and then continue to test with subscribers, subscriber trunks and all else we need on Kamailio. I'm quite fluent in Asterisk (our Voicemail and Announcement Services are based on Asterisk) and SIP and have come across many different PBX, but after reading parts of the documentation, and trying myself to understand how Kamailio talks to other SIP endpoints I am a bit at a loss. I would have expected that I would configure our carrier switch as 'remote SIP gateway' or whatever it would be called. But I start fearing, Kamilio works completely differently. So I might need a bit of pointing the right direction to get started. I did try with the 'CarrierRoute Management' or 'Dispatcher' functions, but I only find very little documentation on this. So how do I get started? How do I, for starters, tell Kamailio what the Hostname or IP of my Carrier switch is and how do I set a 'default' call route to this? Kind regards -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

Storing an integer value into AVP from REDIS string
by Stephen Bucklin 21 May '19

21 May '19

Hi, Any quick thoughts. I am using a REDIS "GET" to retrieve a JSON obj as a string from REDIS and then the JANSSON get value to extract a key value from the string and pop it into an AVP ($avp(val)) and wish to ensure the value is an integer in the AVP. New to this, be kind :-) Steve

4 6

How to define IP Trunk in kamailio
by Mary Honey 21 May '19

21 May '19

Hello everyone, I install and setup an IMS Network by kamailio-5.2.2 . I have p-cscf, i-cscf and s-cscf in three different systems . my Hss is Fokus. Registration and call between subscribers of network work successfully . now I want to define IP Trunk in my network which can pass external calls through my IMS network. I don't know how to define IP Trunk and changes in which element(pcscf, icscf or scscf) should be done.

2 3

Rewrite contact header
by R, Rajkumar (Raj) 21 May '19

21 May '19

Hello, We are using alias routing from NATHelper module and I've a case where the alias part 'alias=ip~port~transport' needs to be removed from the contact header in 2xx message. There's contact header reference "$ct" pseudo variable but it is read only. Is there any approach available to achieve this as I haven't found anything in the forum archive search so far. thanks, raj

2 2

Unable to CDRs on "Busy Here 486" from remote side and save start and end time on successful call in table
by Shahid Mehmood 21 May '19

21 May '19

Hello, i want to save the status each call into the database. but failed to 1. record CDRs for "Busy Here 486/ service unavailable 503 " from remote side in database. 2. on successful call unable to record total time of call in database. Version of Kamailio Is *version: kamailio 4.4.7 (x86_64/linux) 35d8abflags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: 35d8ab* *Also Attached Config File* *Thanks*

1 0

kamailio Early IMS security
by eyas barhouk 20 May '19

20 May '19

hello community Does kamailio support Early IMS security authentication schema to be able to integrate with Huawei HSS ?? and if it's does how to configure it? B.R Eyas

1 0

Plans for next major release - v5.3.0
by Daniel-Constantin Mierla 20 May '19

20 May '19

Hello, a while ago, during the last IRC devel meeting, it was proposed to either do the next major release (5.3.0) before the usual summer holidays or leave it for autumn. Given that time is approaching, we have to decide. If it is going to happen before the summer holidays, probably we should freeze the development in 2-3 weeks, then do 3-4 weeks of testing and release during the first part of July. It would be a shorter development cycle, so I do not expect much needs for testing. Only a few new modules and plenty of work to consolidate and enhance the existing code. Any preference for one of the two options? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

3 2

Kamailio usrloc mode 3 using mongodb
by José Seabra 20 May '19

20 May '19

Hello there, I'm testing usrloc module with mongodb and i have a couple of questions to do. What i would like to know/have confirmation is if the fields that are specified in the following link https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-location-… can be used to create indexes in mongodb? I'm noticing that when a register is updated/deleted or expires kamailio prints the following INFO logs: *db_mongodb [mongodb_dbase.c:526]: db_mongodb_get_columns(): unhandled data type column (methods) type id (10), use DB1_STRING as default* *db_mongodb [mongodb_dbase.c:526]: db_mongodb_get_columns(): unhandled data type column (instance) type id (10), use DB1_STRING as default* These fields (*methods and **instance*) are stored in mongodb document with a null string, e.g: *{* * "_id" : ObjectId("5cdc775a6472740055289a11"),* * "username" : "x1000",* * "contact" : "sip:x1000@192.168.1.1:5674;transport=UDP",* * "expires" : ISODate("2019-05-15T20:33:26Z"),* * "q" : -1,* * "callid" : "sipp-ci2-edge-sip-proxy1-20190515203220-1-21(a)172.31.2.61 <sipp-ci2-edge-sip-proxy1-20190515203220-1-21(a)172.31.2.61>",* * "cseq" : 2,* * "flags" : 0,* * "cflags" : 0,* * "user_agent" : "sipp",* * "received" : "sip:191.30.10.23:8888 <http://191.30.10.23:8888>",* * "path" : "<sip:172.31.2.61:5060;received=sip:191.30.10.23:8888;lr>",* * "socket" : "udp:172.31.2.11:5060 <http://172.31.2.11:5060>",* * "methods" : null,* * "last_modified" : ISODate("2019-05-15T20:32:26Z"),* * "ruid" : "uloc-15-5cdc7756-55-1",* * "instance" : null,* * "reg_id" : 0,* * "server_id" : 21,* * "connection_id" : -1,* * "keepalive" : 0,* * "partition" : 0* *}* Should i be concerned about that log msg? how can i avoid it? Thanks Regards -- Cumprimentos José Seabra

2 5

BYE relays Digest and others in Proxy-Authorization header
by Sergiu Pojoga 19 May '19

19 May '19

Hi ppl, Noticed BYE requests relayed from caller to callee have the *Proxy-Authorization* header including *Digest*, *nonce* and *response*. Is that by SIP definition or am I doing something wrong? My concern is the callee gets to see caller's authentication credentials. Thanks.

4 6

Centos7 Packages
by Matt Lamer 18 May '19

18 May '19

I am trying to install the kam server on centos7. When running the install I get the following: Install 5 Packages (+2 Dependent packages) Total size: 14 M Total download size: 14 M Installed size: 63 M Downloading packages: Delta RPMs disabled because /usr/bin/applydeltarpm not installed. kamailio-debuginfo-5.2.2-7.el7 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not Satisfiable Anyone else having this issue?

2 1

"If you're struggling with your Tele callers to improve sales? you're not alone.
by Prakash 17 May '19

17 May '19

http://www.doocti.com +919790776655 Doocti Next Generation CTI for Smarter Sales Teams http://www.doocti.com Dear Customer, Greeting from Tevatel. Hope this mail finds you well. We are happy to announce the launch of our new product Doocti - Call center connector for crm What is Doocti? Doocti is a Computer telephony integration platform that can integrate your communication devices with your Helpdesk, CRM, and Other Key Tools. Increase agent productivity, improve customer experience and enable personalized service through deep customer insights at the point of interaction. Features of Doocti Single click Dial On-Screen Customer Profile Build on the Customer Profile Efficient Call Disposition Auto Dial Sticky Agent Feedback from the clients Reduce call time by 10-20 seconds per call Instantly retrieves and screen-pops customer related information Provides a better and more immediate response to the customers Offers a more personalized experience to the caller Dials out automatically and faster with a single mouse click Handles higher call volumes Increases first call resolution Intergrated with Currently we have associated with more than 20 brands. Curious to see what's under the hood? Enjoy a detailed look at the Doocti experience in a demo guided by a member of our team. Get a meeting on the calendar today. mailto:prakash.n@tevatel.com?subject=Need More Information about Doocti Request for a Demo [ mailto:prakash.n@tevatel.com?subject=Need More Information about Doocti ] Communications Simplified Contact Us: +919790776655 | prakash.n(a)tevatel.com No:76, Swamibakthan Street, Nammalwarpet, Chennai - 600012. www.tevatel.com | www.doocti.com http://viewstripo.email/?utm_source=templates&utm_medium=email&utm_campaign… ---------------------------------------------------------------------- This email was sent by prakash.n(a)tevatel.com to sr-users(a)lists.sip-router.org Not interested?Unsubscribe - https://tihk.maillist-manage.com/ua/optout?od=27218d28c96aa859ed0ac1287be0e… Update profile - https://tihk.maillist-manage.com/ua/upc?upd=164d3589027540e5&r=164d35890275… Asterfone technologies pvt ltd | Swamibakthan St, Thiruvalluvar Nagar, Nammalwarpet, Ayanavaram, Chennai, Tamil Nadu 600010 Our Privacy Policy [ ] and Terms of Use. [ ]

2 1

rtpengine sessions on timeout not freeing up RTP ports
by Istvan Mogyorosi 17 May '19

17 May '19

Hello Another rtpengine related question after having successfully implemented a dispatcher configuration. Call flow is ok, everything works well until I tried to increase call volume. Kamailio log showing "proxy replied with error: Ran out of ports" In the beginning both channels are up, then I can see a teardown on the incoming channel loosing the ability to collect DTMF. Using the CLI tool I have statistics showing that none (!) of the calls are "regular terminated", they are on timeout. After experimenting with handling BYE, I ended up could getting a "regular terminated" call by using route[RELAY] .... if (is_method("BYE")) { rtpengine_delete(); } and then I can see some good things! Received command 'delete' from 127.0.0.1:58538 Scheduling deletion of call branch 'as65b95845' (via-branch '') in 5 seconds Replying to 'delete' from 127.0.0.1:58538 (elapsed time 0.000216 sec) Please correct me if I'm doing wrong this way! cheers Istvan

1 0

RFC: Kamailio command line parameters
by Daniel-Constantin Mierla 17 May '19

17 May '19

Hello, a while ago I added support for long format parameter names for Kamailio command line. That's opening the door for more options to be set via command line arguments and I discovered that especially in a containerized environment, it is a trend to start applications using more command line parameters, than relying on config files. The current command line parameters are listed by: kamailio -h Use the master branch for it, because has many more than 5.2.x. Anyhow, the purpose of this message is to see if anyone else considers useful to add other command line parameters, and if yes, which ones. A typical candidate will be a global parameter that you have to change when relocating to/spinning up another Kamailio environment and you need to change the kamailio.cfg. Those that generate the interest of many people, can be easily added now. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Testing the KEMI interpreters for functions with 4 and 5 parameters
by Daniel-Constantin Mierla 17 May '19

17 May '19

Hello, because there were many variants not handled for KEMI functions with 4 and 5 parameters, I made a script to generate the code in order to be sure none is missing. The script was pushed in Kamailio repo at: * https://github.com/kamailio/kamailio/tree/master/misc/tools/kemi You don't have to use it, I listed it for reference, just in case the code needs adjustments in the future. If anyone is using a KEMI interpreter and has functions exported by Kamailio with 4 or 5 parameters, it would be appreciated if you can run some tests with master branch and report the results. All kemi interpreter modules were updated, so it doesn't matter which one you test. If the results are good, then the patches will be backported to branch 5.2, so there will be no case of unhanded functions with these numbers of parameters. The cases with up to 3 parameters were not affected. For 6 params, still only string values are handled. The other variants can be added, but I think they are not needed for the moment, so I tried to keep it small for the moment. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Kamailio World thanks
by Daniel Pocock 16 May '19

16 May '19

Hi Daniel and Kamailio team, Well done on another successful Kamailio World in 2019. As usual, it was a great opportunity to meet people and learn new things. Regards, Daniel

2 1

siputils add_uri_param not available in KEMI python3
by Patrick Murphy 15 May '19

15 May '19

Hello, I am using the following version of Kamailio and using app_python3 with KEMI but I cannot find any exported function of add_uri_param from the siputils module. Any idea how I can perform: KSR.siputils.add_uri_param("name=value") ? version: kamailio 5.2.2 (x86_64/linux) 9260a2 Cheers, Pat.

1 1

Using a single virtual domain in a multi-domain setup
by Rhys Hanrahan 15 May '19

15 May '19

Hi All, I am trying to setup Kamailio to allow me to have multiple different domains (and IPs) treated as one. I am using the domain module for this. As an example I have: * A handset registered to Kamailio by IP address. * A SIP trunk between Asterisk and Kamilio using an FQDN that uses SRV records for redundancy on Asterisk's side. I have added all domains and IPs under the same "domain identifier". It seems the "old" domain module now uid_domain had this capability of a single virtual domain, but is this possible with the more current domain.so module? From what I've said the uid_* modules older modules from the SIP Router project, and should be replaced by more recent Kamailio modules? - https://www.kamailio.org/docs/modules/5.0.x/modules/uid_domain.html VS. - https://kamailio.org/docs/modules/5.1.x/modules/domain.html MariaDB [kamailio]> select * from domain; +----+------------------------------------+------------+---------------------+ | id | domain | did | last_modified | +----+------------------------------------+------------+---------------------+ | 1 | 12.34.56.53 | vocalpoint | 2000-01-01 00:00:01 | | 2 | 23.34.56.54 | vocalpoint | 2000-01-01 00:00:01 | | 3 | sbc.foo | vocalpoint | 2000-01-01 00:00:01 | | 4 | sbc.nsw.foo | vocalpoint | 2000-01-01 00:00:01 | | 5 | sbc.vic.foo | vocalpoint | 2000-01-01 00:00:01 | | 6 | sbc.foo2 | vocalpoint | 2000-01-01 00:00:01 | | 7 | sbc.nsw.foo2 | vocalpoint | 2000-01-01 00:00:01 | | 8 | sbc.vic.foo2 | vocalpoint | 2000-01-01 00:00:01 | +----+------------------------------------+------------+---------------------+ 8 rows in set (0.000 sec) I managed to get registrations working by hardcoding a static realm in the config, but when when usrloc does a lookup on a handset, Kamailio can't find user(a)sbc.nsw.foo and Kamailio returns 404. It looks like this is because in the locations table it is listed with a domain of 12.34.56.53 (because the handset is registered by IP). How should this work normally? Or is it even possible? Thanks, Rhys.

2 4

Sip proxy for Asterisk with RFC3966 and 100Rel
by paolo.visnoviz＠vipcomputers.it 15 May '19

15 May '19

Dear Sirs, a telco provider give us a trunk that require 100rel and TEL RFC 3966. Usually we use Asterisk. Now Asterisk 16 with PJSIP support 100Rel, but not RFC3966, and they don't actually works to implement it. Asterisk chan_sip, instead, support RFC3966, but not 100Rel and is not actively developed on by Digium or Sangoma. So I'm stuck. Now, do you think it was possible to use Kamailio to act like a proxy sip but implementing RFC3966 for incoming and outgoing calls and RFC3966 and 100Rel for outgoing calls? And it is possible to connect all to an Asterisk server? Thank you in advance Best regards. Paolo Visnoviz

3 9

Possible memory leak in siptrace module in kamailio 5.1/5.2
by davy van de moere 15 May '19

15 May '19

Dear, I'm using siptrace within a python handler in the KEMI framework. KSR.siptrace.sip_trace_dst("sip:<myhomerip>:9060") I notice an ever increasing usage of memory, until the system runs out of memory. (definition of a memory leak :) ). Looking at the memory information kamailio is giving me I notice this: May 14 05:49:48 server kamailio[36452]: NOTICE: PYT {INVITE} {1769595029-1513210001-1955500877} {1} : qm_sums: count= 79286 size= 5812048 bytes from siptrace: siptrace.c: ki_sip_trace_dst_cid(729) May 14 05:51:17 server kamailio[36452]: NOTICE: PYT {REGISTER} { 4_1419955802(a)192.168.0.153} {2582} : qm_sums: count= 79335 size= 5814840 bytes from siptrace: siptrace.c: ki_sip_trace_dst_cid(729) I notice this on kamailio 5.1.8 (from a debian repo), and on kamailio 5.2.2 (from source). I can reproduce the problem very easily, and I can apply patches in the source to try out things. Could I be wrongly using siptrace? Can I give you more debugging info? Kind regards, Davy

3 4

define a DBURL with SLQOPS fails
by David Villasmil 14 May '19

14 May '19

Hello guys, So, i'm defining: DBURL "mysql://user:password@dbhost/dbname" and later on... modparam("sqlops","sqlcon","cb=>"DBURL) and this fails miserably.. What do do on this case? Thanks all! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 12

Npdi question
by Duarte Rocha 14 May '19

14 May '19

Greetings, Regarding portability issues, is there a difference between "npdi" and "npdi=yes" ? Are there any more possible values besides "yes" and no value? Best Regards, Duarte Rocha

1 0

sdp_remove_codecs_by_id does not remove 101
by Juha Heinanen 14 May '19

14 May '19

I tried to use sdp_remove_codecs_by_id("101") to remove "codec" 101 (dtmf) from sdp, but nothing happened. It is still there: m=audio 31128 UDP/TLS/RTP/SAVPF 101 96. c=IN IP4 192.168.43.88 a=label:1. a=rtcp-rsize. a=ssrc:2880178656 cname:sip:test@test.tutpro.com. a=rtpmap:101 telephone-event/8000. a=rtpmap:96 opus/48000/2. a=fmtp:101 0-15. Is that by design or a bug? If not a bug, is there some other means to remove 101? -- Juha

2 2

sip trunk with kamailio
by arish haque 14 May '19

14 May '19

Hi, I am trying the SIP trunk from kamailio with following configurations - cfg_get.pstn.gw_ip="10.0.x.y route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } if(!($rU=~"^(\+|00)918884978235")) 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; } I am able to see kamailio sending INVITE to SIP Provider but no response back. Any other, configuration is required which I'm missing. I have attached the tcpdump snapshot of the above scenario. Thanks, Arish

2 1

Theoritical Anycast Setup
by Daniel Oakes 14 May '19

14 May '19

Hi there, Long time user, 1st time poster - apologies if someone has posted this previously. We have a product from a vendor that utilizes Kamailio as a SIP router - but we're wanting to extend it a little bit by the use of anycast (we're a small ISP). Basically the scenario is that we want to utilize Quagga via BGP to anycast an address, which addresses a glaring load balancing issue with have with people who use only A records. Problem is anycast doesn't work so well in this scenario, just by adding the Anycast as a listen address on the proxy. What we'd want to do is rewrite the SIP so that all responses come back out saying to use the 'real ip'. So I have two variables - say REAL_IP_ADDRESS and ANYCAST_IP_ADDRESS - request will come in via ANYCAST, but I want all responses from then on to be via REAL_IP_ADDRESS. We have six Kamailio boxes in various geographical locations, so the idea is they hit one based on closest, and we rewrite it so they end up with the real ip address to talk to. Anyone done something similar? I'm guessing possibly, but I can't find anything specific enough for me to use. Help appreciated! Regards, Daniel

2 1

No audio after IP address change
by Chris Scheid 11 May '19

11 May '19

Hi, we are using rtpengine with Kamailio using DTLS-SRTP and SRTP making audio calls from two Android devices. Regular audio calls work fine but we have issues with the following scenario: 1. Device 1 (72.66.83.8, on WiFI) calls Device 2 (174.204.14.29). SIP server has IP 209.182.216.71. 2. Device 2 switches networks from WiFI to 3G (new IP: 97.47.70.223). Re-invite flow completes. 3. Any audio from Device 2 cannot be heard on Device 1. But audio from the Device1 can be heard on the Device 2. In the rtpengine stats I see that there were some packets sent from the SIP server to the Device 1 after network change: [1557513316.000308] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: ------ Media #1 (audio over UDP/TLS/RTP/SAVP) using iLBC/8000 [1557513316.000318] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20052 <> 97.47.70.223:4667 , SSRC 77c38349, 55 p, 3410 b, 0 e, 32 ts [1557513316.000325] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20053 <> 97.47.70.223:4640 (RTCP), SSRC 77c38349, 12 p, 1120 b, However, in my UDP trace, I don't see any communication between the SIP server and the Device 1 after network change (IP 97.47.70.223). This seems to be the reason why we don't get audio on Device 1. I'm looking for direction on how to troubleshoot this issue. I'm not so familiar with rtpengine. Any assistance greatly appreciated. Thanks! RTP Engine Config: [rtpengine] #####table = 0 listen-ng = 127.0.0.1:5050 timeout = 60 silent-timeout = 3600 tos = 184 port-min = 20000 port-max = 60000 interface = 209.182.216.71 log-level = 7 log-stderr = false log-facility = local0 START CALL --------------------------------------------------------------------------------------------------------------------- [1557513231.973294] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'offer' from 127.0.0.1:45431 [1557513231.973357] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'offer' from 127.0.0.1:45431: { "sdp": "v=0 o=- 3766502030 3766502030 IN IP4 192.168.1.200 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 5095 UDP/TLS/RTP/SAVP 104 98 97 99 3 0 8 9 96 c=IN IP4 192.168.1.200 b=TIAS:64000 a=rtcp:5096 IN IP4 192.168.1.200 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ssrc:2009301833 cname:32ababda5d2c847b a=setup:actpass a=fingerprint:SHA-256 4F:69:B6:89:51:98:6C:A4:63:CF:F7:CD:22:D2:BE:FD:6E:3F:F4:2A:8B:39:AD:E4:C0:8D:AE:BD:A3:70:12:49 ", "ICE": "remove", "DTLS": "passive", "replace": [ "origin", "session-connection" ], "transport-protocol": "UDP/TLS/RTP/SAVP", "call-id": "05c7dcfe-06f3-4630-9 799-0557b76cc340", "received-from": [ "IP4", "72.66.83.8" ], "from-tag": "2684eff8-3602-41ed-aca2-9f7938362462", "command": "offer" } [1557513231.973444] NOTICE: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating new call [1557513231.973496] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Default sink codec is speex/16000 [1557513231.973503] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for iLBC/8000 [1557513231.973508] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for speex/16000 [1557513231.973512] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/16000 [1557513231.973515] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for speex/8000 [1557513231.973519] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/8000 [1557513231.973523] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for speex/32000 [1557513231.973526] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/32000 [1557513231.973530] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for GSM/8000 [1557513231.973534] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec GSM/8000 [1557513231.973538] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for PCMU/8000 [1557513231.973542] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMU/8000 [1557513231.973545] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for PCMA/8000 [1557513231.973549] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMA/8000 [1557513231.973545] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for PCMA/8000 [1557513231.973549] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMA/8000 [1557513231.973552] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for G722/8000 [1557513231.973556] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec G722/8000 [1557513231.973560] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for telephone-event/8000 [1557513231.974291] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 192.168.1.200:5095 [1557513231.974561] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 192.168.1.200:5096 [1557513231.974874] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: offer time = 0.001488 sec [1557513231.974882] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'offer' from 127.0.0.1:45431 [1557513231.974900] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'offer' to 127.0.0.1:45431: { "sdp": "v=0 o=- 3766502030 3766502030 IN IP4 209.182.216.71 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 20032 UDP/TLS/RTP/SAVP 104 98 97 99 3 0 8 9 96 c=IN IP4 209.182.216.71 b=TIAS:64000 a=ssrc:2009301833 cname:32ababda5d2c847b a=rtpmap:104 iLBC/8000 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:104 mode=30 a=fmtp:96 0-16 a=sendrecv a=rtcp:20033 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:G9MvyRPBAVFsrMypvr1EAunZiX/vc04sXEq4mXvl a=setup:actpass a=fingerprint:sha-1 EE:4A:ED:7D:5A:94:A2:FF:F2:09:64:07:41:FC:15:80:D3:F9:C0:B4 ", "result": "ok" } [1557513232.077386] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Processing incoming DTLS packet [1557513232.078219] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Sending DTLS packet [1557513232.189220] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Processing incoming DTLS packet [1557513232.189868] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: DTLS handshake successful [1557513232.189899] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: DTLS-SRTP successfully negotiated [1557513232.189905] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: SRTP keys, incoming: [1557513232.189914] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: --- AES_CM_128_HMAC_SHA1_80 key q67B3h3GP/dhbQeKsgBnvA== salt EcoNJ6Hqj0lRtvZ8Z+Y= [1557513232.189919] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: SRTP keys, outgoing: [1557513232.189923] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: --- AES_CM_128_HMAC_SHA1_80 key 0nTo2gy4XJ2GXlroyyH6nA== salt fbIfUIa4IPiYLvn5CkI= [1557513232.189929] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Sending DTLS packet confirmed [1557513243.648460] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SRTCP output wanted, but no crypto suite was negotiated [1557513243.648486] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 201 [1557513243.648496] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about 9fda818d: FL 205 TL 6486816 HSR 613804890 J 3100337071 LSR 1419186059 DLSR 531075031 [1557513243.648505] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513243.649142] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 201 [1557513243.649158] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about d4b1278d: FL 23 TL 2271802 HSR 278642673 J 2408385537 LSR 3 653820601 DLSR 1701759944 [1557513243.649169] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513243.777813] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'answer' from 127.0.0.1:47861 [1557513243.777860] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'answer' from 127.0.0.1:47861: { "sdp": "v=0 o=- 3766502031 3766502032 IN IP4 100.81.62.76 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 5079 UDP/TLS/RTP/SAVP 104 96 c=IN IP4 100.81.62.76 b=TIAS:64000 a=rtcp:5080 IN IP4 100.81.62.76 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ssrc:2006816651 cname:1c0965113e9b1046 a=setup:active a=fingerprint:SHA-256 AD:2D:0F:A6:11:DD:75:CC:3C:50:3A:9A:F0:29:20:8D:95:BC:3E:2A:A0:34:F6:67:14:A7:87:EB:8D:66:77:B1 ", "ICE": "remove", "replace": [ "origin", "session-connection" ], "transport-protocol": "UDP/TLS/RTP/SAVP", "call-id": "05c7dcfe-06f3-4630-9799-0557b76cc340", "received-from": [ "IP4", "174.204.14.29" ], "from-tag": "2684eff8-3602-41ed-aca2-9f7938362462", "to-tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "command": "a nswer" } [1557513243.777998] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for iLBC/8000 [1557513243.778010] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for telephone-event/8000 [1557513243.778088] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 100.81.62.76:5079 [1557513243.778117] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: DTLS: Peer certificate accepted [1557513243.778123] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 100.81.62.76:5080 [1557513243.778166] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: answer time = 0.000267 sec [1557513243.778175] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'answer' from 127.0.0.1:47861 [1557513243.778187] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'answer' to 127.0.0.1:47861: { "sdp": "v=0 o=- 3766502031 3766502032 IN IP4 209.182.216.71 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 20052 UDP/TLS/RTP/SAVP 104 96 c=IN IP4 209.182.216.71 b=TIAS:64000 a=ssrc:2006816651 cname:1c0965113e9b1046 a=rtpmap:104 iLBC/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:104 mode=30 a=fmtp:96 0-16 a=sendrecv a=rtcp:20053 a=setup:passive a=fingerprint:sha-1 EE:4A:ED:7D:5A:94:A2:FF:F2:09:64:07:41:FC:15:80:D3:F9:C0:B4 ", "result": "ok" } [1557513243.864786] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Processing incoming DTLS packet [1557513243.865568] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Sending DTLS packet [1557513243.912858] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Processing incoming DTLS packet [1557513243.913111] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS: Peer certificate accepted [1557513243.913627] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS handshake successful [1557513243.913668] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS-SRTP successfully negotiated [1557513243.913675] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: SRTP keys, incoming: [1557513243.913682] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: --- AES_CM_128_HMAC_SHA1_80 key +C2ck1NPvsfHYoKflimqaA== salt xM4aFoL60/ePpmQJsT8= [1557513243.913686] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: SRTP keys, outgoing: [1557513243.913689] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: --- AES_CM_128_HMAC_SHA1_80 key VvTNUyescFh+vNj16/fePw== salt m3EuTJjAesJHaSXfqKU= [1557513243.913694] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Sending DTLS packet [1557513244.475362] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513244.475405] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 3873048209 PC 2516599107 OC 673505029 NTP TS 1161195675/4 008032614=1161195675.933193 [1557513244.475412] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about bb94938e: FL 243 TL 11665253 HSR 1302964504 J 2830082235 LS R 1964993324 DLSR 1625750139 [1557513244.475422] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513244.492749] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SRTCP output wanted, but no crypto suite was negotiated [1557513244.492775] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513244.492791] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 2981125880 PC 4052113656 OC 2653921327 NTP TS 4076801750/ 1160081551=4076801750.270103 [1557513244.492797] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 3813d27e: FL 75 TL 11393233 HSR 1719559869 J 944495881 LSR 790240102 DLSR 447292567 [1557513244.492804] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513247.217290] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Confirmed peer address as 174.204.14.29:10523 [1557513248.958592] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513248.958638] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 3148803183 PC 137367242 OC 1506652935 NTP TS 3606800652/1 621372750=3606800652.377505 [1557513248.958647] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about 468fa6ae: FL 57 TL 15596558 HSR 3764947872 J 3769875796 LSR 4259175195 DLSR 3903464346 [1557513248.958657] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513248.958667] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Confirmed peer address as 174.204.14.29:10496 i[1557513249.259063] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513249.259108] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 3424021083 PC 3823893658 OC 584056375 NTP TS 3244065141/3 391763074=3244065141.789706 [1557513249.259117] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about e7313af6: FL 98 TL 10263186 HSR 2762600672 J 838876573 LSR 3367219470 DLSR 50549698 [1557513249.259131] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513249.259142] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Confirmed peer address as 72.66.83.8:5096 t[1557513249.562239] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Confirmed peer address as 72.66.83.8:5095 SWITCH NETWORKS [1557513253.843937] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513253.844019] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 1364757765 PC 1063357287 OC 2194907961 NTP TS 2472417766/ 1884000851=2472417766.438653 [1557513253.844029] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about 77b8933: FL 163 TL 2305413 HSR 2377964402 J 118139039 LSR 2 460485565 DLSR 3416921415 [1557513253.844040] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513254.590466] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513254.590508] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 2527484095 PC 1878948628 OC 3882176963 NTP TS 213338522/6 38146641=213338522.148580 [1557513254.590517] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about b7800717: FL 93 TL 16503124 HSR 338254320 J 1043490285 LSR 801212789 DLSR 2849120558 [1557513254.590530] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513258.679347] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SRTCP output wanted, but no crypto suite was negotiated [1557513258.679385] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513258.679411] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 1747525463 PC 239751946 OC 2859161225 NTP TS 3835334254/3 577221347=3835334254.832887 [1557513258.679418] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about e03bd142: FL 40 TL 11515285 HSR 3870936491 J 4150119236 LSR 2903910302 DLSR 700479064 [1557513258.679427] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513259.940740] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SRTCP output wanted, but no crypto suite was negotiated [1557513259.940770] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513259.940793] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 2745657868 PC 580822102 OC 4072320279 NTP TS 252333929/20 93935903=252333929.487532 [1557513259.940801] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 476ec510: FL 61 TL 5399464 HSR 3682903462 J 1600791504 LSR 497094189 DLSR 826539320 [1557513259.940814] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513263.532695] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513263.532722] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 301750475 PC 2434799802 OC 863475295 NTP TS 3625503376/30 14095725=3625503376.701774 [1557513263.532729] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about 228989b7: FL 148 TL 10638176 HSR 2137788972 J 2678688015 LS R 2118438561 DLSR 1024262683 [1557513263.532738] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513264.051305] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513264.051352] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 1190725747 PC 3072491822 OC 537479863 NTP TS 817717026/42 82321058=817717026.997056 [1557513264.051359] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 920db8ff: FL 98 TL 6598240 HSR 2045142812 J 3171281598 LSR 2224002690 DLSR 2514604823 [1557513264.051369] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513264.058891] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 201 [1557513264.058920] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 5c8f2711: FL 24 TL 1700303 HSR 2993289918 J 618517078 LSR 1 835926259 DLSR 3106454771 [1557513264.058891] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 201 [1557513264.058920] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 5c8f2711: FL 24 TL 1700303 HSR 2993289918 J 618517078 LSR 1 835926259 DLSR 3106454771 [1557513264.058934] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513264.266612] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'offer' from 127.0.0.1:60736 [1557513264.266654] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'offer' from 127.0.0.1:60736: { "sdp": "v=0 o=- 3766502030 3766502031 IN IP4 192.168.1.200 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 5097 UDP/TLS/RTP/SAVP 104 98 97 99 3 0 8 9 96 c=IN IP4 10.255.81.66 b=TIAS:64000 a=rtcp:5098 IN IP4 10.255.81.66 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ssrc:2009301833 cname:32ababda5d2c847b a=setup:actpass a=fingerprint:SHA-256 4F:69:B6:89:51:98:6C:A4:63:CF:F7:CD:22:D2:BE:FD:6E:3F:F4:2A:8B:39:AD:E4:C0:8D:AE:BD:A3:70:12:49 ", "ICE": "remove", "DTLS": "passive", "replace": [ "origin", "session-connection" ], "transport-protocol": "UDP/TLS/RTP/SAVP", "call-id": "05c7dcfe-06f3-4630-9 799-0557b76cc340", "received-from": [ "IP4", "97.47.70.223" ], "from-tag": "2684eff8-3602-41ed-aca2-9f7938362462", "to-tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d 00", "command": "offer" } [1557513264.266814] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Default sink codec is speex/16000 [1557513264.266826] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/16000 [1557513264.266831] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/8000 [1557513264.266835] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/32000 [1557513264.266838] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec GSM/8000 [1557513264.266842] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMU/8000 [1557513264.266846] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMA/8000 [1557513264.266850] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec G722/8000 [1557513264.267614] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sending DTLS packet [1557513264.268089] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sending DTLS packet [1557513264.268146] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sending DTLS packet [1557513264.268220] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 10.255.81.66:5097 [1557513264.268627] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 10.255.81.66:5098 [1557513264.269072] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: offer time = 0.002384 sec [1557513264.269084] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'offer' from 127.0.0.1:60736 [1557513264.269098] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'offer' to 127.0.0.1:60736: { "sdp": "v=0 o=- 3766502030 3766502031 IN IP4 209.182.216.71 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 20064 UDP/TLS/RTP/SAVP 104 98 97 99 3 0 8 9 96 c=IN IP4 209.182.216.71 b=TIAS:64000 a=ssrc:2009301833 cname:32ababda5d2c847b a=rtpmap:104 iLBC/8000 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:104 mode=30 a=fmtp:96 0-16 a=sendrecv a=rtcp:20065 a=setup:actpass a=fingerprint:sha-1 EE:4A:ED:7D:5A:94:A2:FF:F2:09:64:07:41:FC:15:80:D3:F9:C0:B4 ", "result": "ok" } [1557513264.355109] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SRTCP output wanted, but no crypto suite was negotiated [1557513264.355140] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 201 [1557513264.355150] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 511a36de: FL 144 TL 3729734 HSR 2556650506 J 4034165594 LSR 900570716 DLSR 3118967760 [1557513264.355166] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513264.355186] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 201 [1557513264.355193] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about aea81208: FL 180 TL 10378624 HSR 1586287909 J 2246375086 LS R 3231023203 DLSR 4073439206 [1557513264.355201] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513264.355210] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Switching local interface to 209.182.216.71:20033 [1557513264.358206] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 201 [1557513264.358229] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about b59d47bc: FL 242 TL 13886413 HSR 3966401064 J 2679493971 LS R 3040876731 DLSR 2668321392 [1557513264.358239] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513264.358262] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20064]: SRTP output wanted, but no crypto suite was negotiated [1557513264.425636] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'answer' from 127.0.0.1:45431 [1557513264.425679] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'answer' from 127.0.0.1:45431: { "sdp": "v=0 o=- 3766502031 3766502033 IN IP4 100.81.62.76 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 5079 UDP/TLS/RTP/SAVP 104 96 c=IN IP4 100.81.62.76 b=TIAS:64000 a=rtcp:5080 IN IP4 100.81.62.76 a=sendrecv a=rtpmap:104 iLBC/8000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ssrc:2006816651 cname:1c0965113e9b1046 a=setup:active a=fingerprint:SHA-256 AD:2D:0F:A6:11:DD:75:CC:3C:50:3A:9A:F0:29:20:8D:95:BC:3E:2A:A0:34:F6:67:14:A7:87:EB:8D:66:77:B1 ", "ICE": "remove", "replace": [ "origin", "session-connection" ], "transport-protocol": "UDP/TLS/RTP/SAVP", "call-id": "05c7dcfe-06f3-4630-9799-0557b76cc340", "received-from": [ "IP4", "174.204.14.29" ], "from-tag": "2684eff8-3602-41ed-aca2-9f7938362462", "to-tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "command": "a nswer" } [1557513264.426806] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: answer time = 0.001099 sec [1557513264.426820] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'answer' from 127.0.0.1:45431 [1557513264.426831] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'answer' to 127.0.0.1:45431: { "sdp": "v=0 o=- 3766502031 3766502033 IN IP4 209.182.216.71 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 20052 UDP/TLS/RTP/SAVP 104 96 c=IN IP4 209.182.216.71 b=TIAS:64000 a=ssrc:2006816651 cname:1c0965113e9b1046 a=rtpmap:104 iLBC/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:104 mode=30 a=fmtp:96 0-16 a=sendrecv a=rtcp:20053 a=setup:passive a=fingerprint:sha-1 EE:4A:ED:7D:5A:94:A2:FF:F2:09:64:07:41:FC:15:80:D3:F9:C0:B4 ", "result": "ok" } [1557513264.700442] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Processing incoming DTLS packet [1557513264.701235] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Sending DTLS packet [1557513264.892393] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Processing incoming DTLS packet [1557513264.893153] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS handshake successful [1557513264.893193] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS-SRTP successfully negotiated [1557513264.893202] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: SRTP keys, incoming: [1557513264.893211] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: --- AES_CM_128_HMAC_SHA1_80 key dob4sywpHJJ8qNgFzemlzQ== salt y7/1RgzeAtPkG+GfMLI= [1557513264.893216] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: SRTP keys, outgoing: [1557513264.893222] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: --- AES_CM_128_HMAC_SHA1_80 key ontM1nC56XojOkmzb8520g== salt Hom6TrHovUBUaF1g6Iw= [1557513264.893228] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Sending DTLS packet [1557513269.223923] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Confirmed peer address as 97.47.70.223:4667 [1557513269.337491] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513269.337543] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 402044265 PC 1976845075 OC 2962195809 NTP TS 1329245270/2 977782988=1329245270.693319 [1557513269.337552] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about c1e9445b: FL 196 TL 15740765 HSR 1231260891 J 3373413243 LS R 2989889476 DLSR 1721296716 [1557513269.337562] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513269.337571] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Confirmed peer address as 97.47.70.223:4640 [1557513269.355712] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513269.355744] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 46888194 PC 1178262997 OC 1943197043 NTP TS 748304878/374 [1557513269.355749] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 4d78530a: FL 59 TL 15587360 HSR 641166863 J 455122196 LSR 2 981586141 DLSR 1538485973 [1557513269.355755] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513269.355763] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Confirmed peer address as 174.204.14.29:10496 [1557513269.355768] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Switching local interface to 209.182.216.71:20065 [1557513269.376119] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 201 [1557513269.376149] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 60b442ee: FL 185 TL 2381849 HSR 1868471658 J 3454737373 LSR 3540730944 DLSR 753951698 [1557513269.376164] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513269.967467] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20064]: Confirmed peer address as 174.204.14.29:10523 s[1557513274.064040] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SRTCP output wanted, but no crypto suite was negotiated [1557513274.064063] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513274.064084] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 2043506601 PC 348629863 OC 1509649344 NTP TS 4002640369/1 330623176=4002640369.309810 [1557513274.064090] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 1b7ce7ec: FL 17 TL 10773844 HSR 2006802227 J 3496394774 LSR 1855863287 DLSR 207902258 [1557513274.064105] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add 1b7ce7ec) - deleting SSRC 9fda8 18d [1557513274.064114] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding w[1557513274.337285] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513274.337341] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 3588538011 PC 4113250253 OC 3762757735 NTP TS 3675530209/ 2088363704=3675530209.486235 [1557513274.337352] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 1904c0d9: FL 190 TL 1277330 HSR 3700606795 J 2076634040 LSR 2826708796 DLSR 3084916091 [1557513274.337367] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add 1904c0d9) - deleting SSRC d4b12 78d [1557513274.337378] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding itch complete[1557513278.796840] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513278.796890] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 1179546147 PC 253436483 OC 401514196 NTP TS 591585900/307 4094117=591585900.715743 [1557513278.796900] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about f25fab39: FL 196 TL 1861219 HSR 1237831119 J 2726348211 LSR 3841137799 DLSR 843496811 [1557513278.796913] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add f25fab39) - deleting SSRC bb949 38e [1557513278.796921] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513279.310594] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SRTCP output wanted, but no crypto suite was negotiated [1557513279.310626] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513279.310646] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 624630617 PC 1167418798 OC 2870042862 NTP TS 2807429790/9 64181525=2807429790.224491 [1557513279.310652] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about bb84140b: FL 193 TL 5406510 HSR 1470777656 J 207786249 LSR 3684806947 DLSR 307709608 [1557513279.310669] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add bb84140b) - deleting SSRC 3813d 27e [1557513279.310679] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513279.678189] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20064]: SRTP output wanted, but no crypto suite was negotiated [1557513283.537471] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513283.537514] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 1085702245 PC 129819128 OC 1350967259 NTP TS 967851883/21 63508869=967851883.503731 [1557513283.537521] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 169e1507: FL 46 TL 3777017 HSR 3092777669 J 3495001522 LSR 3630093505 DLSR 120738828 [1557513283.537537] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add 169e1507) - deleting SSRC 468fa 6ae [1557513283.537546] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513284.300047] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513284.300094] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 3660274611 PC 3339904092 OC 976632286 NTP TS 3250964209/2 354170892=3250964209.548123 [1557513284.300103] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about a9cab1ac: FL 9 TL 7152183 HSR 1071025251 J 612018816 LSR 12 43749409 DLSR 721434160 [1557513284.300119] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add a9cab1ac) - deleting SSRC e7313 af6 [1557513284.300130] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding bye [1557513286.941239] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'delete' from 127.0.0.1:49479 [1557513286.941278] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'delete' from 127.0.0.1:49479: { "call-id": "05c7dcfe-06f3-4630-9799-0557b76cc340", "received-from": [ "IP4", "174.204.14.29" ], "from-tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "command": "delete" } [1557513286.941434] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Scheduling deletion of call branch 'c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00' (via-branch '') in 3 0 seconds [1557513286.941444] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: delete time = 0.000152 sec [1557513286.941474] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'delete' from 127.0.0.1:49479 [1557513286.941618] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'delete' to 127.0.0.1:49479: { "created": 1557513231, "created_us": 973175, "last signal": 1557513264, "SSRC": { "3046983612": { }, "2450372863": { }, "1622426350": { }, "2930250248": { }, "419741913": { }, "579439031": { }, "136 0672478": { }, "3145995275": { }, "125536563": { }, "2848633260": { }, "3762016578": { }, "379458823": { }, "1552885521": { }, "3078620951": { }, "12997 31210": { }, "461170668": { }, "2009301833": { }, "3253290075": { }, "1198441744": { }, "4066356025": { }, "2006816651": { } }, "tags": { "c71ac9de-9bf5- 4f39-9fda-6bd3aaf97d00": { "tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "created": 1557513231, "in dialogue with": "2684eff8-3602-41ed-aca2-9f7938362462", "me dias": [ { "index": 1, "type": "audio", "protocol": "UDP/TLS/RTP/SAVP", "streams": [ { "local port": 20064, "endpoint": { "family": "IPv4", "address": "174.204. 14.29", "port": 10523 }, "advertised endpoint": { "family": "IPv4", "address": "100.81.62.76", "port": 5079 }, "crypto suite": "AES_CM_128_HMAC_SHA1_80", "last packet": 1557513286, "flags": [ "RTP", "filled", "confirmed", "kernelized", "no kernel support", "DTLS fingerprint verified" ], "SSRC": 2006816651, "stats": { " packets": 595, "bytes": 40090, "errors": 0 } }, { "local port": 20065, "endpoint": { "family": "IPv4", "address": "174.204.14.29", "port": 10496 }, "advertised endpoint": { "family": "IPv4", "address": "100.81.62.76", "port": 5080 }, "crypto suite": "AES_CM_128_HMAC_SHA1_80", "last packet": 1557513284, "flags": [ "RTCP ", "filled", "confirmed", "kernelized", "no kernel support" ], "SSRC": 2006816651, "stats": { "packets": 15, "bytes": 1306, "errors": 0 } } ], "flags": [ "initi alized", "send", "recv", "DTLS-SRTP", "DTLS role passive", "ICE controlling" ] } ] }, "2684eff8-3602-41ed-aca2-9f7938362462": { "tag": "2684eff8-3602-41ed-aca2- 9f7938362462", "created": 1557513231, "in dialogue with": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "medias": [ { "index": 1, "type": "audio", "protocol": "UDP/TL S/RTP/SAVP", "streams": [ { "local port": 20052, "endpoint": { "family": "IPv4", "address": "97.47.70.223", "port": 4667 }, "advertised endpoint": { "family": " IPv4", "address": "10.255.81.66", "port": 5097 }, "crypto suite": "AES_CM_128_HMAC_SHA1_80", "last packet": 1557513284, "flags": [ "RTP", "filled", "confirmed", "kernelized", "no kernel support", "DTLS fingerprint verified" ], "SSRC": 2009301833, "stats": { "packets": 55, "bytes": 3410, "errors": 0 } }, { "local port": 20053, "endpoint": { "family": "IPv4", "address": "97.47.70.223", "port": 4640 }, "advertised endpoint": { "family": "IPv4", "address": "10.255.81.66", "port": 5098 }, "last packet": 1557513283, "flags": [ "RTCP", "filled", "confirmed", "kernelized", "no kernel support" ], "SSRC": 2009301833, "stats": { "packets": 10, "bytes": 936, "errors": 0 } } ], "flags": [ "initialized", "send", "recv", "DTLS-SRTP", "DTLS role passive" ] } ] } }, "totals": { "RTP": { "packets": 650, "bytes": 43500, "errors": 0 }, "RTCP": { "packets": 25, "bytes": 2242, "errors": 0 } }, "result": "ok" } [1557513287.087244] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513287.087244] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513287.087282] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 1725021880 PC 1560319321 OC 3364521432 NTP TS 1121433750/3398500622=1121433750.791275 [1557513287.087290] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 9ba3be3b: FL 18 TL 4080170 HSR 3714308681 J 2085763389 LSR 1453166610 DLSR 3867570679 [1557513287.087303] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add 9ba3be3b) - deleting SSRC 77b8933 [1557513287.087311] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513287.087324] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 201 [1557513287.087328] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 477b6daa: FL 136 TL 7495912 HSR 3723043622 J 76729872 LSR 791112841 DLSR 2264873880 [1557513287.087332] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add 477b6daa) - deleting SSRC b7800717 [1557513287.087338] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513287.165301] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513287.165344] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 4203649664 PC 1900914033 OC 1842228915 NTP TS 2577169143/1958584749=2577169143.456018 [1557513287.165355] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 643dc91f: FL 113 TL 8548413 HSR 913668020 J 2977588542 LSR 2353716277 DLSR 1468396047 [1557513287.165370] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add 643dc91f) - deleting SSRC e03bd142 [1557513287.165382] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513287.165396] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 201 [1557513287.165403] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 7d2457ab: FL 45 TL 2024736 HSR 174032952 J 3137856066 LSR 2564654212 DLSR 844577710 [1557513287.165410] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add 7d2457ab) - deleting SSRC 476ec510 [1557513287.165418] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513316.000248] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Call branch 'c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00' (via-branch '') deleted, no more branches remaining [1557513316.000290] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Final packet stats: [1557513316.000297] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --- Tag '2684eff8-3602-41ed-aca2-9f7938362462', created 1:25 ago for branch '', in dialogue with 'c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00' [1557513316.000308] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: ------ Media #1 (audio over UDP/TLS/RTP/SAVP) using iLBC/8000 [1557513316.000318] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20052 <> 97.47.70.223:4667 , SSRC 77c38349, 55 p, 3410 b, 0 e, 32 ts [1557513316.000325] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20053 <> 97.47.70.223:4640 (RTCP), SSRC 77c38349, 12 p, 1120 b, 0 e, 29 ts [1557513316.000330] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --- Tag 'c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00', created 1:25 ago for branch '', in dialogue with '2684eff8-3602-41ed-aca2-9f7938362462' [1557513316.000339] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: ------ Media #1 (audio over UDP/TLS/RTP/SAVP) using iLBC/8000 [1557513316.000347] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20064 <> 174.204.14.29:10523, SSRC 779d978b, 595 p, 40090 b, 0 e, 30 ts [1557513316.000355] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20065 <> 174.204.14.29:10496 (RTCP), SSRC 779d978b, 17 p, 1490 b, 0 e, 29 ts [1557513316.000521] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sending DTLS packet

2 1

DMQ re-ordering concern
by Julien Chavanton 11 May '19

11 May '19

Yesterday at Kamailio World there was an interesting question, raising a concern about replication of user location and DMQ in general, we should track it in the mailing list and use it as a feature request and I can share my initial thoughts on the topic. The concern was about DMQ reordering transactions. With SIP registration this seems like a minor concern, high responsiveness at the cost of potential small inconsistency seems like a good option. Considering that, the state will automatically correct itself and that this can be controlled using the expiration timer. This will become even more insignificant when the replicas can only be used as primary server failover. (it becomes a very small concern only when the primary server fails) With other type of data re-ordering may result in more problematic use cases. My impression is that DMQ should try to be good only with volatile data, data that will expire anyway, like caching with expiration, registration, Dialogs, etc. I think it may be worth looking at adding support for ordering/queuing in DMQ. Performance and simplicity could be maintained by doing mainly transactions in parallel with a configurable re-ordering best effort to minimize the impact of the problem. For example : 1- trying to re-order for a configurable amount of time. (32 seconds to match the RFC for SIP re-transmissions) 2- adding tractability for failed transactions. 3- the possibility to trigger a re-sync. Another option is to look at preserving strict ordering but I can imagine this could add more problems in some cases ...

2 4

Question regarding NAPTR and RURI
by Schneuwly Mathias RUAG 10 May '19

10 May '19

Hi I've a question about NAPTR lockup and rewriting the RURI. Following a little example: - Alice at location A tries to call Bob at location B - location A uses the RURI naptr.locationB.com to reach location B - this NAPTR resolves to hostb.locationB.com and host B is listening on port 5070 hostb has the following alias defined: - udp:hostb.locationB.com:5070 The problem now is, that the incoming RURI is "sip:bob@naptr.locationB.com" and the check to myself is not True. Is there a possibility to rewrite the RURI to the result of the NAPTR query? If I would add a second alias like "udp:naptr.locationB.com:5070" a check to myself would still fail as the RURI does not contain a port. So I would have to add an alias like "udp:naptr.locationB.com" without port, then it would work.? Is this the way to go? Because the documentation says it is important to include the port, otherwise the loose_route() will not work as expected... Thanks for your replies! Mathias

1 0

Help identifying mem leaks
by Duarte Rocha 10 May '19

10 May '19

Greetings, Recently one of my Kamailio's started started answering with a lot of 500 replies to requests because of insufficient private memory. The service Kamailio had been running for 6 months now. In order to solve this i increased the private memory settings and restarted the Kamailio. Before that, when searching a bit about the issue i found an old thread where Daniel suggested using " kamcmd corex.pkg_summary idx 1" to check for pkg leaks in the sip worker process. I ran this before doing the restart in order to identify leaks and found the summary in the logs. However, i am having a hard time interpreting this summary and being able to know if i have or not a mem leak. What should i be looking for? This is the beggining of the summary : qm_status: (0x7ff3c8cef010): qm_status: heap size= 8388608 qm_status: used= 928576, used+overhead=1464080, free=6924528 qm_status: max used (+overhead)= 1488472 qm_status: dumping all alloc'ed. fragments: qm_status: 0. N address=0x7ff3c8d286d8 frag=0x7ff3c8d286a0 size=1024 used=1 qm_status: alloc'd from core: core/str_hash.h: str_hash_alloc(59) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed qm_status: 1. N address=0x7ff3c8d28b40 frag=0x7ff3c8d28b08 size=256 used=1 qm_status: alloc'd from core: core/str_hash.h: str_hash_alloc(59) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed qm_status: 2. N address=0x7ff3c8d28ca8 frag=0x7ff3c8d28c70 size=184 used=1 qm_status: alloc'd from core: core/counters.c: cnt_hash_add(332) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Any help? Best Regards, Duarte Rocha

1 0

How to configure Kamailio 5.1 with Asterisk pjsip
by przeqpiciel 10 May '19

10 May '19

Hi, There is any working examples how to configure dispatcher module of Kamailio? I have working registrations but working in half dialing function.Signalling are come to asterisk and rtp start sending to my IP but i cant hear anything. Asterisk are doing registrar job - i dont know if it is good idea, maybe not.

5 11

RTP multiplexing, batching and compression (KamailioWorld followup)
by Stefan Sayer 09 May '19

09 May '19

Hi, as a followup to the 5 min 5 slides talk on Tuesday at the Kamailio world conference, I wanted to send links to that implementation, as I did get some interest about this afterwards. This transparent RTP mux gateway implementation in SEMS is a multiplexing, batching and compression mechanism, which can save quite some bandwidth on low bitrate codecs - can be useful if you need to work with very low bitrate links, e.g. on satellite (I tested with 10 G729 calls instead of 5 on a 128kbit sat link). It's a bit like ROHC (RFC3095) but on application level, only for RTP, and by using configuration for IP addresses instead of negotiation. Or as osmux, but without separate negotiation (but in-band), and for all codecs, and SIP. Or iax2 trunking mode, but with normal SIP/RTP. https://github.com/sems-server/sems/blob/sayer/rtp-mux/doc/rtp_mux/rtp_mux_… https://github.com/sems-server/sems/blob/sayer/rtp-mux/doc/rtp_mux/rtp_mux.… For now it's in the sayer/rtp-mux branch, but it's planned to merge it to master, possibly with simplification of the configuration. On another note, many thanks to the organizers of the Kamailio World conference! It's both really nice to meet, and very useful, and I think as well individually as well as for the community as a whole. Best Regards, Stefan

1 0

CRM / Windows integration
by Mark Boyce 09 May '19

09 May '19

Hi All Does anyone have any pointers for integrating Kamailio with CRMs (Dynamics / Salesforce / Zoho) or native Windows? We’re looking for basic click to dial / screen pop. Thanks Mark -- Mark Boyce Dark Origins Ltd

4 5

RE-invite in case of SIP is Active
by Gaurav Bmotra 09 May '19

09 May '19

HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 How can i use Re-invite in kamailio if SIP is active so that call can reconnect in case of network loss plz help -- *Regards:* Gaurav Kumar

2 2

Call reconnect in case of loss network
by Gaurav Bmotra 09 May '19

09 May '19

HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 when i call from one to another extension , in case of loss network it disconnect the call or stuck there , how can i make it as re-connection when network loss , and it will try to reconnect for some seconds like 30 sec "Example for callingn skype it reconnectcall in case of lossnetwork " plz help -- *Regards:* Gaurav Kumar

1 0

Sending in-dialog requests from custom module's C code
by Ivan Ribakov 09 May '19

09 May '19

I’m working on a custom Kamailio module where I need to send new-dialog and in-dialog requests on timer (hence I’m being forced to generate new messages from C code). So far I’ve been using modules/tm/tm_load.h defined interface to generate messages and handle callbacks. New dialog messages are sent and processed normally. To do that I’m: 1. Calling set_uac_request() to define request parameters 2. Calling tmb.t_request_outside() to send it outside any existing dialog I was able to send in-dialog requests (or so I thought) in a similar fashion, but I soon realised that responses to those requests were dropped because they couldn’t be matched against any known existing transaction. I’m attaching log messages that I believe support this theory and I’ve also observed UDP retransmissions. tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 50576 label 0 branch 0 tm [t_lookup.c:990]: t_reply_matching(): no matching transaction exists tm [t_lookup.c:993]: t_reply_matching(): failure to match a transaction tm [t_lookup.c:1088]: t_check_msg(): msg (0x5abcb50) id=1 global id=1 T end=(nil) tm [t_reply.c:2195]: reply_received(): transaction not found - (branch -1) The way I’m currently generating in-dialog requests is very similar to what tmb.t_request_outside() does, the main difference being that I do the dialog setup manually, based on the call-ID, cseq and from/to tags (I’m tracking transaction identifiers separately) and then pass resulting uac_req_t to tmb.t_request_within() - https://gist.github.com/IvanRibakov/3302cb286b1f4b786d109b406f2435a2 <https://gist.github.com/IvanRibakov/3302cb286b1f4b786d109b406f2435a2> Now, the question part - does anyone know what I’m doing wrong/missing? As I mentioned, when looking at the generated message bodies, they look ok to me (left - initial request that started the dialog, right - first in-dialog request), so I’m guessing I’m missing some Kamailio internal steps needed to register new transaction. SIP flow (up to the point when first UDP retransmission happens) I apologise in advance for the bulky question and will be extremely thankful for any guidance. Regards, Ivan

2 2

GDB debugging Kamailio 5.2.2 IMS authorization failed by the SCSCF
by Woscek, Martin W. 08 May '19

08 May '19

Hi, Looking for some advice running the debugger and setting breakpoints in Kamailio modules such as the SCSCF. Example: We were trying to watch the authenticate() function: 714 int authenticate(struct sip_msg* msg, char* _realm, char* str2, int is_proxy_auth) { Our Kamailio 5.2.2 code was compiled and linked from source with the -g option already enabled by default. We also dropped our Workers to 1 from 16 in the scscf.xml file for the SCSCF to limit the thread count and get the following list of threads below: We tried the first cdp threads: 8462, 8463, and 8464 and while we could set a break point at line 714 for the authenticate() function, it didn't actually break there at all. Question: What pid will we use to catch the authenticate() called during an initial registration from a boghe client? Anything special we may be missing, gdb related or Kamailio settings? kamcmd -s /run/kamailio_scscf/kamailio_ctl kamcmd 1.5 Copyright 2006 iptelorg GmbH This is free software with ABSOLUTELY NO WARRANTY. For details type `warranty'. kamcmd> ps 8456 main process - attendant 8457 udp receiver child=0 sock=10.0.20.1:6060 8458 slow timer 8459 timer 8460 secondary timer 8461 ctl handler 8462 cdp worker child=0 8463 cdp receiver peer unknown 8464 cdp_receiver_peer=hss522.irisims.org 8465 cdp_acceptor 8466 cdp_timer 8470 notification_worker 8471 notification_worker 8472 notification_worker 8473 notification_worker 8477 JSONRPCS FIFO 8478 JSONRPCS DATAGRAM Any advice appreciated. Thanks, Martin Woscek

2 1

http_client_query module?
by Andrew Chen 08 May '19

08 May '19

Hey guys, Need help here. What module besides tls.so I need to load in order to get http_client_query to work? The module doc says tls.so: #!ifdef WITH_TLS loadmodule "tls.so" #!endif but I keep getting this when doing config check: root@ashmaintpsg51:/etc/kamailio# kamailio -c kamailio.cfg 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 322, column 71: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 323, column 45: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 324, column 42: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 325, column 48: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 326, column 44: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 327, column 71: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 328, column 67: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 329, column 41: Can't set module parameter 0(12971) ERROR: <core> [core/cfg.y:3309]: yyparse(): cfg. parser: failed to find command http_client_query (params 2) 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 859, column 105: unknown command, missing loadmodule? and I have the package installed on my ubuntu 18.04: ii kamailio-*tls*-modules:amd64 5.1.2-1ubuntu2 amd64 TLS support for the Kamailio SIP server (authentication, transport) thoughts? -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 3

Udp errors on High Availability Setup
by Duarte Rocha 08 May '19

08 May '19

Greetings, I have two Machines with Kamailio in a High Availability (HA) configuration. Both machines are active and processing calls and both have one HA IP that can jump to the other machine in case of failure. I also have probing for NAPs configured with each IP as the socket. So, in each machine i have this configuration : listen=udp:HA_Address1 listen=udp:HA_Address2 NAP 1 being probed from socket HA_Address1 NAP 1 being probed from socket HA_Address2 I also have /proc/sys/net/ipv4/ip_nonlocal_bind with value 1. Everything works correctly with this configuration. However when the probing is sent for the NAP' my logs get filled with the same error messages. udp_send(): sendto(sock, buf: 0x7f2d4c91fa18, len: 340, 0, dst: (xxx.xxx.xxx.xxx:xxx), tolen: 16) - err: Invalid argument (22) CRITICAL: <core> [core/udp_server.c:602]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed The error happens because it is trying to probe a NAP using a socket that isn't active at the moment in the machine. Is there a way for this error not to be logged or be disabled without reducing Log Level to suppress CRITICAL errors? I think this is a standard High Availability configuration. Is there some setting that can disable this kind of errors? Best Regards, Duarte Rocha

3 5

HELP ME
by khadime gaye 07 May '19

07 May '19

Good morning, How I can resolve this problem ? << ERROR: Could not load the script in /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL' >>

2 1

implementing custom authentication method
by Walter.Martín Villalba 03 May '19

03 May '19

Dear list, I'm implementing a custom method for user authentication and would like to kindly ask for some guidance regarding the server side changes that may be needed in order to keep things running smoothly. The idea is to replace the standard user/pass-based digest authentication with a custom one based on Open ID Connect (OIDC) access tokens. I already have some pieces working (receiving access token in REGISTER message, validating the token by calling custom python routine, etc.), but I'm missing the bits needed to keep track of users who have successfully authenticated. While looking at sipcomm.cfg I found calls to *www_authenticate* (defined in modules/auth_db/authorize.c), which in turn calls to *mark_authorized_cred*, *check_auth_hr* (or *auth_check_hdr_md5*), and *generate_avps*,. Some of these functions are indeed changing some values in the DB and/or in memory, and this is where I would like some guidance. Is there any documentation I could look at which hopefully explains what DB values and/or data structures in memory need to be updated in order to keep track of successfully authenticated users? I believe with this last piece I'd be able to wrap up my custom authentication implementation. Thanks in advance! Regards, Walter Martín.

1 0

Kamailio bypassing RTP media
by Ali Taher 03 May '19

03 May '19

Hi everyone, I'm new to kamailio. I plan to setup a sip server cluster, Does someone can give me some suggestions if I can use kamailio as front server, which handle sip message and bypass rtp media message to backendserver, this mean the kamailio handle only sip message, the rtp packets are sent directed to backend server. The sip sequences is 1.softswitch send invite to kamailio 2.kamailio send invite to backend server. 3.backend server return 200 ok to kamailio 4.kamailio return 200 ok to softswitch, the ip and port in sip is from the backend server's msg, this make the softswitch send rtp direct to backend server. 5.rtp packets are transfered between softswitch and backend server. .... 6.backend server send bye to kamailio 7.kamailio send bye to softswitch. Any help is appreciated. Regards, Ali Taher

4 5

forcing socket doesn't work for ACK
by David Villasmil 03 May '19

03 May '19

Hello guys, I have public and private IPs, and i need to force the sending socket replying to internal out externals endpoint, so i'm using this: function to check the destination (i tried using $sndto(ip) but it always returns NULL): route[CHECK_DEST_NET] { if ( $du=~"sip:172\..*" ) { setflag(FLAG_TO_PRIVATE); xlog("L_ERR", "[CHECK_SOURCE] Packet going to PRIVATE -> [$du]\n" ); } else { setflag(FLAG_TO_PUBLIC); xlog("L_ERR", "[CHECK_SOURCE] Packet going to PUBLIC -> [$du]\n" ); } return; } on my relay: route(CHECK_DEST_NET); if (isflagset(FLAG_TO_PRIVATE)) { xlog("L_ERR", "[RELAY] forcing socket to PRIVATE NET\n" ); force_send_socket(LISTEN_INSIDE_IF:LISTEN_INSIDE_PORT); } else { xlog("L_ERR", "[RELAY] forcing socket to PUBLIC NET\n" ); force_send_socket(LISTEN_OUTSIDE_IF:LISTEN_OUTSIDE_PORT); } This works nicely, except for ACKs and potentially other packets sent statelessly? Anyone knows how to fix this? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

6 26

run some extra things before relying message
by Mojtaba 02 May '19

02 May '19

Hello, I have encountered with problem in Kamailio. At first glance it looks simple. Considering the Data Lump feature in Kamailio, All changes in SIP body, does not apply quickly. So if we are using RTPEngine or RTPProxy, How we can get body message (Actually IP and port from SDP after rtpengine_manage function) before relaying message? Suppose i want get some information and run some extra things before relying message? I just added my extra things in t_on_reply block, but it's strange why it doesn't work before relaying! -Mojtaba Esfandiari.S

2 2

Problem with topoh
by Phillman25 Kyriacou 02 May '19

02 May '19

Dear List I have an issue with topoh module. Specifically for in-dialog requests, if the encoded uri is altered i.e. initial encoded contact uri is Contact: <sip:X.X.X.X;line=sr-huAWi7oPBYsF.cC0L2lWOUlPwbSWOUl6wURTw24W> and UAS sends an in-dialog INVITE like this adding tgrp and user=phone: INVITE sip:X.X.X.X;line=sr-huAWi7oPBYsF.cC0L2lWOUlPwbSWOUl6wURTw24W;tgrp=1069912311;user=phone Kamailio cant seem to match/decode this and responds "404 Not Found" I have also set: modparam("topoh", "uri_prefix_checks", 0) As specified here: https://www.kamailio.org/docs/modules/4.4.x/modules/topoh.html#idp37582868 But still doesn't work. Is there a bug with this module? # ==================================================== # ================ TOPOLOGY HIDING =================== # ==================================================== modparam("topoh", "mask_key", "secret") modparam("topoh", "mask_ip", "X.X.X.X") modparam("topoh", "mask_callid", 0) modparam("topoh", "uparam_name", "line") modparam("topoh", "uparam_prefix", "sr-") modparam("topoh", "vparam_name", "branch") modparam("topoh", "vparam_prefix", "z9hG4bKsr-") modparam("topoh", "uri_prefix_checks", 0) Thanks in advance for your support Phillip

2 4

Stack smash error when using auth_ephemeral with KEMI
by Grant Bagdasarian 02 May '19

02 May '19

Hello, I was trying out the auth_ephemeral module inside the xhttp event_route in Python KEMI, and discovered a possible bug. https://pastebin.com/4Ghumfis The following will result in the stack smash error. Whenever I leave out the unixtimestamp the module will complain about not being able to convert to an int, which make sense, but doesn't lead to a stack smash error and a complete crash of Kamailio. KSR.auth_ephemeral.autheph_authenticate("1576734089:grant", "1234"). According to the documentation this module can only be used in REQUEST_ROUTE, but I found a presentation online which also used this module and that particular function in event_route, which is what I need for WebSocket authentication. Hope this helps. Please let me know if you need more information. Regards, Grant

2 1

Kamailio 5.2.2 IMS authorization failed by the SCSCF
by Woscek, Martin W. 02 May '19

02 May '19

Hi, We are getting a SIP 401 Unauthorized -Challenging the UE error originating by the SCSCF. It looks to be related to the nonce/cnonce. Is there something we have misconfigured in our SCSCF module? Log output: May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18489]: DEBUG: ims_auth [authorize.c:786]: authenticate(): Nonce or response missing: nonce len [-108158976], response16 len[1873606560] May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: DEBUG: ims_auth [utils.c:168]: get_nonce_response(): Calling find_credentials with realm [ims522.irisims.org] May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: DEBUG: ims_auth [utils.c:195]: get_nonce_response(): Found nonce response May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: INFO: ims_auth [authorize.c:827]: authenticate(): uri=sip:irisims.org nonce=9863c63855c8dd888c5c18e79fc61156 response=ce7ce73872b92aff2defa76dd1aff1ac qop=auth-int nc=00000001 cnonce=4fe4daacd69fa3d609c8b396d331eae6 hbody=d41d8cd98f00b204e9800998ecf8427e May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: DEBUG: ims_auth [rfc2617.c:135]: calc_response(): calc_response(_ha1=7a1b479a0513914e71bcd6d005b3e17d, _nonce=9863c63855c8dd888c5c18e79fc61156, _nc=00000001,_cnonce=4fe4daacd69fa3d609c8b396d331eae6, _qop=auth-int, _auth_int=1,_method=REGISTER,_uri=sip:irisims.org,_hentity=d41d8cd98f00b204e9800998ecf8427e) Thanks, Martin

2 1

New voisonics.com - Kamailio related products and services
by David Cunningham 02 May '19

02 May '19

Hello, Usually it's our client's VoIP technology being updated. Fortunately we found a little spare time to completely revamp the Voisonics website as well! https://voisonics.com/ Voisonics provides products and services in the commercial VoIP and hosted PBX sector; in particular relating to Kamailio, Asterisk, and Enswitch. The business has been specialising in this area for almost 10 years. Please feel free to contact me off-list with any questions. Kind regards, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

1 0

Support for JWT token authentication?
by David Dean 01 May '19

01 May '19

I came across this thread from Jan 2015 discussing the addition of JWT token authentication to Kamailio: https://github.com/kamailio/kamailio/issues/29 Was JWT authentication ever added? Or has anyone else on this list got JWT working somehow?

1 0

CSeq not incrementing - app_ruby
by Andrew White 01 May '19

01 May '19

Hi all, I’m still building with app_ruby and loving it - for the most part! I’m having an issue with one of my providers responding with a 482 on auth invite. Researching, it appears that my CSeq isn’t incrementing and this is the reason for the issue. I already have the relevant flag set in my kamailio.cfg however: modparam("dialog", "track_cseq_updates", 1) During my uac_auth, I’ve tried manually setting the diff, as well as running msg_iflag_reset (as suggested by Daniel - https://github.com/kamailio/kamailio/issues/1359 <https://github.com/kamailio/kamailio/issues/1359> - unfortunately the function isn’t exported in KEMI): if KSR::UAC.uac_auth() then KSR.info("UAC authed, relaying") #KSR::COREX.msg_iflag_reset("UAC_AUTH") KSR::PV.sets("$dlg_var(cseq_diff)", "1") KSR.info("CSeq diff: #{KSR::PV.gete("$dlg_var(cseq_diff)")}") KSR::TM.t_relay() Unfortunately in both cases the actual CSeq doesn’t increment in the second INVITE: INVITE 1 (unauthenticated): > CSeq: 102 INVITE INVITE 2 (with Authorization header): > CSeq: 102 INVITE I’m unsure if I’ve missed something silly around the track_cseq_updates flag, or if app_ruby somehow isn’t interacting with the dialog module when running uac_auth to trigger the increment? Thanks! Kind regards, Andrew

3 6

listening at link local IPv6 address
by Juha Heinanen 01 May '19

01 May '19

My K host is behind a mobile WiFi hotspot and I tried to make K listen at the IPv6 address fe80::6e29:95ff:fe7d:37e6/64 that the host has received from the hotspot. Using that address on listen line gives error: Apr 21 08:55:18 salmon /usr/bin/sip-proxy[6887]: ERROR: <core> [core/udp_server.c:395]: udp_init(): bind(8, 0x7ff97dfb3fe4, 28) on FE80:0:0:0:6E29:95FF:FE7D:37E6: Invalid argument Apr 21 08:55:18 salmon /usr/bin/sip-proxy[6887]: ERROR: <core> [core/udp_server.c:401]: udp_init(): might be caused by using a link local address, try site local or global I can ping that address like this: $ ping6 fe80::6e29:95ff:fe7d:37e6%wlp1s0 PING fe80::6e29:95ff:fe7d:37e6%wlp1s0(fe80::6e29:95ff:fe7d:37e6%wlp1s0) 56 data bytes 64 bytes from fe80::6e29:95ff:fe7d:37e6%wlp1s0: icmp_seq=1 ttl=64 time=0.030 ms I then tried to add %wlp1s0 suffix to the listen address, but got error: Apr 21 09:10:35 salmon sip-proxy[7896]: 0(7948) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 408, column 34: ip address, interface name or hostname expected Did I do something wrong? If not, why is it that K cannot listen at a link local address? It is something that Linux networking stack does not support? -- Juha

6 33

Pipelimit with clean_unused
by Cinthia Leung 30 Apr '19

30 Apr '19

Hello, What is your recommendation on using clean_unused with db defined pipes? >>pl_check("systemwide"); If I leave the system idle, they are eventually "cleaned" as expected. In our use case, that means failed calls unfortunately. I am trying to see if I can mix both db/static pipes with dynamic/run-time pipes. Thanks Cindy

1 1

Next hop of reply, in onreply_route
by Alex Balashov 30 Apr '19

30 Apr '19

I am fairly sure it’s been asked a number of times, but is there a way to know the destination, and more especially the transport, of the next hop of a reply message, viewable from onreply_route? For requests, we have $nh and many other PVs which in some way evidence where it’s going. And yes, I’m aware that this can be previewed from the onsend_routes*, but I need to amalgamate that knowledge with some transaction-specific data. Thanks! — Sent from mobile, with due apologies for brevity and errors.

2 2

"ERROR: reply too big" and ctl binrpc size
by Karsten Horsmann 30 Apr '19

30 Apr '19

Hi all, i use kamailio 5.0.7 in production and i noticed that i cant listen the dialogs after a few days of traffic on this server. Then i got "kamcmd dlg.list_ctx " - "ERROR: reply too big". So now my question - i want to debug my "sleeping dialogs" with dlg.list - how big should it change the values of without knowing the size of the current filled up binrpc stuff? Or is there a better option to debug dialog-module? modparam("ctl", "binrpc_max_body_size", ??) modparam("ctl", "binrpc_struct_max_body_size", ?) -- Cheers and see you in Berlin *Karsten Horsmann*

2 1

Re: [SR-Users] SIP basic flow question: R-URI in ACK
by Yufei Tao 30 Apr '19

30 Apr '19

Hi Daniel, Thank you very much for clarifying! Cheers, Yufei On 30 Apr 2019 at 08:10, <Daniel-Constantin Mierla <miconda(a)gmail.com>> wrote: Hello, there is an errata for that RFC, but it refers only to the BYE, where it adds back the transport=tcp, but the ACK R-URI should follow the same rules, see: * https://www.rfc-editor.org/errata/eid5294 Cheers, Daniel On 28.04.19 22:54, Yufei Tao wrote: Hi, I have a question regarding basic SIP flow for establishing a call session, and wonder if anyone can help me clarify. For normal call set up, INVITE-OK-ACK, from RFC3261, I believe the ACK’s R-URI should be a copy of the Contact header of the OK message. However in RFC3665 sections 3.1 and 3.2 for example, the ACK’s R-URI missed the parameters from the Contact header of the OK, e.g. In OK the Contact header is: Contact: <sip:bob@client.biloxi.example.com;transport=tcp> <sip:bob@client.biloxi.example.com;transport=tcp> And ACK the R-URI is: ACK sip:bob@client.biloxi.example.com SIP/2.0 Which has got the parameter ‘transport=tcp’ removed – why is this? In this case if the proxy connected to Bob handles ACK in the usual way as it would for all in-dialog requests, i.e. based on routing headers only and not extra processing, it’ll try to relay the ACK message to Bob using the default transport UDP which is not expected and will fail. Can anyone help explain why the parameters are removed in RFC3665 examples please? Have I missed anything? Thank you very much! Yufei _______________________________________________ Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

docker image for non x86 arch
by Sergey Safarov 30 Apr '19

30 Apr '19

Is the community involved in kamailio docker image for ARM architecture and others not x86 platforms? ARM ARM 64 IBM POWER IBM Z PowerPC 64 LE

3 3

SIP basic flow question: R-URI in ACK
by Yufei Tao 30 Apr '19

30 Apr '19

Hi, I have a question regarding basic SIP flow for establishing a call session, and wonder if anyone can help me clarify. For normal call set up, INVITE-OK-ACK, from RFC3261, I believe the ACK’s R-URI should be a copy of the Contact header of the OK message. However in RFC3665 sections 3.1 and 3.2 for example, the ACK’s R-URI missed the parameters from the Contact header of the OK, e.g. In OK the Contact header is: Contact: <sip:bob@client.biloxi.example.com;transport=tcp> And ACK the R-URI is: ACK sip:bob@client.biloxi.example.com SIP/2.0 Which has got the parameter ‘transport=tcp’ removed – why is this? In this case if the proxy connected to Bob handles ACK in the usual way as it would for all in-dialog requests, i.e. based on routing headers only and not extra processing, it’ll try to relay the ACK message to Bob using the default transport UDP which is not expected and will fail. Can anyone help explain why the parameters are removed in RFC3665 examples please? Have I missed anything? Thank you very much! Yufei

3 2

remove_record_route - bug?
by Sergiu Pojoga 29 Apr '19

29 Apr '19

Hi ppl, After a branch route fails, I need to re-evaluate the next destination and adjust RR parameters, if need be, like for example adjust the nat=yes|priv. failure_route[MANAGE_PSTN_FAILURE] { ... remove_record_route(); record_route(); route(NATMANAGE); ... route(RELAY); } The result is that the new branch is stripped of "Record-Route: <sip:MYIP>", but the params remained. Moreover, record_route hasn't been re-added as instructed in the failure route. Request headers of the initial branch that will eventually timeout and fail: 2019/04/27 18:29:18.034992 10.22.0.1:5060 -> 10.22.0.100:5060 INVITE sip:1514XXXXXXX@10.22.0.100 SIP/2.0 Record-Route: <sip:10.22.0.1;r2=on;lr=on;did=9e2.3dd2;nat=priv> Record-Route: <sip:65.XX.XX.1;r2=on;lr=on;did=9e2.3dd2;nat=priv> New request branch after failure: 2019/04/27 18:29:19.538292 65.XX.XXX.1:5060 -> 65.XX.XX.2:5060 INVITE sip:1514XXXXXXX@65.XX.XX.2 SIP/2.0 ;lr=on;did=9e2.3dd2;nat=priv> kamailio -v version: kamailio 5.1.6 (x86_64/linux) 7d1964 Much obliged. --Sergiu

4 5

Vide SIP conferencing kamailio with freeswitch
by Gaurav Bmotra 28 Apr '19

28 Apr '19

HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 i want a solution for video conferencing with kamailio as i am using freeswitch with it is there reference guide for that plz help -- *Regards:* Gaurav Kumar

2 1

Re: [SR-Users] [sr-dev] Vide SIP conferencing kamailio with freeswitch
by Henning Westerholt 28 Apr '19

28 Apr '19

Hello, please ask this usage related questions on our "sr-users" list. Kamailio is not doing much regarding the video calls. E.g. if your calls work for voice with kamailio, they should also work for video. All the setup needs to be done in freeswitch. I found a (outdated) howto with google: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video Maybe this helps. Henning Am 27.04.19 um 13:22 schrieb Gaurav Bmotra: HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 i want a solution for video conferencing with kamailio as i am using freeswitch with it is there reference guide for that plz help _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org<mailto:sr-dev@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

1 0

How to configure ipv6 in kamailio
by Santosh Kumar Dash 28 Apr '19

28 Apr '19

Hi All, I have a kamailio installed in my setup which is working properly in IPV4 configuration. I want to perform IPV6 configuration, I don't have any idea how to configure it, I have tried few things by changing *kamailio.cfg* file but did not work. I am sharing my *kamailio.cfg *file which is working fine in ipv4 Kindly help me to configure for IPV6. lets say my Ipv6 address is : 2402:3a80:d1a:f45a:fe44:2e54:ed2b:9bb2 Thanks,

3 3

Can't set module parameter
by Tmbc 26 Apr '19

26 Apr '19

Hi all Having this issue with kamailio: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <dst_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 119, column 50: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <grp_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 120, column 50: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <cnt_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 121, column 50: Can't set module parameter here is full conf file: https://dpaste.de/bP3E OS: NAME="CentOS Linux" VERSION="7 (Core)" Software version: kamailio 5.2.2 (x86_64/linux) 67f967 Thanks in advance, tmbc

2 1

Re: [SR-Users] Config to control TCP connection reuse
by R, Rajkumar (Raj) 24 Apr '19

24 Apr '19

I think elaborating on the setup may help someone to provide inputs on this. Below is the setup. Client 1 --> Kamailio proxy --> Backend SIP server Client 2 --> Kamailio proxy --> Backend SIP server Client 3 --> Kamailio proxy --> Backend SIP server After I register all three clients, see only one TCP connection is being used between "Kamailio proxy - Backend SIP server". Is there a way to achieve separate TCP connection to Backend SIP server from kamailio. thanks, raj From: R, Rajkumar (Raj) Sent: Monday, April 22, 2019 3:45 PM To: sr-users(a)lists.kamailio.org Subject: Config to control TCP connection reuse Hi, In my setup(kamailio in stateless mode to a backend server), for any number of client registrations kamailio (in stateless mode) is reusing the TCP connection. Let's say if I register 3 different clients through kamailio to my backend server - see only one TCP connection is used, which I do not want. I have gone through the core modules to see if any TCP config param exists to control on this, but couldn't find one. Could anyone help how to achieve different TCP connections to same destination in kamailio? thanks, raj

2 1

DMQ_USRLOC
by Gholamreza Sabery 24 Apr '19

24 Apr '19

Hi, Recently, I started to use dmq_usrloc module. I successfully set it up. Unfortunately, the documentation does not specify the behavior of dmq_usrloc module with other Kamailio modules. For example, I see a bunch of KDMQ messages are exchanged between nodes, but how the lookup() function behaves with dmq_usrloc, or what is it's effect on a db_mode of 3! Regards

5 6

JSON error - control character 0xd
by David Dean 23 Apr '19

23 Apr '19

I'm trying to send an SDP from a SIP client to Janus using JSON over HTTP. The problem is that Janus is erroring on the SDP content with the following error: JANUS HTTP: Get SDP for echo plugin: {#012 "janus": "error",#012 "error": {#012 "code": 454,#012 "reason": "JSON error: on line 9: control character 0xd near '\"v=0'"#012 }#012} Result code 200 I tried using {s.escape.common} to escape any quotations, commas etc but it doesn't appear to replace the control characters with newlines. Any suggestions on how best to remove these control characters so that Janus will accept the SDP over JSON? Here is the relevant section of kamailio.cfg: # HTTP: Request SDP for echo test plugin sdp_get("$avp(sdp)"); $var(res) = http_connect("janus", "/janus/$var(JANUS-ID)/$var(ECHO-ID)", "application/json", "{ \"janus\" : \"message\", \"transaction\" : \"testtesttest99\", \"body\" : { \"audio\" : true }, \"jsep\" : { \"type\" : \"offer\", \"sdp\" : \"$(avp(sdp){s.escape.common})\" }}", "$avp(janus-pluginsession)"); xlog("L_INFO", "JANUS HTTP: Get SDP for echo plugin: $avp(janus-pluginsession) Result code $var(res)\n");

3 6

Direction of replies
by Duarte Rocha 23 Apr '19

23 Apr '19

Greetings, Is there a way to know in reply_route if a reply has direction downstream or upstream regarding the first INVITE? I know there are vars and methods for finding this out on requests, but i can't find anything for replies. Thanks in advance, Best Regards, Duarte Rocha

2 1

CDR start time of a canceled call
by Jan-Hendrik Dörner 23 Apr '19

23 Apr '19

Hello, I just noticed that the logged „start_time" as well as „end_time" of a canceled call is set to the „invite time“. If done this way, it is harder to calculate the „ringing time“. Is there an option to change this behaviour (i.e. to log the time of the received „bye“ instead)? (If done so, the ringing time would always be the time between start and invite …) Or is there a special reason for this behaviour? My CDR settings are as followed: modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) # log unanswered / open dialogs modparam("acc", "cdr_on_failed", 1) # log also not answered calls modparam("acc", "cdr_extra", "inviteTime=$dlg_var(inviteTime)" modparam("acc", "cdr_log_enable", 1) # activate write to syslog modparam("acc", "cdr_facility", "LOG_LOCAL6") # syslog modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "time_mode", 0) #syslog timestamp modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) # needed for syslog modparam("acc", "log_missed_flag", FLT_ACCMISSED) # need for logging missed calls in syslog Regards Jan-Hendrik

1 0

Kamailio-5.2.1 - Issues in NDB_MONGODB module initialisation
by José Seabra 22 Apr '19

22 Apr '19

Hello there, I have NDB_MONGODB loaded in kamailio with following configurations: *loadmodule "ndb_mongodb"* *modparam("ndb_mongodb", "server", "name=mgs1;uri='mongodb://mongodbserver:27017/proxyregistrar'")* Kamailio breaks up on start up and prints the following messages: *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: <core> [core/modparam.c:83]: set_mod_param_regex(): 'ndb_mongodb' matches module 'ndb_mongodb'* *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: <core> [core/sr_module.c:711]: find_param_export(): found <server> in module ndb_mongodb [/usr/lib64/kamailio/modules/ndb_mongodb.so]* *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: <core> [core/modparam.c:99]: set_mod_param_regex(): found <server> in module ndb_mongodb [/usr/lib64/kamailio/modules/ndb_mongodb.so]* *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: ndb_mongodb [mongodb_client.c:155]: mongodbc_add_server(): added server[mgs1]=mongodb://10.225.121.12:27017/proxyregistrar <http://10.225.121.12:27017/proxyregistrar>* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [core/sr_module.c:942]: init_mod_child(): idx 21 rank 8: ndb_mongodb [tcp receiver (generic) child=3]* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 23* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: ALERT: <core> [main.c:739]: handle_sigs(): child process 3181 exited by a signal 11* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: ALERT: <core> [main.c:742]: handle_sigs(): core was generated* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [core/tcp_main.c:3513]: handle_ser_child(): dead child 16, pid 3181 (shutting down?)* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: INFO: <core> [main.c:764]: handle_sigs(): terminating due to SIGCHLD* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa6d0e0, 23, -1, 0x0) fd_no=23 called* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [main.c:766]: handle_sigs(): terminating due to SIGCHLD* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: ndb_mongodb [ndb_mongodb_mod.c:128]: mod_destroy(): cleaning up * Core dump back trace output: *(gdb) bt full* *#0 0x00007f3daa134f93 in _mongoc_client_new_from_uri () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#1 0x00007f3daa156553 in mongoc_uri_get_ssl () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#2 0x00007f3daa134f88 in _mongoc_client_new_from_uri () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#3 0x00007f3daa135029 in mongoc_client_new () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#4 0x00007f3daae6d75f in mongodbc_init () at mongodb_client.c:63* * rsrv = 0x7f3db0f86728* * __FUNCTION__ = "mongodbc_init"* *#5 0x00007f3daae73c46 in child_init (rank=3) at ndb_mongodb_mod.c:113* * __FUNCTION__ = "child_init"* *#6 0x00000000005470d7 in init_mod_child (m=0x7f3db0f85e88, rank=3) at core/sr_module.c:846* * __FUNCTION__ = "init_mod_child"* *#7 0x00000000005474b6 in init_child (rank=3) at core/sr_module.c:874* * ret = 32573* *#8 0x00000000004b3840 in fork_process (child_id=3, desc=0x7ffecc20c7f0 "udp receiver child=2 sock=10.225.121.93:5060 <http://10.225.121.93:5060>", make_sock=1) at core/pt.c:341* * pid = 0* * child_process_no = 3* * ret = -1* * new_seed1 = 1273263324* * new_seed2 = 1531871627* * sockfd = {14, 15}* * __FUNCTION__ = "fork_process"* *#9 0x0000000000424b9b in main_loop () at main.c:1631* * i = 2* * pid = 3807* * si = 0x7f3db123a988* * si_desc = "udp receiver child=2 sock=10.225.121.93:5060 <http://10.225.121.93:5060>\000\177\000\000P\310 \314\376\177\000\000\000e|\261=\177\000\000 \313 \314\376\177\000\000\063\225{\000\000\000\000\000\027\000\000\000\000\000\000\000\000\000\000 \000\000\000\000\000\000\200\000\000\000\000\000\217e|\261=\177\000\000`~x\000\000\000\000\000\230\304\022\261=\177\000"* * nrprocs = 4* * woneinit = 1* * __FUNCTION__ = "main_loop"* *#10 0x000000000042c4e9 in main (argc=11, argv=0x7ffecc20cc08) at main.c:2675* * cfg_stream = 0x2893010* * c = -1* * r = 0* * tmp = 0x7ffecc20e6fd ""* * tmp_len = 0* * port = 0* * proto = 2496* * options = 0x7697a0 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:"* * ret = -1* * seed = 173321635* * rfd = 4* * debug_save = 0* * debug_flag = 0* * dont_fork_cnt = 0* * n_lst = 0x7ffecc20cac0* * p = 0x7f3db177497d <intel_check_word+173> "H\205\300t\025\017\266P\003\017\266\312D9\371\017\204", <incomplete sequence \316>* * st = {st_dev = 19, st_ino = 21919, st_nlink = 2, st_mode = 16832, st_uid = 993, st_gid = 2, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1551179759, tv_nsec = 561093463}, st_mtim = {tv_sec = 1551179759,* *---Type <return> to continue, or q <return> to quit---* * tv_nsec = 561093463}, st_ctim = {tv_sec = 1551179759, tv_nsec = 561093463}, __unused = {0, 0, 0}}* * __FUNCTION__ = "main"* *(gdb)* NDB_MONGODB configuration is pretty simples and currently i'm not making any mongodb operations, i'm just loading the module and trying to connect against mongodb. Any thoughts on this issue? Thank you Best regards -- Cumprimentos José Seabra

2 1

rtpengine_offer() with ICE not updating o= and m= IP addresses
by David Dean 22 Apr '19

22 Apr '19

I'm using the following rtpengine_offer() to force the use of ICE relay and also replace o= and m= rtpengine_offer("replace-origin replace-session-connection ICE=force-relay RTP"); The SDP is being updated to include an ICE relay candidate, but the IP addresses in the o= and m= lines are not changing to the servers IP address (X.X.X.X). I also tried using the media-address parameter but that didn't change the behaviour. When I use rtpengine_answer() then it's not providing an ICE candidate nor changing o= or m=, which I imagine is expected. What am I missing? INVITE from the client (before rtpengine)------------------------------------------------------ INVITE sip:server.domain.com SIP/2.0Via: SIP/2.0/TCP 154.20.1.8:37520;branch=z9hG4bK-1710993570;rportFrom: <sip:username@server.domain.com>;tag=281751229To: <sip:server.domain.com>Contact: <sip:username@154.20.1.8:37520;transport=tcp>Call-ID: c715a8f9-4dd1-38d8-acbd-54153310f59fCSeq: 1756522847 INVITEContent-Type: multipart/mixed;boundary=7efec5a8-3311-0157-fc8b-9af106265507Content-Length: 1506Max-Forwards: 70Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERPrivacy: noneSupported: 100rel --7efec5a8-3311-0157-fc8b-9af106265507Content-Type: application/sdp v=0o=organization 1983 678901 IN IP4 154.20.1.8s=-c=IN IP4 154.20.1.8t=0 0a=tcap:1 RTP/AVPFm=audio 15418 RTP/AVP 114 115 0 101i=speecha=ptime:20a=minptime:1a=maxptime:255a=silenceSupp:off - - - -a=rtpmap:114 AMR-WB/16000/1a=imageattr:114 octet-align=0a=fmtp:114 octet-align=0a=rtpmap:115 AMR-WB/16000/1a=imageattr:115 octet-align=1a=fmtp:115 octet-align=1a=rtpmap:0 PCMU/8000/1a=rtpmap:101 telephone-event/8000/1a=fmtp:101 0-16a=pcfg:1 t=1a=sendrecva=rtcp-muxa=ssrc:2047236273 cname:e6dcab7948b70d52cd51b75e505c49eba=ssrc:2047236273 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2a=ssrc:2047236273 label:organization@audio INVITE from the server (after rtpengine)------------------------------------------------------ SIP/2.0 200 OKVia: SIP/2.0/TCP 154.20.1.8:37520;branch=z9hG4bK-1710993570;rport=37520;received=2.221.73.230From: <sip:username@server.domain.com>;tag=281751229To: <sip:server.domain.com>;tag=0ea7681ffd64b0ffcde1bff3393fd505.ae3aCall-ID: c715a8f9-4dd1-38d8-acbd-54153310f59fCSeq: 1756522847 INVITEContact: <sip:X.X.X.X:443;transport=tcp>Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERPrivacy: noneContent-Type: multipart/mixed;boundary=7efec5a8-3311-0157-fc8b-9af106265507Server: kamailio (5.2.2 (x86_64/linux))Content-Length: 1965 --7efec5a8-3311-0157-fc8b-9af106265507Content-Type: application/sdp v=0o=organization 1983 678901 IN IP4 154.20.1.8s=-c=IN IP4 154.20.1.8t=0 0a=tcap:1 RTP/AVPFm=audio 15418 RTP/AVP 114 115 0 101i=speecha=minptime:1a=maxptime:255a=silenceSupp:off - - - -a=imageattr:114 octet-align=0a=imageattr:115 octet-align=1a=pcfg:1 t=1a=rtcp-muxa=ssrc:2047236273 cname:e6dcab7948b70d52cd51b75e505c49eba=ssrc:2047236273 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2a=ssrc:2047236273 label:organization@audioa=rtpmap:114 AMR-WB/16000/1a=rtpmap:115 AMR-WB/16000/1a=rtpmap:0 PCMU/8000/1a=rtpmap:101 telephone-event/8000/1a=fmtp:114 octet-align=0a=fmtp:115 octet-align=1a=fmtp:101 0-16a=sendrecva=ptime:20a=ice-ufrag:kzgbFAnua=ice-pwd:hnEu9EwTrdIZnTWhYbxtmdugvra=candidate:dzkwmPcB28NPHfRA 1 UDP 16777215 X.X.X.X 30366 typ relay raddr X.X.X.X rport 30366a=candidate:dzkwmPcB28NPHfRA 2 UDP 16777214 X.X.X.X 30367 typ relay raddr X.X.X.X rport 30367a=sendrecva=ice-ufrag:uDE9wbvYa=ice-pwd:dx4CcIjqUNFfnRPJEF8GjatXdU

3 3

Config to control TCP connection reuse
by R, Rajkumar (Raj) 22 Apr '19

22 Apr '19

Hi, In my setup(kamailio in stateless mode to a backend server), for any number of client registrations kamailio (in stateless mode) is reusing the TCP connection. Let's say if I register 3 different clients through kamailio to my backend server - see only one TCP connection is used, which I do not want. I have gone through the core modules to see if any TCP config param exists to control on this, but couldn't find one. Could anyone help how to achieve different TCP connections to same destination in kamailio? thanks, raj

1 0

Best tutorial for beginners kamailio
by Gaurav Bmotra 22 Apr '19

22 Apr '19

Hi Plz sujjest best kamailio tutorial or any source where a beginner can start kamailio.. Thanks

2 1

push to talk over cellular application server and related media server implementation
by Thierry Luo 21 Apr '19

21 Apr '19

Hi All, I don't know if this is the appropriate place to ask the following questions: 1. Anyone please recommend open-source implementation of PoC (Push-to-talk Over Cellular) application server? Kamailio can do this? 2. Anyone please recommend open-source implementation of PoC related media server/replicator? Thierry Luo

3 2

Re: [SR-Users] authentication using auth_db
by arish haque 21 Apr '19

21 Apr '19

Hi David, This can solve the issue but depth(records) of that view would be too much. The architecture is like this: One kamailio fronting multiple asterisk. Each of asterisk boxes have their Specific users databases. Thus, I'm trying to create db connections inside request route or completely write my own authentication service if I get the user's Password. So, that would be great if I would be able to pass the db_url to auth_db module in request route. Let me know what you think. Thanks, Arish

2 1

Re: [SR-Users] authentication using auth_db
by arish haque 20 Apr '19

20 Apr '19

Hi David, Thanks for the reply. I checked out sqlops module, this will solve the problem only when I get the authentication password (secret) of the user inside request route. For this I tried checking kamailio pseudovars but haven't found one. Therefore, I am sticking with auth_db module.If this is possible by any other way do let me know. thanks, Arish

2 1

Kamailio, TLS, and LetsEncrypt
by Gaurav Bmotra 19 Apr '19

19 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i am trying to configure LetsEncrypt (tls) for kamailio sing following link https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ but unable to start kamailio ,i m getting these error ------->>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Apr 19 11:23:57 ubuntufresh systemd[1]: Starting Kamailio (OpenSER) - the Open Source SIP Server... Apr 19 11:23:58 ubuntufresh kamailio[5365]: INFO: <core> [core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Apr 19 11:23:58 ubuntufresh kamailio[5365]: INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) Apr 19 11:23:58 ubuntufresh kamailio[5365]: Listening on Apr 19 11:23:58 ubuntufresh kamailio[5365]: Aliases: Apr 19 11:23:58 ubuntufresh kernel: kamailio[5378]: segfault at 0 ip 00007f8d6fdac646 sp 00007ffe8e2337f8 error 4 in libc-2.27.so [7f8d6fcfb000+1e7000] Apr 19 11:23:58 ubuntufresh kamailio[5365]: ERROR: <core> [core/daemonize.c:303]: daemonize(): Main process exited before writing to pipe Apr 19 11:23:58 ubuntufresh systemd[1]: kamailio.service: Control process exited, code=exited status=255 Apr 19 11:23:58 ubuntufresh systemd[1]: kamailio.service: Failed with result 'exit-code'. Apr 19 11:23:58 ubuntufresh systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<------------------------------------ plz help -- *Regards:* Gaurav Kumar

3 5

auth by IP
by Yu Boot 19 Apr '19

19 Apr '19

Following code snippet from default kamailio.cfg never gives 403 if you smart enough to set "fromdomain" parameter on Asterisk to Kamailio's IP. How to fix it? I want password-based registration (which is OK now) and permit calls via Kamailio only from permitted IPs. # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; }

2 4

TLS implementation for kamailio
by Gaurav Bmotra 19 Apr '19

19 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i want to implement TLs for kamailio plz help -- *Regards:* Gaurav Kumar

3 2

Microsoft Direct Routing Assistance And Kamailio Integration
by Phillip Cooper 18 Apr '19

18 Apr '19

Hi All, I am looking for some assistance on setting up Microsoft Direct Routing with Kamailio as an SBC. I have a partial config working, I have rewrote the headers going to Direct routing and can place calls in to and out of Direct routing. I have problems with Call hang ups and choppy audio only on the audio stream from Direct Routing. 1) place a call in to direct routing, Hang the call up from the caller and Direct routing hangs up the call. But If you place a call in to Direct routing and hang up the call from direct routing the direct routing hangs up but the call is still active on the caller. IF you call from direct routing through kamailio to a callee and hang up from direct routing the call is hung up correctly. 2) on some calls made to or from direct routing, the audio stream coming from direct routing is sometimes choppy. Its always the audio from direct routing never the audio going to direct routing. My set up: Asterisk <-> Kamailio <-> Direct Routing SIP: Asterisk <-> Kamailio <-> direct routing RTP: Asterisk <-> Direct Routing Kamailio is rewriting the R-URI, TO, FROM, Contact headers going to Direct routing, as Microsoft need the sbc hostname to be in the contact header in a particular format. I am looking for help on the above or if someone would be kind enough to provide a working config that would be awesome! :) Many Thanks in advance for assistance. :) Phil --- This email has been checked for viruses by AVG. https://www.avg.com

1 0

authentication using auth_db module
by arish haque 18 Apr '19

18 Apr '19

Hi all, I'm trying to establish mysql database connection inside request route (at runtime) based on the type of user, as credentials are stored in multiple databases (may be different servers) for the users. Therefore, I am trying to find out that is there anyway to connect to mysql dynamically in request route or pass 'db_url' parameter dynamically to auth_db module. Hope this explains the requirement. Looking for suggestions. With Regards, Arish

2 1

How to append a body to send_reply() ?
by David Dean 18 Apr '19

18 Apr '19

Hello Kamailio gurus, I'm successfully using send_reply() with append_to_reply() to add some custom headers. Now I need to add some text into the reply body. How can I add a body to send_reply() ? Thx, Dave

2 7

After-parsing config file?
by David Villasmil Govea 18 Apr '19

18 Apr '19

Hello all, Kind of a weird question: is it possible to get a config file AFTER it’s been parsed? I.e: after it’s skipped all non-defined blocks, after substdef have been applied, etc. In the form of a normal config file Is this possible? -- DVG -- Imagination is more important than knowledge Albert Einstein

1 0

Use domain instead of IP in kamailio for register user
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i wan to to use domain instead of IP for registration in kamailio Plz help -- *Regards:* Gaurav Kumar

3 3

Does Kamailio provide any API for other program to create sip account
by Gaurav Bmotra 18 Apr '19

18 Apr '19

Hi i am Gaurav I am developing a sip VoIP app in Android, which want to provide the feature of allowing user to create sip account from client side. I use Kamailio as sip server. My thought is when user register to the the user account system, the user account system can communicate with Kamailio, create a sip account associated with the user and pass the sip account to the client, the client then can configure itself using the sip account. Does Kamailio provide any API or module for me to do this? Thanks! -- *Regards:* Gaurav Kumar

2 1

re-connect call in case of network issue
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 how can we implement re-connecting call instead of disconnect in case of network problem plz help thanks -- *Regards:* Gaurav Kumar

1 0

VRF routing with kamailio 5.2
by Jonathan Hunter 18 Apr '19

18 Apr '19

Hi All, Hope everyone in community is well. I was wondering if anyone could let me know if VRF is supported with kamailio 5.2 ? Essentially I am looking to put kamailio between two SBCs and bridge the networks together. So currently Server A 1.1.1.1 connects to Server B 2.2.2.2 and they pass calls fine, and I want to add kamailio in between without changing any IPs, so using VRF I have one network interface intA with 1.1.1.1 and the other intB with 2.2.2.2. What I want to do is listen on both interfaces, which is fine but bridge or force traffic between the two. Is this possible with kamailio? As doing force socket and listening on both 1.1.1.1 and 2.2.2.2 kamailio just loops and routes to itself, and force socket doesnt help. Do you have any suggestions? Would running two instances of kamailio on same server help or even introducing to separate kamailio instances and force routing between the two for requests. I am currently just testing with SIP OPTIONS and it is already proving tricky. Any help or comments appreciated. I appreciate this might not be the best solution however we have a lead time with providers to make changes which are proving quite slow and impractical so just trying to work a solution. Thanks Jon

2 1

integration kamailio with Freeswitch for video conferencing
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 and freeswitch 1.8 on debian 9 i want to configure kamailio with freeswitch for video conformance plz provide any reference guide plz help Thanks -- *Regards:* Gaurav Kumar

1 0

video chat with kamailio
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i m working on video chat app so how can i use kamailio for video chat OR i need to use other integration like freeSwitch with kamailio ? Plz help -- *Regards:* Gaurav Kumar

2 1

Support for TLS v1.3 (RFC 8446)
by g.Helio 18 Apr '19

18 Apr '19

2 1

Via shows wrong protocol of Kamailio
by Andrew Chen 18 Apr '19

18 Apr '19

Hi all, Got another anomaly in 4.1.5. I have this block of code which does t_relay_tcp: if ($var(isTpgw) == 1) { xlog("L_INFO","[$ci][$mi][CSeq $cs] isTpgw = $var(isTpgw)"); if (!t_relay_to_tcp()) { xlog("L_INFO","[$ci][$mi][CSeq $cs] $ua Relaying [$rm] from ($si) to ($rd) via tcp....Error...\n"); t_on_failure("DISPATCHER_FAIL"); } else { xlog("L_INFO","[$ci][$mi][CSeq $cs] $ua Relaying [$rm] from ($si) to ($rd) via tcp\n"); } } But when I looked at my sip capture tool, it showed the Via marked by the Kam to use UDP: INVITE sip:######@<some domain>:5060 SIP/2.0 Record-Route: <sip:206.81.191.30;r2=on;lr=on> Record-Route: <sip:206.81.191.30;transport=tcp;r2=on;lr=on> Call-ID: DL9dc0c8dd93-1079779342(a)gs-3924.GSLAB.COM To: "sip:6204857@sjomaintpsg50.fuzemeeting.com" <sip:6204857@sjomaintpsg50.fuzemeeting.com> From: "LifesizeSIP" <sip:LifesizeSIP@gs-3924.GSLAB.COM:5060>;tag=DL325a9e10c7;epid=062EE468 CSeq: 1 INVITE Max-Forwards: 69 Via: SIP/2.0/UDP 206.81.191.30;branch=z9hG4bKb16b.bd6a0791c798e3c859f8608993d584ae.1;i=5611 Via: SIP/2.0/TCP 10.101.199.105:5060;rport=53272;branch=z9hG4bK-fc8090ecf8-DL 206.81.191.30 is the Kam. I also log the activity as seen in the code with those xlog: Apr 11 19:38:37 sjomaintpsg50 /usr/local/sbin/kamailio[16348]: INFO: <script>: [DL9dc0c8dd93-1079779342(a)gs-3924.GSLAB.COM][42][CSeq 1] isTpgw = 1 Apr 11 19:38:37 sjomaintpsg50 /usr/local/sbin/kamailio[16348]: INFO: <script>: [DL9dc0c8dd93-1079779342(a)gs-3924.GSLAB.COM][42][CSeq 1] LifeSize Softphone 8.1.12 (Windows) Relaying [INVITE] from (10.101.19 9.105) to (206.81.191.72) via tcp So why did the Kamailio mark the Via header on the outgoing INVITE UDP? -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 1

"481 Dialog/Transaction Does Not Exist" after sending SUBSCRIBE with Expires: 0
by David Dean 18 Apr '19

18 Apr '19

When a client sends a SUBSCRIBE with Expires: 0, Kamailio responds correctly with a "202 OK". But then Kamailio sends a NOTIFY to the same client with "Subscription-State: terminated;reason=timeout". The client rejects this NOTIFY with a "481 Dialog/Transaction Does Not Exist". Why does Kamailio send this NOTIFY, and is there any way to fix or disable it because it's always rejected by the client? The client is IMSDroid, and Kamailio is 5.2.2 with a default configuration. Packet 1 = Client sending SUBSCRIBE with Expires: 0 SUBSCRIBE sip:X.X.X.X:443;transport=tcp SIP/2.0Via: SIP/2.0/TCP 192.168.0.21:45336;branch=z9hG4bK-670576127From: <sip:username@domain.com>;tag=1372208921To: <sip:kamailio.domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6Contact: <sip:username@192.168.0.21:45336;transport=tcp>Call-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0CSeq: 1321654295 SUBSCRIBEExpires: 0Content-Length: 299Max-Forwards: 70Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERAccept: application/pidf+xmlEvent: presence Packet 2 = Kamailio responding with 202 OK SIP/2.0 202 OKVia: SIP/2.0/TCP 192.168.0.21:45336;branch=z9hG4bK-670576127From: <sip:username@domain.com>;tag=1372208921To: <sip:kamailio@domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6Call-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0CSeq: 1321654295 SUBSCRIBEExpires: 0Contact: <sip:X.X.X.X:443;transport=tcp>Server: kamailio (5.2.2 (x86_64/linux))Content-Length: 0 Packet 3 = Kamailio sending a NOTIFY NOTIFY sip:username@192.168.0.21:45336;transport=tcp SIP/2.0Via: SIP/2.0/TCP X.X.X.X:443;branch=z9hG4bKefe7.5f8ec435000000000000000000000000.0To: <sip:username@domain.com>;tag=1372208921From: <sip:kamailio@domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6CSeq: 3 NOTIFYCall-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0Content-Length: 0Max-Forwards: 70Event: presenceContact: <sip:X.X.X.X:443;transport=tcp>Subscription-State: terminated;reason=timeout Packet 4 = Client responding to the NOTIFY with "481 Dialog/Transaction Does Not Exist" SIP/2.0 481 Dialog/Transaction Does Not ExistVia: SIP/2.0/TCP X.X.X.X:443;branch=z9hG4bKefe7.5f8ec435000000000000000000000000.0From: <sip:kamailio@domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6To: <sip:username@domain.com>;tag=1372208921Call-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0CSeq: 3 NOTIFYContent-Length: 0 There are two entries in the Kamailio log: INFO: {1 1321654295 SUBSCRIBE e8cffa8f-948d-eb99-69ca-c9f18bdf11a0} presence [notify.c:1614]: send_notify_request(): NOTIFY sip:username@domain.com via sip:Y.Y.Y.Y:45336;transport=tcp on behalf of sip:kamailio.domain.com for event presence : e8cffa8f-948d-eb99-69ca-c9f18bdf11a0 ERROR: {2 3 NOTIFY e8cffa8f-948d-eb99-69ca-c9f18bdf11a0} presence [subscribe.c:497]: delete_subs(): Failed to delete subscription from memory [slot: 174 ev: sip:kamailio.domain.com pu: presence ci: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0 ft: 1372208921 tt: c4eb72dabf44e7588b972bd3b454ea11-0cd6] These are the presence settings in the Kamailio config file: #!ifdef WITH_PRESENCE# ----- presence params -----modparam("presence", "db_url", DBURL)modparam("presence", "send_fast_notify", 0) # ----- presence_xml params -----modparam("presence_xml", "db_url", DBURL)modparam("presence_xml", "force_active", 1)modparam("presence_xml", "force_dummy_presence", 0)#!endif Appreciate your help.

2 1

Rewrite private IP in SDP
by Bonjour Madame 18 Apr '19

18 Apr '19

Hello all, I am dealing with a no audio issue, phones are not even ringing. There is a private IP address in the "VIA" section of the message header of the INVITE: Via: SIP/2.0/UDP 192.168.1.25:65291 ;rport;branch=z9hG4bKPj9e3849d1981a4094a8323dad5826ec34 and in the contact too: Contact: <sip:5002@192.168.1.25:65291;ob> In the "status 100 Attempting to connect your call" from the server, I can see the public IP of the phone under received: Via: SIP/2.0/UDP 192.168.1.25:65291 ;rport=53990;branch=z9hG4bKPj9e3849d1981a4094a8323dad5826ec34;received=PHONE_PUBLIC_IP And in the "status 200 ok" from the server, I have: Via: SIP/2.0/UDP 192.168.1.25:65291 ;received=PHONE_PUBLIC_IP;rport=53990;branch=z9hG4bKPj9e3849d1981a4094a8323dad5826ec34 Record-Route: <sip:SERVER_PUBLIC_IP;lr=on;ftag=9284528256a54f8b81e692133a15684c> I don't know what steps to take to resolve this issue.

3 3

Record route dynamic IP
by eyas barhouk 17 Apr '19

17 Apr '19

hello dears i'm trying to build kamailio & asterisk with the following : * Kamailio will listen to All the interfaces : listen=udp:0.0.0.0:5050 listen=udp:0.0.0.0:5060 * it will add the record header as in the following part : if (!isflagset(FLT_RR)) record_route(); setflag(FLT_RR); when the sip message get from kamailio to asterisk the record route header will be like "<sip:0.0.0.0:5060;....etc.>" So is there any way to make the Via and record route headers has a specific IP instead of (0.0.0.0) i mean Something like "record_route("$Ri")". thanks & regards

2 1

How can i change kamailio SIP port form 5060 to any other
by Gaurav Bmotra 17 Apr '19

17 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i want to change kamailio SIP port 5060 to other port for security purpose plz help me -- *Regards:* Gaurav Kumar

2 3

TCP connection lifetime
by Aymeric Moizard 17 Apr '19

17 Apr '19

Hi Again, Here is an issue with TCP connection being kept for more: Yesterday, I have discovered that a User-Agent (<Avaya IP Phone 1120E (SIP1120e.04.04.30.00)> tried to register a lot. It was sending REGISTER over new established TCP socket *every 2 seconds*. All the REGISTER was rejected with 401. (may be the device was misconfigured? or not receiving any of my answer? I can't tell) NOTE: You can see the expires header was very large: 86400, ie: 24 hours... I was checking the TCP/TLS connections on my server and discovered more than 1000 TCP established connection to that user/ip, and thus, I have tried to understand what happened. Checking the logs, I received 4855 REGISTER from this device from "Mar 25 03:47:09" to "Mar 25 07:56:13" which is a rate of approx one new TCP connection every 2.5 seconds... Today, I decided to check it again around 11am. jack@sip:~$ sudo kamctl stats tcp { "jsonrpc": "2.0", "result": [ "tcp:con_reset = 1857", "tcp:con_timeout = 35927", "tcp:connect_failed = 25", "tcp:connect_success = 2", "tcp:current_opened_connections = 2291", "tcp:current_write_queue_size = 0", "tcp:established = 80778", "tcp:local_reject = 0", "tcp:passive_open = 80776", "tcp:send_timeout = 2", "tcp:sendq_full = 0" ], "id": 7305 } There was still A LOT of established connections. And the connections have been established more than 24 hours ago. At 11H16: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 1161 At 11H22: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 1018 At 11H35: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 655 At 13H $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 0 So the established connections are all gone now. Between 11h16 and 11H35, I was seeing the server regularly sending [FIN, ACK] over each TCP established connection, with retransmissions for all of them. (no incoming trafic) I do not have numbers/capture/stats, but I think that kamailio was already closing some connection yesterday. I don't know when kamailio started to try closing those connections. I'm now back with this status: At 13pm: jack@sip:~$ sudo kamctl stats tcp { "jsonrpc": "2.0", "result": [ "tcp:con_reset = 1896", "tcp:con_timeout = 38042", "tcp:connect_failed = 26", "tcp:connect_success = 2", "tcp:current_opened_connections = 939", "tcp:current_write_queue_size = 0", "tcp:established = 81950", "tcp:local_reject = 0", "tcp:passive_open = 81948", "tcp:send_timeout = 2", "tcp:sendq_full = 0" ], "id": 12734 } With around 155 registration entries using TCP and TLS in my location database. As you can see, tcp:current_opened_connections = 939 is still pretty high compared to my currently registred users. I have "modparam("registrar", "max_expires", 86400)", because I'm keeping contact entries (even with TCP connection down) for push notifications. I have "tcp_connection_lifetime=3600" configured. Question 1 With "tcp_connection_lifetime=3600", I would expect kamailio to close the established connection after 3600 seconds without traffic. It is pretty obvious that no data has been exchanged over the 4855 established connection during a day. Despite the issue with the Avaya phones is solved automatically after a day, I guess similar stuff or happening, at a different rate, for other users as well. (because current_opened_connections is way higher than registred TCP/TLS users) Question 2 I can list TLS connection with "kamctl rpc tls.list" Can I get a similar list for TCP? (lsof returns a lot of duplicates...) Tks Aymeric -- Antisip - http://www.antisip.com

2 4

Route header
by Duarte Rocha 17 Apr '19

17 Apr '19

Greetings, Is there a difference for Kamailio between these 2 situations. 1 - Method with 2 Route Headers, each one with one address and one uri 2 - Method with 1 Route Header that includes 2 Route URI Best Regards, Duarte Rocha

1 0

allow calls only from registered users
by Gaurav Bmotra 17 Apr '19

17 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i wan to implement user based calling on kamailio , that only can call how registered with user plz help -- *Regards:* Gaurav Kumar

2 1

Send 200 OK response with body to INVITE requests
by David Dean 16 Apr '19

16 Apr '19

Hi all, I'm building an automated load testing system for an rtp media server with TURN that uses Kamailio for SIP. When a client sends an INVITE to Kamailio, I need Kamailio to respond with "200 OK" and the SDP generated by rtpengine. I've enabled auto-answer in the client, so as soon as it receives the response from Kamailio it will start sending RTP packets. Right now I'm using send_reply("200", "OK"); which is sending the right 200 response back to the client, but it doesn't include any body. How can I get Kamailio to send a "200 OK" and add the rtpengine_answer into the body? Appreciate any help, I've been struggling with this for three days. Cheers. rtpengine_offer("SIP-source-address ICE=force-relay RTP");send_reply("200", "OK");

4 7

Getting a new RTP Port on reINVITE during answered call
by Sergio Charrua 16 Apr '19

16 Apr '19

Hi all, I've the following infrastructure (fake IPs) 62.128.128.68 (Telco SBC Inbound) | | 10.19.139.66 (Kamailio + RTPEngine) | | 10.19.139.69 (Asterisk 13) | | 62.128.128.68 (Telco SBC Outbound) Test scenario is: - a mobile calls a destination; the call comes in from Telco SBC Inbound, goes through Kamailio+RTPengine and then to Asterisk which redirects the call to Telco SBC Outbound. - Asterisk box is set with canreinvite=no and nat=force_rport,comedia so it stays in the path when a call comes in SBC Inbound, it is correctly redireted to SBC Outbound. The call is answered, and after 5 or so seconds, SBC Inbound sends a new INVITE to force codecs, which Asterisk replies correctly. the problem is that between Kamailio and SBC Inbund, when Asterisk replies to the second invite, the media port changes, resulting in audio loss: during the call, audio from destination to originator works OK but originator to destination is mute. According to the telco's guys,this is due to the fact that Kamailio is changing during call the media port from original media port to a new one, on reply to the second Invite, and this is wrong and reason why sound from the originator to destination is lost (but not from destination to originator) I'm lost too here..... Any clue? Can anyone help? Resuming I need to make sure that, if during a call some INVITES are sent from SBC to Asterisk, that the reply keeps the same IP_ADDR:port from beginning to the end of the call. As it is today, the second invite changes te RTP Media port and no audi is heard.... If needed I can send PCap files. please find below my kamailio.cfg file #!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* sets the dispatcher list file path */ #!define DS_LIST "/etc/kamailio/dispatcher.list" /* sets the RTP Engine address and port; RTP Engine is a NAT enabled RTP Proxy */ #!define RTP_ENGINE_ADDR "udp:127.0.0.1:60000" // was udp:localhost:60000 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@10.19.139.110/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ //listen=udp:10.19.139.66:5060 listen=udp:eth1:5060 sip_warning=no; ####### Modules Section ######## #set module path mpath="/usr/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.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 "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "dispatcher.so" loadmodule "rtpengine.so" loadmodule "nathelper.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- 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) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- 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 we 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 */ 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") //;calltype=$avp(calltype)") # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- modparam("dispatcher", "list_file", DS_LIST) modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("path", "use_received", 1) # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", RTP_ENGINE_ADDR) modparam("nathelper", "received_avp", "$avp(s:rcv)") /************************/ # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) /***********************/ modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:**********@*************/kamailio") ####### Routing Logic ######## # main request routing logic route { xlog("L_INFO","$rm from $si"); xlog("L_DBG","[$fU@$si:$sp]{$rm} {$ru} "); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) // && $fU=="ping" )// && $si=="62.28.173.10" && $fU=="ping")// && strempty((a)ruri.user) && (uri == myself)) { sl_send_reply("200","OK"); exit; } # per request initial checks route(REQINIT); rtpengine_manage(); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO","Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ xlog("L_INFO","Recording Route"); record_route(); if (is_method("INVITE")) { if (has_body("application/sdp")) { if (rtpengine_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } } if (is_method("ACK") && has_body("application/sdp")) rtpengine_answer(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } //route(SIPOUT); xlog("L_INFO","Setting PRESENCE"); # handle presence related requests route(PRESENCE); # handle registrations xlog("L_INFO","Handling REGISTRAR"); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); xlog("L_INFO","**********END*************"); } onreply_route[1] { if (has_body("application/sdp")) rtpengine_answer(); } onreply_route[2] { if (has_body("application/sdp")) rtpengine_offer(); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO","REQINIT Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } xlog("L_INFO","REQINIT Finishing"); } /***********************/ # Caller NAT detection route[NATDETECT] { xlog("L_INFO","Entering NATDTECT"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO","Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO","Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO","NAT Detect set FLT_NTS = " + FLT_NATS); xlog("L_INFO","Finishing NATDETECT"); return; } /***********************/ # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Entering WITHDLG"); if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO","Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("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 route(RELAY); exit; } else { # ACK without matching transaction. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO","Finishing WITHINDLG"); exit; } # Routing to foreign domains route[SIPOUT] { xlog("L_INFO","Entering SIPOUT"); if (uri==myself) { xlog("L_INFO","URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO","Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO","******** RELAY *******"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO","branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_INFO","onreply_route NOT SET!"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_INFO","failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO","t_relay returns FALSE"); sl_reply_error(); } exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO","Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO","Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO","Entering NATMANAGE"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xlog("L_INFO","nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO","NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } xlog("L_INFO","Starting RTPEngine"); rtpengine_manage(); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO","adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { xlog("L_INFO","Entering DISPATCH"); # hash over callerid dispatching on gateways group '1' if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #we mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } route[CDRS]{ sql_query("ca","call kamailio_cdrs();","rb"); sql_query("ca","call kamailio_rating('default');","rb"); } Cheers, *Sérgio *

1 0

new RTP Port after INVITE during call
by Sergio Charrua 16 Apr '19

16 Apr '19

Hi all, I've the following infrastructure 62.128.128.68 (Telco SBC Inbound) | | 10.19.139.66 (Kamailio + RTPEngine) | | 10.19.139.69 (Asterisk 13) | | 62.128.128.68 (Telco SBC Outbound) Test scenario is: - a mobile calls a destination; the call comes in from Telco SBC Inbound, goes through Kamailio+RTPengine and then to Asterisk which redirects the call to Telco SBC Outbound. - Asterisk box is set with canreinvite=no and nat=force_rport,comedia so it stays in the path when a call comes in SBC Inbound, it is correctly redireted to SBC Outbound. The call is answered, and after 5 or so seconds, SBC Inbound sends a new INVITE to force codecs, which Asterisk replies correctly. the problem is that between Kamailio and SBC Inbund, when Asterisk replies to the second invite, the media port changes, resulting in audio loss: during the call, audio from destination to originator works OK but originator to destination is mute. According to the telco's guys,this is due to the fact that Kamailio is changing during call the media port from original media port to a new one, on reply to the second Invite, and this is wrong and reason why sound from the originator to destination is lost (but not from destination to originator) I'm lost too here..... Any clue? Can anyone help? Resuming I need to make sure that, if during a call some INVITES are sent from SBC to Asterisk, that the reply keeps the same IP_ADDR:port from beginning to the end of the call. As it is today, the second invite changes te RTP Media port and no audi is heard.... If needed I can send PCap files. please find below my kamailio.cfg file #!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* sets the dispatcher list file path */ #!define DS_LIST "/etc/kamailio/dispatcher.list" /* sets the RTP Engine address and port; RTP Engine is a NAT enabled RTP Proxy */ #!define RTP_ENGINE_ADDR "udp:127.0.0.1:60000" // was udp:localhost:60000 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@10.19.139.110/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ //listen=udp:10.19.139.66:5060 listen=udp:eth1:5060 sip_warning=no; ####### Modules Section ######## #set module path mpath="/usr/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.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 "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "dispatcher.so" loadmodule "rtpengine.so" loadmodule "nathelper.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- 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) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- 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 we 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 */ 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") //;calltype=$avp(calltype)") # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- modparam("dispatcher", "list_file", DS_LIST) modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("path", "use_received", 1) # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", RTP_ENGINE_ADDR) modparam("nathelper", "received_avp", "$avp(s:rcv)") /************************/ # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) /***********************/ modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:**********@*************/kamailio") ####### Routing Logic ######## # main request routing logic route { xlog("L_INFO","$rm from $si"); xlog("L_DBG","[$fU@$si:$sp]{$rm} {$ru} "); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) // && $fU=="ping" )// && $si=="62.28.173.10" && $fU=="ping")// && strempty((a)ruri.user) && (uri == myself)) { sl_send_reply("200","OK"); exit; } # per request initial checks route(REQINIT); rtpengine_manage(); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO","Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ xlog("L_INFO","Recording Route"); record_route(); if (is_method("INVITE")) { if (has_body("application/sdp")) { if (rtpengine_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } } if (is_method("ACK") && has_body("application/sdp")) rtpengine_answer(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } //route(SIPOUT); xlog("L_INFO","Setting PRESENCE"); # handle presence related requests route(PRESENCE); # handle registrations xlog("L_INFO","Handling REGISTRAR"); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN //route(PSTN); # dispatch destinations route(DISPATCH); xlog("L_INFO","**********END*************"); } onreply_route[1] { if (has_body("application/sdp")) rtpengine_answer(); } onreply_route[2] { if (has_body("application/sdp")) rtpengine_offer(); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO","REQINIT Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } xlog("L_INFO","REQINIT Finishing"); } /***********************/ # Caller NAT detection route[NATDETECT] { xlog("L_INFO","Entering NATDTECT"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO","Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO","Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO","NAT Detect set FLT_NTS = " + FLT_NATS); xlog("L_INFO","Finishing NATDETECT"); return; } /***********************/ # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Entering WITHDLG"); if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO","Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("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 route(RELAY); exit; } else { # ACK without matching transaction. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO","Finishing WITHINDLG"); exit; } # Routing to foreign domains route[SIPOUT] { xlog("L_INFO","Entering SIPOUT"); if (uri==myself) { xlog("L_INFO","URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO","Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO","******** RELAY *******"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO","branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_INFO","onreply_route NOT SET!"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_INFO","failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO","t_relay returns FALSE"); sl_reply_error(); } exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO","Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO","Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO","Entering NATMANAGE"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xlog("L_INFO","nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO","NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } xlog("L_INFO","Starting RTPEngine"); rtpengine_manage(); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO","adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN xlog("L_INFO","Entering PSTN"); # route to PSTN dialed numbers starting with '+' or '0000' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) { xlog("L_INFO","$rU is not outbound call to PSTN"); return; } if (!starts_with("$si","10.19.139")) { xlog("L_INFO","$rU is not comming from any Media GW"); return; } # only local users allowed to call if(from_uri!=myself && !ds_is_from_list("1") ) { xlog("L_INFO", "from_uri $fu != myself, so this is Not Allowed!!"); sl_send_reply("403", "Not Allowed"); exit; } xlog("L_INFO","$si request"); xlog("L_INFO","$rU is an outbound call to PSTN"); $ru = "sip:" + $rU + "@62.28.173.10"; // + $sel(cfg_get.pstn.gw_ip); route(RELAY); /* if(!ds_select_dst("20", "4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); */ exit; #!endif xlog("L_INFO"," NO PSTN SET!!"); return; } # Dispatch requests route[DISPATCH] { xlog("L_INFO","Entering DISPATCH"); # hash over callerid dispatching on gateways group '1' if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Dispatch requests route[DISPATCH_TO_PSTN] { # round robin dispatching on gateways group '20' - PSTN if(!ds_select_dst("20", "10")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #we mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } route[CDRS]{ sql_query("ca","call kamailio_cdrs();","rb"); sql_query("ca","call kamailio_rating('default');","rb"); } Cheers, *Sérgio *

1 0

single Contact per AOR dilema
by Sergiu Pojoga 16 Apr '19

16 Apr '19

Hi ppl, Say we want to allow 1 contact per AOR. modparam("registrar", "max_contacts", 1) Say some UA looses connection before de-registering, then comes back shortly after (before Contact expires) and tries to register with a different Contact, but gets rejected with error "test_max_contacts(): too many contacts for AOR". What's the approach to handle this situation? I'd like the registrar to accept and save that AoR based on the latest registration. Similar to how Asterisk's single-contact stack `chan_sip` works - whoever was the latest to register successfully gets to be the winner. Thanks, --Sergiu

2 2

dlg_onroute(): failed to update dialog lifetime
by Olli Attila 15 Apr '19

15 Apr '19

Hello, I just activated snmpstats module on Kamailio 5.0.7 and added the following line to route(INITIAL) at the bottom of the route like this (For noc group to monitor Kamailio dialogs): # Per SIP request initial checks route[INITIAL] { ... dlg_manage(); } Now I started to gets this error few times in an hour: /usr/sbin/kamailio[22194]: CRITICAL: dialog [dlg_timer.c:200]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7f798ca51b98 tl->next=(nil) tl->prev=(nil) Apr 15 09:54:05 /usr/sbin/kamailio[22194]: ERROR: dialog [dlg_handlers.c:1439]: dlg_onroute(): failed to update dialog lifetime I see no call failures because of this but still interested what is happening here? --Olli Attila

1 0

one way Audio issue
by Gaurav Bmotra 12 Apr '19

12 Apr '19

i have installed kamailio 5.1.x on Ubuntu server 18.4 issue is i am able to connect the call but there is one way audio not both way as i have install rtpproxy also with following config /etc/default/rtpproxy EXTRA_OPTS="-F -s udp:127.0.0.1:7722 -l PUBLIC_IP -d DBUG:LOG_LOCAL0" /etc/kamailio/kamailio.cnf modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") after restart kamailio i got this logs dev /usr/sbin/kamailio[6926]: INFO: rtpproxy [rtpproxy.c:1559]: rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled during calling i got this logs Apr 12 16:05:42 ip-172-31-38-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: rtpproxy [rtpproxy.c:2588]: force_rtp_proxy(): incorre Apr 12 16:05:42 ip-172-31-38-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): dr Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error whi Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: acc [acc.c:477]: acc_db_request(): failed to insert in Apr 12 16:05:43 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13475]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: rtpproxy [rtpproxy.c:2588]: force_rtp_proxy(): incorre Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction ans Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): drive Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: acc [acc.c:477]: acc_db Thanks Gaurav *Regards:* Gaurav Kumar

1 0

Re: [SR-Users] [sr-dev] Audio missing during call
by Mikko Lehto 12 Apr '19

12 Apr '19

Gaurav Bmotra <saigauravmehra91(a)gmail.com> wrote: > HI i have install Kamailio 5.1.2 on Ubuntu 18.4 > there is a audio issue , when we call call from one extension to another ,, > in bigning call goes good , after 9-10 sec it miss audio for one side (one > side can hear other side can't hear ) Hi, mailing list sr-dev is for Kamailio development related topics, please subscribe to sr-users for more general queries. Before posting, make sure you have described in detail your environment and also troubleshooting steps you have done so far: - what modifications you made to default configuration - what happens during SIP transactions in fail scenario - are you using RTP relay or not (rtpproxy/rtpengine) - what have you observed from packet captures Good luck -- Mikko

2 1

command out of sync with db_cluster and db_mysql
by Simone Lazzaris 12 Apr '19

12 Apr '19

Hi everybody; I'm trying to use db_cluster to connect my kamailio installation to a mariadb galera cluster, and it's driving me crazy. In my lab it works without issues, but when I try the setup in the production node, I get some strange errors: Apr 11 14:21:31 ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): driver error on query: Commands out of sync; you can't run this command now (2014) Apr 11 14:21:31 ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query This is my configuration: modparam("db_cluster", "connection","con1=>mysql://openser:PASSWD@X.X.X.30/ openser") modparam("db_cluster", "connection","con2=>mysql://openser:PASSWD@X.X.X.31/ openser") modparam("db_cluster", "connection","con3=>mysql://openser:PASSWD@X.X.X.32/ openser") modparam("db_cluster", "cluster", "cls1=>con1=9r8r;con2=9r8r;con3=9r8r") #!define DBURL "cluster://cls1" #!define SQLOPSURL "database=>cluster://cls1" modparam("dispatcher", "db_url", DBURL) modparam("acc", "db_url", DBURL) modparam("usrloc", "db_url", DBURL) modparam("auth_db", "db_url", DBURL) modparam("permissions", "db_url", DBURL) modparam("alias_db", "db_url", DBURL) modparam( "sqlops", "sqlcon", SQLOPSURL ) I'm using version 5.1.4 of kamailio, freshly compiled from source on a debian 9. Any ideas? *Simone Lazzaris* *Qcom S.p.A.* simone.lazzaris(a)qcom.it[1] | www.qcom.it[2] * LinkedIn[3]* | *Facebook[4]* [5] -------- [1] mailto:simone.lazzaris@qcom.it [2] https://www.qcom.it [3] https://www.linkedin.com/company/qcom-spa [4] http://www.facebook.com/qcomspa [5] https://www.qcom.it/includes/email-banner.gif

2 2

kamailio, asterisk and BLF seems its only using json?
by PICCORO McKAY Lenz 12 Apr '19

12 Apr '19

hi, i try to follow the how to in the wiki https://kb.asipto.com/kamailio:presence:k43-blf and configure my two only phones.. but for any states only one led its blinking, the busy state.. no other state work, (if i connect directly to asterisk all work perfeclty, i using asterisk 13 with kamailio 4.4.4) also read the archived https://lists.kamailio.org/pipermail/sr-users/2013-May/078234.html but no luck wht happened with that how to? it's different for asterisk if are integrated? do i need a pjsip asterisk event the oldes chan? see that in presense module parameters, the server address said that value of Contact header filed for 200 OK replies to SUBSCRIBE and PUBLISH and in NOTIFY messages umm if i have integrated and working asterisk (relatime mode) with kamailio, that address muswt be asterisk? taking in cosideration that i have a route rule said that send auth to asterisk and trust any incomming from it? -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

1 0

having issue updating $td
by Andrew Chen 11 Apr '19

11 Apr '19

Hey guys, Question. I have this block of code at the beginning of route block: if(is_ip($td)) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] IP detected in To domain. Resolving. \n"); exec_avp("host $td | awk '{print substr($$5, 1, length($$5)-1)}'", "$avp(s:newtd)"); xlog("L_INFO", "[$ci][$mi][CSeq $cs] new newtd = $avp(s:newtd)"); $td = $avp(s:newtd); xlog("L_INFO", "[$ci][$mi][CSeq $cs] new td = $td"); } but $td never updates. Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] IP detected in To domain. Resolving. Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] new newtd = sjomaintpsg50.fuzemeeting.com Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] new td = 206.81.191.30 Is there a reason why? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 3

Kamailio v5.1.8 Released
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, Kamailio SIP Server v5.1.8 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.8 or to v5.2.x series. For more details about version 5.1.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-1-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.2 being released out of it. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

Kamailio v5.1.8 Released
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, Kamailio SIP Server v5.1.8 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.8 or to v5.2.x series. For more details about version 5.1.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-1-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.2 being released out of it. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

Releasing Kamailio v5.1.8
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, I am considering to release v5.1.8, an minor version to be built from branch 5.1, on Thursday, this week (April 11, 2019). As usual, check that at least the patches for issues you reported were backported in order to be sure they were not missed during the typical backporting done by developers so far. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 1

Using AWS boto3 with KEMI Python routing scripts
by Michael Loughrey 10 Apr '19

10 Apr '19

Hi All, looking for some advice regarding the proper way to initialise an AWS API boto3 client object for send SNS messages within a KEMI Python 2.7 routing script. I'm sure process forking has a major impact on how this works - from kamailio.cfg fork=yes children=4 - but would greatly appreciate some guidance. I have tried various methods to allocate an client object self.sns_client = boto3.client('sns', region_name=MY_AWS_REGION) 1. within __init__(), only once from module initialisation 2. on demand within a function call by any client code e.g. called from ksr_route_request(), each creating it's own client object and so far all have resulted in intermittent crashes within botocore/client.py ultimately crashing thus : Core was generated by `/usr/local/sbin/kamailio -P /usr/local/kamailio/run/kamailio.pid -f /usr/local/'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f8f9734f754 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 (gdb) bt #0 0x00007f8f9734f754 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #1 0x00007f8f9734f82e in X509_VERIFY_PARAM_free () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #2 0x00007f8f97642f5c in SSL_free () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #3 0x00007f8f86b09c64 in PySSL_dealloc.lto_priv.5 () at ./Modules/_ssl.c:1598 #4 0x00007f8f99cbb007 in insertdict_by_entry (mp=0x7f8f841f1e88, key='_sslobj', hash=<optimized out>, ep=<optimized out>, value=<optimized out>) at ../Objects/dictobject.c:519 #5 0x00007f8f99cbe2cf in insertdict (value=None, hash=1051385741686792393, key='_sslobj', mp=0x7f8f841f1e88) at ../Objects/dictobject.c:556 #6 dict_set_item_by_hash_or_entry (value=None, ep=0x0, hash=1051385741686792393, key='_sslobj', op={'server_hostname': u'sns.us-east-1.amazonaws.com', '_connected': True, '_context': <SSLContext at remote 0x7f8f84232398>, 'server_side': False, '_makefile_refs': 0, '_closed': False, '_sslobj': None, 'do_handshake_on_connect': True, 'suppress_ragged_eofs': True}) at ../Objects/dictobject.c:795 #7 PyDict_SetItem (op=<optimized out>, key=<optimized out>, value=<optimized out>) at ../Objects/dictobject.c:848 #8 0x00007f8f99becec1 in _PyObject_GenericSetAttrWithDict (obj=<optimized out>, name='_sslobj', value=None, dict={'server_hostname': u'sns.us-east-1.amazonaws.com', '_connected': True, '_context': <SSLContext at remote 0x7f8f84232398>, 'server_side': False, '_makefile_refs': 0, '_closed': False, '_sslobj': None, 'do_handshake_on_connect': True, 'suppress_ragged_eofs': True}) at ../Objects/object.c:1529 #9 0x00007f8f99bed437 in PyObject_SetAttr ( v=<SSLSocket(server_hostname=u'sns.us-east-1.amazonaws.com', _connected=True, _context=<SSLContext at remote 0x7f8f84232398>, server_side=False, _makefile_refs=0, _closed=False, _sslobj=None, do_handshake_on_connect=True, suppress_ragged_eofs=True) at remote 0x7f8f841f62a8>, name=<optimized out>, value=None) at ../Objects/object.c:1247 Here's my setup host : Debian9/stretch python --version Python 2.7.13 kamailio -v version: kamailio 5.2.2 (x86_64/linux) 67f967-dirty flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, MEM_JOIN_FREE ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 67f967 -dirty compiled on 23:44:31 Apr 8 2019 with gcc 6.3.0 Cheers Mike

2 3

Rtpengine - no RTP packets going out
by Istvan Mogyorosi 10 Apr '19

10 Apr '19

Dear all, This is my first post after reading a lot in this mailing-list. I'm trying to use Kamailio 5.1 with the dispatcher module and rtpengine acting as SIP + RTP proxy. I have 6 asterisk servers in a private subnet that should talk with the peer via a single IP like this: Asterisk 1..n|---> | GW.PRIVATE.IP -o- GW.PUBLIC.IP |----> PEER.SIP.TRUNK I'm on Centos 7, with firewalld configured, iptables module is loaded and the rule is well defined. Packet forwarding is also enabled. Chain rtpengine (1 references) target prot opt source destination RTPENGINE udp -- anywhere anywhere RTPENGINE id:40 My call flow seems to be fine, Kamailio/rtpengine private IP is the outboundproxy parameter of Asterisk instances. My problem is that RTP packets are not present on the public interface, the rtpengine final log showing the 2 sessions, but I'm not sure this is what I want or simply the firewall does not let it out ? (To be more precise PEER.SIP.TRUNK is the trunk for SIP traffic, I have multiple IP addresses for media to connect to, reinvites are allowed) Closing call due to timeout Final packet stats: --- Tag 'as6d12caea', created 1:00 ago for branch '', in dialogue with 'as541b1e61' ------ Media #1 (audio over RTP/AVP) using unknown codec --------- Port GW.PRIVATE.IP:10000 <> 192.168.30.13:11152, SSRC 0, 0 p, 0 b, 0 e, 60 ts --------- Port GW.PRIVATE.IP:10001 <> 192.168.30.13:11153 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 60 ts --- Tag 'as541b1e61', created 1:00 ago for branch '', in dialogue with 'as6d12caea' ------ Media #1 (audio over RTP/AVP) using unknown codec --------- Port GW.PUBLIC.IP:10000 <> PEER.SIP.TRUNK:28216, SSRC 0, 0 p, 0 b, 0 e, 60 ts --------- Port GW.PUBLIC.IP:10001 <> PEER.SIP.TRUNK:28217 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 60 ts Best regards, Istvan

2 2

Kamailio and CDR
by Yu Boot 10 Apr '19

10 Apr '19

Is there any other way to retrieve CDRs from Kami other than dialog/acc modules and writing CDRs to syslog only? Thanks.

6 13

Help request in regards to Kamailio
by Anas ZAHHAF 10 Apr '19

10 Apr '19

Dear, I am writing to ask for help, I found your Email after I've subscribed to Kamailio.org. I am a beginner in Linux environment ( I worked on Windows for 20 years), it would be very helpful if you can help me installing Kamailio on a computer running Debian and how to use the load balancing module so that I can split the charge between 2 asterisk servers, by sharing any useful tutorial or document or even by Teamviewer if this doesn't disturb you. Please help me, all the tutorials I find on internet aren't made for beginners. I will be awaiting your reply and excuse me again if I disturb you. Best regards, -- *Anas ZAHHAF* *105, rue des Moines75017 PARIS* *+33(0)7.66.31.31.85* azahhaf(a)zaion.ai <niluse(a)zaion.ai> | www.zaion.ai ᐧ

2 1

change failure status if reason is q.850 cause=31
by Techinnovation 10 Apr '19

10 Apr '19

Hello, How can I change the status where Reason: q.850;cause=31. ? This code change status for all where we receive status 480 to 404. Is it possible to this code also to add second condition - if reason is like "q.850;cause=31" failure_route[MANAGE_FAILURE] { ... if (t_check_status("480")) { t_reply("404","Unallocated number"); exit; } Need to modify this massage. SIP/2.0 480 Temporarily Not Available. Allow: UPDATE,REFER,INFO. Call-ID: 6b11b153405a5103330697b774b97f5a@192.168.2.2:5060. Contact: <sip:192.168.3.3:5060>. CSeq: 102 INVITE. From: "380668375820" <sip:380668375820@192.168.2.2>;tag=as445242c0. Reason: q.850;cause=31. Server: Cirpack/v4.76 (gw_sip). To: <sip:9052946032@192.168.3.3:5060>;tag=00-07929-6e4a3079-75447e2a5. Content-Length: 0. Thanks. BR, Aleksandr

2 1

Question regarding retcode
by Andrew Chen 09 Apr '19

09 Apr '19

Hey all, Question regarding the type of retcode. I ran a curl command and $retcode returned 200. I check against that variable if it's 200 or not. I tried to check 200 as a string and integer but never matches so I ran these lines of code: xlog("L_INFO", "[$ci][$mi][CSeq $cs] Validating meeting ($rU). Result: $var(result) Return code: $retcode \n"); if(is_int($retcode)) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is an integer \n"); } if(typeof("$retcode", "str")) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is a string \n"); } None of these xlog outputed. How do I check for $retcode in if or switch statement? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Centos 7 opensuse repo issues for 5.2
by Jeffrey Gray 09 Apr '19

09 Apr '19

Good afternoon, We are unable to get kamailio 5.2 to install using the recommended opensuse repos for Centos 7. Is there any way to resolve the following errors: Thanks very much for your assistance! using the following repo setup in /etc/yum.repos.d/ [home_kamailio_v5.2.x-rpms] name=RPM Packages for Kamailio v5.2.x (CentOS_7) type=rpm-md baseurl=http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rp… gpgcheck=1 gpgkey=http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpm… enabled=1 Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.2-3.el7.centos will be installed --> Finished Dependency Resolution Dependencies Resolved ============================================================================================================================================================================= Package Arch Version Repository Size ============================================================================================================================================================================= Installing: kamailio x86_64 5.2.2-3.el7.centos home_kamailio_v5.2.x-rpms 5.6 M Transaction Summary ============================================================================================================================================================================= Install 1 Package Total download size: 5.6 M Installed size: 25 M Downloading packages: kamailio-5.2.2-3.el7.centos.x8 FAILED ================- ] 2.3 MB/s | 1.7 MB 00:00:01 ETA http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.2.x-rpms clean metadata Trying other mirror. Error downloading packages: kamailio-5.2.2-3.el7.centos.x86_64: [Errno 256] No more mirrors to try. If we run the suggested clean metadata command, then the error changes to: sudo yum -y install kamailio Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirrors.advancedhosters.com * epel: mirror.cogentco.com * extras: mirrors.advancedhosters.com * updates: mirrors.advancedhosters.com Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.2-3.el7.centos will be installed --> Finished Dependency Resolution Dependencies Resolved ============================================================================================================================================================================= Package Arch Version Repository Size ============================================================================================================================================================================= Installing: kamailio x86_64 5.2.2-3.el7.centos home_kamailio_v5.2.x-rpms 5.6 M Transaction Summary ============================================================================================================================================================================= Install 1 Package Total download size: 5.6 M Installed size: 25 M Downloading packages: Delta RPMs disabled because /usr/bin/applydeltarpm not installed. kamailio-5.2.2-3.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not Satisfiable Trying other mirror. Error downloading packages: kamailio-5.2.2-3.el7.centos.x86_64: [Errno 256] No more mirrors to try.

3 3

Upstream ignores reply, because To-tag doesn't match?
by Sergiu Pojoga 09 Apr '19

09 Apr '19

Hi ppl, Scenario: invite from upstream is t_relayed to a client gateway. After `fr_inv_timeout` occurs, in a failure route I simply t_reply with "503 - Service unavailable", then exit(). However, the upstream provider simply ignores this reply, call doesn't hang up. This doesn't happen when the client's gateway generates identical 503 or other negative replies. I suspect this is happening because the To-tag in the Kam generated 503 reply doesn't match with the To-tag that previously was forwarded from the client's gateway to the upstream provider in the `180 Ringing`. Any suggestions what can be done about it? 2019/04/06 10:42:46.127177 65.39.xxx.xxx:5060 -> 208.72.xxx.xxx:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 208.72.xxx.xxx:5060;received=208.72.xxx.xxx;rport=5060;branch=z9hG4bK1982422573 Record-Route: <sip:65.39.xxx.xxx;lr=on;did=915.8af1> From: "SERGIU" <sip:514XXXXXXX@208.72.xxx.xxx>;isup-oli=00;tag=1975755942 To: <sip:514XXXXXXX@65.39.xxx.xxx>;tag=as2ab54180 Call-ID: DID-28270826(a)208.72.xxx.xx CSeq: 477023 INVITE Supported: replaces, timer, path Contact: <sip:1514XXXXXXX@65.39.xxx.xxx:5060> Content-Length: 0 2019/04/06 10:42:51.072516 65.39.xxx.xxx:5060 -> 208.72.xxx.xxx:5060 SIP/2.0 503 Service Unavailable Call-ID: DID-28270826(a)208.72.xxx.xx Via: SIP/2.0/UDP 208.72.xxx.xxx:5060;rport=5060;branch=z9hG4bK1982422573;received=208.72.xxx.xxx From: "SERGIU" <sip:514XXXXXXX@208.72.xxx.xxx>;isup-oli=00;tag=1975755942 To: <sip:514XXXXXXX@65.39.xxx.xxx>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8e82 CSeq: 477023 INVITE Server: KAM Content-Length: 0 Thanks, --Sergiu

3 8

modifying via header from public IP to private IP
by Narasimhaiah, Vinod Maralenahalli 09 Apr '19

09 Apr '19

Hi , I have a cloud server, when I make call from a sip phone (registered with kamailio), in the INVITE header I am seeing the via header with public IP, is there any way that I can modify via header to send only the private IP. The calls are landing on another sip gateway, they will allow only the private IP. Please let me know, is it possible to replace. I tried all the replace functions but it is not working. Thank you vinod.M.N -- ***This transmission contains information that is confidential and proprietary of Kirusa Inc.,its affiliates and/or its partners and customers. This information is intended to be shared only with the recipient of this e-mail who is bound by confidentiality. If you received this e-mail in error, please delete the e-mail and notify the sender by return e-mail.***

1 0

modifying via header from public IP to private IP
by vinod mn 08 Apr '19

08 Apr '19

Hi , I have a cloud server, when I make call from a sip phone (registered with kamailio), in the INVITE header I am seeing the via header with public IP, is there any way that I can modify via header to send only the private IP. The calls are landing on another sip gateway, they will allow only the private IP. Please let me know, is it possible to replace. I tried all the replace functions but it is not working. -- Thanks and regards Vinod.M.N

3 2

How to Register sip trunk in kamailio without username and password?
by Mostafa Moradi 08 Apr '19

08 Apr '19

Hi, I'm mostafa moradi from iran, I Have installed kamailio, and I have a sip trunk without username and password, I want to register the sip trunk in kamailio with UAC Module, but this module only accepts sip trunk with user and password, How Can I Register sip trunk in kamailio without username and password? Thank you in advance Best regards,

4 3

Re: [SR-Users] Timeout between 100 & 183
by Alex Balashov 05 Apr '19

05 Apr '19

On Fri, Apr 05, 2019 at 06:23:16PM -0400, Alex Balashov wrote: > route { > ... > > # Initial INVITE handling: > > if(is_method("INVITE")) { > # Set fr_inv_timer initially to 5 seconds, so that > # if we do not receive an answer or at least one non-100 1xx > # response, trigger a failure_route and try the next gateway > # or whatever. > > t_set_fr(50000); Sorry, this would have been a lot more pedagogically useful if it had read: t_set_fr(5000); -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

1 0

UAC authentication challenge response using KEMI/app_ruby
by Andrew White 05 Apr '19

05 Apr '19

Hi all, I’m continuing on my quest to use app_ruby for my logic - it’s working very well so far! The last major hurdle I’ve come across is UAC response to WWW challenge. On my outbound path, the correct URIs for From and To are set based on provider requirements, and the INVITE is sent using t_relay(): def ksr_route_to_trunk() KSR.info("Route to trunk") if KSR::TM.t_is_set("failure_route") < 0 then KSR::TM.t_on_failure("ksr_failure_manage_trunk") end # Relay KSR::PV.sets("$fu", "sip:#{$uacs[trunk]['r_username']}@#{$uacs[trunk]['r_domain']}") KSR::PV.sets("$fn", '') new_uri = "sip:61400123123@#{uacs[trunk]['r_domain']}" KSR.info("New URI is #{new_uri}") KSR.seturi(uri) KSR::UAC.uac_replace_to('', uri) KSR::TM.t_relay() exit end I’ve stripped it back, but this function is great and works perfectly. An INVITE is sent to the correct place, and the trunk receives the INVITE. It then responds with a 401/WWW-Authenticate. My failure route is called fine. Below is the stripped down code: def ksr_failure_manage_trunk() exit if KSR::TM.t_is_canceled() > 0 KSR.info("Hit trunk failure manage route") KSR::PV.sets("$avp(auser)", 'user') KSR::PV.sets("$avp(apass)", 'pass') KSR::PV.sets("$avp(arealm)", 'test.local') KSR.info("AVPs: #{KSR::PV.gete("$avp(auser)")} #{KSR::PV.gete("$avp(apass)")} #{KSR::PV.gete("$avp(arealm)")}") if KSR::UAC.uac_auth() then KSR.info("UAC authed, relaying") KSR::TM.t_relay() else KSR.info("UAC NOT authed, no relay") end exit end Also the AVP values are set (I believe) correctly: modparam("uac", "auth_username_avp", "$avp(auser)") modparam("uac", "auth_password_avp", "$avp(apass)") modparam("uac", "auth_realm_avp", "$avp(arealm)") This throws the following complaint, and doesn’t send a new response back to the trunk: Apr 5 02:55:22 ip-10-0-0-2 /usr/local/sbin/kamailio[26870]: ERROR: {1 102 INVITE 3a84da15272da66e527a79144c48516c@10.0.0.20:5060} tm [t_fwd.c:1728]: t_forward_nonack(): no branches for forwarding Apr 5 02:55:22 ip-10-0-0-2 /usr/local/sbin/kamailio[26870]: ERROR: {1 102 INVITE 3a84da15272da66e527a79144c48516c@10.0.0.20:5060} tm [tm.c:1619]: _w_t_relay_to(): t_forward_noack failed I don’t understand branches very well, however after some digging around, I appended KSR::COREX.append_branch() in before the t_relay() in the failure. This stops the error and now we send an ACK and a new INVITE, but still without the expected authorisation header. I feel like my lack of understanding about branches is the issue here - I seem to be sending an INVITE, but not adding the auth header that I suspect is being generated somewhere! Thanks for your help. ________________________________ Andrew White - Director uConnected Email: andrew(a)uconnected.com.au Web: www.uConnected.com.au

2 2

Statistics of received replies
by Duarte Rocha 04 Apr '19

04 Apr '19

Greetings, Does Kamailio have statistics like the sl.stats but for received replies instead of locally generated ones? I want to be able to look at how many "491" replies i'm receiving, for example. Best Regards, Duarte Rocha

2 1

Kamailio v5.0.8 Released
by Daniel-Constantin Mierla 04 Apr '19

04 Apr '19

Hello, Kamailio SIP Server v5.0.8 stable release is out. This is a maintenance release of the old stable branch 5.0, that includes fixes since the release of v5.0.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous v5.0.x versions are strongly recommended to be upgraded to v5.0.8 or even better directly to v5.1.x or v5.2.x because v5.0.x is getting out of the official maintenance time frame. Important note: this is the last planned release from branch5.0. For more details about version 5.0.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-0-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 5.2, at this moment with its latest release v5.2.2. See more details about it at: * https://www.kamailio.org/w/kamailio-v5-2-0-release-notes/ Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

Kamailio-error reading: Connection reset by peer
by Mangesh Kharade 04 Apr '19

04 Apr '19

Hi Team, We are facing issue with kamailio application , we are not able to communicate to kamailio application getting below error. Can you please suggestion us how to resolved this issue issue, it production servers and we did not get any clue to resolve it. Error1:- Apr 2 06:53:54 urcmcs /usr/local/sbin/kamailio[2240]: ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) Apr 2 06:53:54 urcmcs /usr/local/sbin/kamailio[2240]: ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading Apr 2 06:54:12 urcmcs /usr/local/sbin/kamailio[2244]: ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) Apr 2 06:54:12 urcmcs /usr/local/sbin/kamailio[2244]: ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading Apr 2 06:54:17 urcmcs /usr/local/sbin/kamailio[2244]: ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) Apr 2 06:54:17 urcmcs /usr/local/sbin/kamailio[2244]: ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading Error1:- Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3008 has closed Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3009 has closed Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3010 has closed Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3011 has closed Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3012 has closed Please let us know if you need more help from our end. Thanks and Regards, Mangesh Kharade Consultant - Microsoft Azure | Managed Services [cid:image003.png@01D15FFB.27B1AA80] Pune IT Park, 34 Aundh Road Bhau Patil Marg, Pune 411020 Tel: +91.20.30204000 | Extn. 9694 VOIP: +1.713.353.7464 Mobile: +91.9168181166 Mangesh.kharade(a)silicus.com<mailto:Mangesh.kharade@silicus.com>| www.silicus.com<http://www.silicus.com/> The information in this email is confidential and may be legally privileged. It is intended solely for the addressee and access to it by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken based on it, is strictly prohibited and may be unlawful.

2 2

Releasing Kamailio v5.0.8
by Daniel-Constantin Mierla 04 Apr '19

04 Apr '19

Hello, I am considering to release v5.0.8 to mark the end of official maintenance for branch 5.0 on Thursday, April 4. If no major regression will be introduced, this will be the last release from branch 5.0. Afterwards, the stable branches maintained will be 5.1 and 5.2. In case you have in mind some specific commits that should get into 5.0 branch, then reply and list them to be sure they are not missed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

2 2

Kamailio & asterisk integration
by ayoub krari 04 Apr '19

04 Apr '19

Good morning, I want to Loadbalance calls equally to asterisk servers, i have already tried with opensips but didnt work for me, i want to try kamailio now, i have tried with the tutorial of " Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database" but i got some errors and packages missing during the installation and the configuration, my Linux distro is Debian 8.7.0 and i’m using Asterisk v 13, i want really to implement Kamailio , any one can help me with steps or anything , Thanks in advance !

2 1

dispatcher "failover feature disabled"
by David Villasmil 03 Apr '19

03 Apr '19

Hello guys, In my config i have: # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") # PVs for hashing modparam("dispatcher", "hash_pvar", "$fU@$ci") # PVs to store results when calling ds_is_from_list modparam("dispatcher", "setid_pvname", "$var(setid)") modparam("dispatcher", "attrs_pvname", "$var(attrs)") # method to probe the gateways modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", " sip:dispatcher@proxy.voice-proxy.singlecommand.com") modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_ping_reply_codes","class=2") modparam("dispatcher", "ds_probing_mode", 1) # size of hash table storing data for call load dispatching, power of two modparam("dispatcher", "ds_hash_size", 10) # expiration time in seconds to remove the load on a destination if no BYE was received modparam("dispatcher", "ds_hash_expire", 3600) # expiration time in seconds to remove the load on a destination if no 200 OK for INVITE was received # and state updated with ds_load_update modparam("dispatcher", "ds_hash_initexpire", 60) modparam("dispatcher", "ds_hash_check_interval", 30) But i'm getting on startup: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled What avp am I missing? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Re: [SR-Users] Adding presentity_uri prefix for multi-tenanted presence/dialoginfo configuration
by Rhys Hanrahan 03 Apr '19

03 Apr '19

Hi All, I managed to solve my need to re-write NOTIFY packets with help from this thread: https://www.mail-archive.com/sr-users@lists.kamailio.org/msg08088.html This seems to work well. I am re-writing the packets as shown below… I guess my only question is, is this relatively safe/smart to do? I basically copied what our standalone Asterisk PBXes are doing in my captures, so I guess so? But I’m not confident enough to say it won’t cause some weird edge case issues. I can only say it works fine with the Snom and Yealink handsets I am testing with. I’m just concerned because it feels a bit like a hack. For SUBSCRIBE I re-write: * Request URI and To header from 101@ sent by the handset to tenant101@ (to match the SIP username of the handset - tenant101@) For NOTIFY I re-write: * From and Contact header to remove the tenant prefix, to go back to 101@ to match the BLF key configuration on the handsets. Any advice would be appreciated on if this is a bad idea, and there’s a better way. Thanks! Rhys Hanrahan Chief Information Officer Nexus One Pty Ltd E: support(a)nexusone.com.au<mailto:support@nexusone.com.au> P: +61 2 9191 0606 W: http://www.nexusone.com.au/ M: PO Box 127, Royal Exchange NSW 1225 A: Level 12 227 Elizabeth St, Sydney NSW 2000 From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of Rhys Hanrahan <rhys(a)nexusone.com.au> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Date: Tuesday, 26 March 2019 at 8:56 pm To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Adding presentity_uri prefix for multi-tenanted presence/dialoginfo configuration Hi Everyone, I’ve recently just started working with Kamailio – thanks everyone for this amazing software. Like many people, I’m in the process of trying to put Kamailio in-front of Asterisk to allow it to scale out, and my plan is for Kamailio to take over registrations, usrloc and presence/dialoginfo, as we’ve had issues where handsets are failing to get BLF updates/notifys, so I am hoping a couple of Kamailio boxes can scale better in this regard. DMQ is particularly exciting in how it will allow me to build a truly distributed platform. But I’m struggling to get presence to work the way I need it to. Our environment is multi-tenanted, but we do not (and can’t really) use multi-domains. Instead, we prefix the SIP usernames with the “tenant name” such as tenanta101 and tenantb101. My problem is that all the BLFs are configured for 101@PBX with no tenant name in the User part of the URI so that internal dialing and call pickup will work. Because in Asterisk we use “subscribecontext” this hasn’t been a problem in the past – Asterisk knows for subscriptions coming from that handset that it belongs to that tenant’s context in the dialplan and they are “isolated”, so having handsets subscribe as 101@ in their request URI was never a problem. Of course, with Kamailio I don’t have subscribecontext, and my main issue is that the “presentity_uri” being stored is 101@ while each of the SIP accounts of the handsets are registered as tenenata101@, and as such, no NOTIFYs are sent by Kamailio because it thinks that there are no watchers for the presentity_uri. If I change the SIP account username to 101 to match the BLF key, NOTIFYs are sent as expected. But then this breaks call pickup and internal dialing using the BLF keys. I would rather handle this in Kamailio than in Asterisk and having to re-configure the BLF keys for hundreds of handsets. I need to do something like: * When a SUBSCRIBE comes in, I need to prefix presentity_uri with the tenant name so the subscription changes from 101@ to tenanta101@ in the active_watchers table. * When Kamailio generates a NOTIFY, the notify would be built based on presentity_uri and would come out as tenanta101@, but the phone’s BLFs are configured for 101@ so I would need to *remove* the tenant prefix before sending out the notify. Is there a good, or recommended way to handle this scenario? Maybe something entirely different to changing the presentity_uri? I’ve written the following config to re-write the request URI and To field in any incoming subscribe requests, and successfully got the active_watchers table to store a presentity_uri containing my tenant prefix. But the problem I am having is that I can’t figure out how to modify the NOTIFY packets before they are sent – and looking at cfgtrace, it seems I might not be able to? My “standard” presence + dialoginfo configuration was taken from here as a starting point: https://kb.asipto.com/kamailio:presence:k43-blf Here is what I have so far in terms of trying to make my modifications – any guidance would be greatly appreciated. It feels like there’s probably a better way to do this than re-writing critical headers like To and Request URI? @@ -466,6 +467,8 @@ request_route { # authentication route(AUTH); + route(REWRITE_PRESENCE); + # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { # Asterisk is our voicemail route(TOASTERISK); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { # See REWRITE_PRESENCE - this should have been executed before we get here. handle_subscribe(); t_release(); } exit; #!endif +route[REWRITE_PRESENCE] { + if (is_method("SUBSCRIBE")) { + xlog("Re-writing subscribe to include tenant prefix\n"); + # The default presentity_uri needs to be prefixed with + # the tenant name + # So re-write the To header + route(TENANTINFO); + # Now grab the tenant name. + $var(subscribe_ru) = "sip:" + $var(tenant_name) + $rU + "@" + $rd; + xlog("Re-writing SUBSCRIBE To header as: $var(subscribe_ru)\n"); + insert_hf("To: $var(subscribe_ru)\r\n", "From"); + $ru = $var(subscribe_ru); + # Force change immediately: + # http://www.kamailio.org/wiki/tutorials/faq/main#why_changes_made_to_headers… + msg_apply_changes(); + } else if (is_method("NOTIFY")) { + xlog("Re-writing NOTIFY to remove tenant prefix\n"); # This code block does not get executed right now. + # We are storing the presentity_uri with a tenant prefix + # handsets do not expect this, only 101@, 102@ etc... In their BLF configs + # So this tenant prefix must be removed when building the reply. + route(TENANTINFO); + $var(notify_ru) = "sip:" + $(rU{s.substr,$(var(tenant_name){s.len}),0}) + "@" + $rd; + xlog("Re-writng NOTIFY to use: $var(notify_ru)\n"); + insert_hf("To: $var(notify_ru)\r\n", "From"); + $ru = $var(notify_ru); + # Force change immediately: + # http://www.kamailio.org/wiki/tutorials/faq/main#why_changes_made_to_headers… + msg_apply_changes(); + } +} Thanks for your time. Rhys Hanrahan Chief Information Officer Nexus One Pty Ltd E: support(a)nexusone.com.au<mailto:support@nexusone.com.au> P: +61 2 9191 0606 W: http://www.nexusone.com.au/ M: PO Box 127, Royal Exchange NSW 1225 A: Level 12 227 Elizabeth St, Sydney NSW 2000 [ttp://quintus.nexusone.com.au/~rhys/nexus1-email-sig.jpg]

1 0

uac_replace_from/to, msg_apply_changes and Re-INVITE
by Henning Westerholt 03 Apr '19

03 Apr '19

Hello, I am currently debugging a strange issue related to the usage of uac_replace_from/to() after msg_apply_changes(). It works all fine, but in a Re-INVITE case it inserts the wrong headers for the 100 - Trying case. The calle side gets then a reply with wrong From/To. This happens all the time with this particular cfg. I already checked for the usual issues: - msg_apply_changes() called several times - msg_apply_changes() called after transactions created/async processing etc.. Any ideas what else could be the problem? The msg_apply_changes() is needed for some SDP filtering logic in the cfg. Thanks and regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

3 8

TOPOS Call-ID masking
by Cinthia Leung 02 Apr '19

02 Apr '19

Greetings, Call-ID masking is not available, although the modparam is there. Just wondering if there are any plans to get that in soon. Thanks. Cindy

2 2

ENUM for CNAM dips
by Spencer Thomason 02 Apr '19

02 Apr '19

Hi all, I’m interested in extending the ENUM module to add support for draft-ietf-enum-cnam-08 CNAM queries using ENUM. I’m looking for high level guidance as to the best way to implement this and any feedback on naming of exported functions and parameters. Maybe something like enum_cnam_query("pvar", "destination", [,"suffix" [,"service"]]) ? Best regards, Spencer

2 1

Info: Kamailio World 2019 – Participation Grants
by Daniel-Constantin Mierla 02 Apr '19

02 Apr '19

Hello, we continue our tradition to give away 3 free passes to students and underrepresented people at Kamailio World Conference (www.kamailioworld.com) – thanks to the sponsors, we are able to offer them in 2019 as well. The passes cover the entire registration fee for all three days and the social networking events – costs with travel and accommodation have to be covered by the participants. Anyone enrolled in an academic activity (including master and PhD programs) as well as underrepresented people can apply for one of the passes via email to me or to: * conference [at] kamailioworld.com or by contacting us via web form at: * https://www.kamailioworld.com/k07/contact-us/ Applications done until April 15, 2019, will be selected by April 18, 2019. Afterwards, if there are still available passes, they will be allocated in the first come first served policy. The details for most of the presentations were published, the full agenda is expected to be ready in a matter of a few days. You can see more details at: * https://www.kamailioworld.com/k07/schedule/ Looking forward to meeting many of you in Berlin, at Kamailio World 2019! It’s going to be another great event about open source real time communications! * https://www.kamailioworld.com/k07/registration/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

Port 5060 omitted in Record-Route
by Lợi Đặng 02 Apr '19

02 Apr '19

Hi all, i was using kamailio 4.2.1 located in 2 networks > listen = tcp:MY_ADDR:5060 advertise MY_ADDR:5060 > listen = tls:MY_ADDR:5061 advertise PUBLIC_NAT_ADDR:5061 when the call made from the inside network to out side, running `record_route()` resulted in 2 Record-Route headers added (enable_double_rr=1) Record-Route: PUBLIC_NAT_ADDR:5061;transport=tls;lr > Record-Route: MY_ADDR;transport=tcp;lr That was totally fine omitting the port in the first Record-Route when using tcp (or udp) on the first realm, but when i start switching to tls, it caused trouble Record-Route: PUBLIC_NAT_ADDR:5061;transport=tls;lr > Record-Route: MY_ADDR;transport=tls;lr The client is then told to send ACK/BYE to `MY_ADDR;transport=tls` located at `MY_ADDR:5061` as per rfc3263, then the call would failed. I had another try with `record_route_preset("PUBLIC_NAT_ADDR:5061;transport=tls", "MY_ADDR:5060;transport=tls");`, it really did add what i want with explicit 5060 port on RR, `ACK/BYE` travel on the correct path, but `loose_route()` only consumes the local `Route` header (it should consume 2). So my assumption is to stick with `record_route()` function to make `loose_route()` work properly. I tried using another port on the local realm, e.g: 5062 and the port is explicitly added to the Record-Route header `MY_ADDR:5062;transport=tls;lr` So is `5060` couldn't be explicitly added to the inbound Record-Route, or i just missed something? Any help will be appreciated. P/S: I also tried 4.4.7 and it still omit my 5060 port in the RR. rgds, Loi Dang Thanh Phone : +84. 774.735.448 Email : loi.dangthanh(a)gmail.com

3 5

Compressing SIP messages
by Igor Olhovskiy 01 Apr '19

01 Apr '19

Hi! Was looking on some way to compress SIP signalling between internal servers. But found out, there are no some pre-defined or recommended mechanism of doing this. Yes, I found gzcompress module, but it will help only in case of big body, but not whole message itself. Like in some cases you can have 4-5 RR headers, which is already a lot of info. What I googled is SigComp Compact Header form As I found, SigComp is really not widely-used method and it support along opensource SIP servers/PBXes is not documented. For Compact Header form - is there any way of using Kamailio to compress message this way? Maybe also there is a way of having some intermediate proxy-layer software, that will gzip/gunzip SIP signalling in realtime? All of this actually cause info in SIP/SDP grows rapidly, unfortunately MTU size is not.

8 11

presence_dialoginfo module not sending publish on dialogue change
by Richard Nunley 01 Apr '19

01 Apr '19

To whom can be of assistance, I’m am working towards enabling presence on my Kamailio proxy but have hit a bit of a brick wall. I’m seeing dialogues being created and persisted to the database, subscriptions are being handled and rows are being written to the active_watchers table but no PUBLISH is ever being received by the server to kickoff the sending of NOTIFY to the active watchers. Further, I’m never seeing a row in the watchers table though I don’t know that this isn’t expect behavior. I’ve attached my current config for review. Any help will be greatly appreciated! Thank in advance!

3 2

Use Kamailio as forward proxy between two networks
by Lauritz 01 Apr '19

01 Apr '19

Hello, I am trying to implement the following network setup using Kamailio and RTPProxy, but was not able to find any recent documentation or blog posts with hints: > local_net1 : local_net2 > 10.250.10.* : 10.250.11.* > : > (1) (2) : > +-------------+ +--------------+ +--------------+ > ! !.11 .12 ! Proxy !.21 ! Debian Host !.22 > ! Starface !---------------! + Kamailio !<<--------------! + Linphone !.23 > ! ! ! + RTPProxy ! ! ! > +-------------+ +--------------+ +--------------+ > eth0 : eth1 > : > : > As you can see, the proxy has two interfaces and its purpose is to forward traffic from the client's network to the Starface's network. Has anyone experience to share or got any other hints where to start? Thank you very much for your help :) Kind regards, Lauritz

2 1

Kamailio Managed Cloud
by Alex Balashov 01 Apr '19

01 Apr '19

For immediate release: ATLANTA, GA (1 April 2019)--Following its long tradition of forward-thinking innovation in open-source Real-Time Communications (RTC), the leadership of the Kamailio SIP server project (formerly known as OpenSER) has launched the Kamailio Managed Cloud (KMC). Alex Balashov of Evariste Systems and Fred Posner of The Palner Group were among several key North American stakeholders on hand to discuss it further. As Posner told us, "Kamailio Managed Cloud is a new, best-of-breed managed Kamailio hosting environment. We provide elasticity by leveraging human capital. Using our patent-pending Orchestration Turk system, our managed cloud technicians will purchase your SuperMicro bare-metal server, known as a KamBox, drive to the data centre, and rack-and-stack it on demand -- all in response to a SOAP API call! And when you want to remove your on-demand KamBox, just issue a 'delete' operation and we will decommission the hardware -- live!" "Dedicated KamBox colocation really puts the Management in Managed Cloud," emphasised Balashov. "But that doesn't mean we keep the customer out of the loop. We are using the most modern XMLHttpRequest and 'AJAX' techniques to provide a real-time dashboard with KamBox status to all users, though it will not work with Internet Explorer 6 -- we're addressing that." "Anyway, there are numerous other cloud platforms out there, many of which rely on virtualisation and automated orchestration and the like. It really takes the human touch out of the equation and gives the short shrift to quality, while also taking away jobs. And besides, how can a so-called 'virtual' machine be truly carrier-grade?" Kamailio Managed Cloud launched in two principal regions today -- US-EAST-1, located in Jacksonville, Florida, USA, and EU-CENTRAL-1, based in the booming tech hub of central Europe and birthplace of OpenSER: Berlin, Germany. Unnamed sources tell us that strategic expansion is planned for several other regions, including Flint, Michigan, USA. The mechanics of Kamailio Managed Cloud have invited questions about how other elastic cloud resources are managed, more especially elastic storage. According to KMC senior management, storage is provided via a mechanism developed by ASIPTO StorageWorks. ASIPTO chief Daniel-Constantin Mierla was on hand at the KMC launch to explain: "It's very easy. When a customer requests more space with a simple and lightweight SOAP call with just a little custom XML namespaces, then depending on the region, Fred or I receive an SMS on our smartphones from our advanced PHP backend and drive to the data centre to insert another hard drive into one of the KamBox's several drive bays. We are using the latest in advanced mobile communications, SMS, to ensure fastest response time. Of course our service level is comprehensive and includes Managed Partition Table Labelling and Managed LVM Insertion. Also there is optional Managed Filesystem Formatting for a complete portfolio of managed services. ASIPTO StorageWorks moves at the speed of global e-business. Just text us your desired filesystem layout and partitioning scheme. It is really that easy!" When asked about business focus, marketing turnaround veteran Henning Westerholt, Executive Vice President of Business Development for KMC responded: "Our target customer is a socially conscious Millennial-run corporation or start-up who see value in the unique blend of managed services and an artisanal, hand-crafted, organic approach that is compatible with social justice and takes a people-centric view." Asked to elaborate upon his vision of social justice, Westerholt added: "Virtualisation, orchestration and auto-scaling take the people out of the equation, and Kamailio is, in the end, about people, as communication in general is about people. What would it be without the people? Moreover, these mechanisms take away gainful opportunities from hard-working university graduates. Kubernetes, Docker Swarm, VMWare, and KVM contravene the social policy objective of full employment." At press time, Posner, who also heads up KMC Labs North America, was said to be lending his pioneering business vision and knack for seeing around technology trend corners to intensive work on next-generation, bleeding-edge functionality for KMC. While Posner was tight-lipped about the specifics, he did state that the 'dispatcher' module is slated to be deprecated in favour of a "human load-balancing approach to SIP requests that is both cloud-native and eco-conscious". Industry analysts agree that protocol-aware load-balancing strategies are complicated and that the application of human judgment to the problem represents a fertile field for dynamic innovation and investment. Reports from regional press confirm that KMC Labs are in the process of building a fullfilment centre for this innovative offering, said to be branded "Central Switchboard", at an undisclosed site in Bangladesh, where thousands of "Load Balance Operator" jobs have been advertised in and around the Dhaka market. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

1 0

db_redis and user
by Joan Salvatella 01 Apr '19

01 Apr '19

Hello, We are working on a project that's using two active-active Kamailio 5.2 instances that redirect traffic to several Asterisk 15.4 instances. Kamailio is acting as a registrar and should store the location information of the registered terminals. The backend database is db_redis. Phone terminals get registered and the information gets looked up from the subscriber schema in redis (That's cool) but we are having problems with the locations. This is the related code: modparam("db_redis", "keys", "location=entry:ruid&usrdom:username,domain&timer:partition,keepalive") if(ds_is_from_list("1", "2", "sip:$fd:5080")) { record_route(); route(INVITE_ASTERISK); exit; } route[INVITE_ASTERISK] { xlog("L_INFO", "[R-INVITE-RESOLVER:$ci] Entering ASTERISK INVITE resolver\n"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; } route(INVITE_POSTROUTE); } route[INVITE_POSTROUTE] { rtpproxy_manage(); t_on_reply("MAIN_REPLY"); if(!t_relay()) sl_reply_error(); } When looking up the location schema this errors appear: 9(39) WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'location' while performing query 9(39) WARNING: db_redis [redis_dbase.c:1101]: db_redis_perform_query(): scan key 0 is 'expires' 9(39) WARNING: db_redis [redis_dbase.c:1101]: db_redis_perform_query(): scan key 1 is 'expires' 9(39) ERROR: <core> [db_ut.c:92]: db_str2int(): Unexpected characters: [-03-29 18:16:36] 9(39) ERROR: <core> [db_val.c:76]: db_str2val(): error while converting integer value from string 9(39) ERROR: db_redis [redis_dbase.c:1062]: db_redis_convert_row(): Failed to convert redis column 'expires' to db value 9(39) ERROR: db_redis [redis_dbase.c:1231]: db_redis_perform_query(): Failed to convert redis reply for row 9(39) ERROR: db_redis [redis_dbase.c:1243]: db_redis_perform_query(): failed to perform the query 9(39) ERROR: db_redis [redis_dbase.c:1829]: db_redis_query(): failed to do the query 9(39) ERROR: usrloc [udomain.c:981]: udomain_contact_expired_cb(): db_query failed Any hints? Thanks and regards,

3 2

Kamailio v5.2 kamctl alias add problem
by Igor Potjevlesch 01 Apr '19

01 Apr '19

Hi kamailios, I meet a problem with the " kamctl alias add" command in the 5.2 version, I saw that the module used to add an alias had changed, and now the command result with an error : [kamailio]# kamctl alias add XXXX@XXXX sip:XXXXX@XXXX { "jsonrpc": "2.0", "error": { "code": 500, "message": "Not enough parameters or wrong format" }, "id": XXXXX } I read the function who return this message " static void ul_rpc_add(rpc_t* rpc, void* ctx)" in the ul_rpc.c in the usr_loc module. I tried to pass the command with two more parameters : # kamctl alias add 0175951068(a)trunk7.bornsip.fr sip:0175951068@trunk7.bornsip.fr NULL NULL { "jsonrpc": "2.0", "result": { }, "id": 29200 } And this time it works correctly, but I saw thaht the test prameters was in the field "path" , do you know why we have to indicate a path ? Are we obliged to indicate this two more parameters ? What is the utility of this field ? Thank you in advance, Igor

2 2

where to find the cfg languaje for program on it?
by PICCORO McKAY Lenz 01 Apr '19

01 Apr '19

recently i wrote a portion of my cfg file but i found that i based my code in other sources and results of try/tach experiments i cannot found the documents for the languaje of the core i only found the link: https://www.kamailio.org/wiki/cookbooks/devel/core but in that link there's no references about languaje per see only about how to write portions of the file and sections.. maybe i wrong or the document are wrote in different way? by example how to evaluate the output of a custom function/section? Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

drop before exit and send ok reply first can be done?
by PICCORO McKAY Lenz 01 Apr '19

01 Apr '19

h have this for a custom rule agains spoecific messages: sl_send_reply("200", "OK"); drop(); exit; my question are if that rules are correct: i want to drop but send firts to client a ok response.. i mean client receive "all are ok dont worry" and then drop the rest packets.. that's correct? Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 2

kamailio and rtpengine
by David Villasmil 01 Apr '19

01 Apr '19

Hello guys, i'm trying to configure rtpengine, but when starting kamailio (with an empty table) i'm getting ```build_rtpp_socks(): no more pkg memory for rtpp_socks``` Can anyone help? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 5

kamailio 5.2.0 with empty table of rtpengine - kamailio doesn't start
by * Paolo Visintin - evosip.cloud 31 Mar '19

31 Mar '19

Hello, I noticed that , with kamailio ver. 5.2.1 starting with empty rtpengine table causes strange messages in kamailio and it doesn't start modparam("rtpengine", "db_url", "text:///tmp/kamailio/dbtext") cat /tmp/kamailio/dbtext/rtpengine id(int,auto) setid(int) url(string) weight(int) disabled(int) stamp(int) 0(2856) WARNING: tm [tm.c:760]: mod_init(): reparse_on_dns_failover is enabled on a multihomed host -- check the readme of tm module! 0(2856) WARNING: rtpengine [rtpengine_db.c:100]: rtpp_load_db(): No rtpproxy instances in database 0(2856) ERROR: rtpengine [rtpengine.c:2768]: select_rtpp_set(): no rtpp_set_list->rset_first 0(2856) NOTICE: rtpengine [rtpengine.c:1664]: mod_init(): Default rtpp set 0 NOT found 1(2857) NOTICE: usrloc [udomain.c:434]: preload_udomain(): filtered by server_id[0] 2(2858) ERROR: rtpengine [rtpengine.c:1722]: build_rtpp_socks(): no more pkg memory for rtpp_socks 2(2858) ERROR: <core> [core/sr_module.c:847]: init_mod_child(): error while initializing module rtpengine (/usr/lib/x86_64-linux-gnu/kamailio/modules/rtpengine.so) (idx: 2 rank: 2 desc: [udp receiver child=1 sock=172.28.0.120:5060]) 2(2858) ERROR: <core> [core/pt.c:342]: fork_process(): init_child failed for process 2, pid 2858, "udp receiver child=1 sock=172.28.0.120:5060" 2(2858) CRITICAL: <core> [main.c:1633]: main_loop(): Cannot fork 4(2860) ERROR: rtpengine [rtpengine.c:1722]: build_rtpp_socks(): no more pkg memory for rtpp_socks 4(2860) ERROR: <core> [core/sr_module.c:847]: init_mod_child(): error while initializing module rtpengine (/usr/lib/x86_64-linux-gnu/kamailio/modules/rtpengine.so) (idx: 4 rank: 4 desc: [udp receiver child=3 sock=172.28.0.120:5060]) 4(2860) ERROR: <core> [core/pt.c:342]: fork_process(): init_child failed for process 4, pid 2860, "udp receiver child=3 sock=172.28.0.120:5060" 4(2860) CRITICAL: <core> [main.c:1633]: main_loop(): Cannot fork any hint on what could be wrong ? (5.2.0 works perfectly) many thanks! Paolo

6 7

app_ruby - KSR::PV.sets method not found
by Andrew White 31 Mar '19

31 Mar '19

Hi all, I’m trying to set a pseudovariable via app_ruby. I’m able to retrieve pseudovariables using PV.gete as such: However trying to set these using sets, I get “No method found”: KSR::PV.sets("$fu", "sip:#{$uacs[trunk]['r_username']}@#{$uacs[trunk]['r_domain']}") Mar 30 04:44:59 ip-10-0-0-2 /usr/local/sbin/kamailio[28421]: ERROR: {1 102 INVITE 0fae8097312bec2b5ec79ada30c227e7@127.0.0.1:5060} app_ruby [app_ruby_api.c:107]: app_ruby_print_last_exception(): exception: undefined method `[]' for nil:NilClass PV.seti appears to work correctly. Am I calling the wrong method? Thanks! ________________________________ Andrew White - Director uConnected Email: andrew(a)uconnected.com.au Web: www.uConnected.com.au

2 4

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30 Mar '19

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1 0

[--SPAM--] Low %
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1 0

Lost connection to MySQL server during query
by JR Richardson 29 Mar '19

29 Mar '19

Hey Folks, Getting some annoying logs, pretty certain these are false positives. I'm running kamailio 4.4.7 using localhost mysql 5.5.60 on debian 7.11. Pretty simple setup, have acc and collecting cdrs. Relevant kamailio.cfg: loadmodule "acc.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----- acc params ----- modparam("acc", "db_url", DBURLRW) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" "accountcode=$avp(s:acctcode)") modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "cb=>mysql://rwuser:rwpass@localhost/kamailio") route[CDRS] { sql_query("cb","call kamailio_cdrs()","rb"); } Logs: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013) ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [call kamailio_cdrs()] CDRs are working fine, no issues with the results, just the log messages. Errors occur every 5 minutes so this is related to the rtimer cycle, mode=1 is set for new connection so there really wouldn't be a lost connection, the old connection should have been relieved on last cycle. I've seen some other post talking about this issue but could not find any resolution or work around. Any help is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

3 3

log timestamp granularity
by Patrick Murphy 28 Mar '19

28 Mar '19

Hello - Is there a way I can print current timestamp in milli or microseconds in the log file? Something like: 2019-03-28 04:50:15.379 would be ideal The closest thing I could find was $TV(Sn) but it doesn't suit the purpose here since its not human readable. Thanks Pat

2 1

flags definition in kamailio.cfg + 5.2 documentation
by Joel Serrano 27 Mar '19

27 Mar '19

Hi, I saw in the Kazoo default.cfg this: ... ####### Flags ####### flags FLAG_INTERNALLY_SOURCED: 1, FLAG_ASSOCIATE_SERVER: 2, FLAG_SKIP_NAT_CORRECTION: 3, FLAG_ASSOCIATE_USER: 4, FLAG_TRUSTED_SOURCE: 5, FLAG_SESSION_PROGRESS: 6, FLAG_IS_REPLY: 7, FLAG_SIP_TRACE: 8; ... So then I went here to read about it: https://www.kamailio.org/wiki/cookbooks/5.2.x/core#flags I found out it's not documented... is this a drop-in alternative for a: #!define FLAG_NAME FLAG_BIT? Sorry, first time I see this if not I would update the docs myself... :( Thanks, Joel.

4 8

Uri's user part parameters
by Duarte Rocha 27 Mar '19

27 Mar '19

Greetings, Does Kamailio have a way of identifying the user-part parameters of an URI? I have situations where i have sip:user;parameter=yes@host. Best Regards, Duarte Rocha

2 1

Manipulating User Agent header in dispatcher OPTIONS
by Andrew White 27 Mar '19

27 Mar '19

Hi all, I’ve set the server_header variable in my primary config, and I’ve got a catch in the default reply route to modify headers where required. I’m masking this value for security reasons, and filling it with my own value for vanity reasons :). I’ve noticed that the OPTIONS packet that is send via the dispatcher module when probing mode is on contains a User-Agent field which has the full version and OS type. Unfortunately I don’t necessarily have trust of the end destination being dispatcher to, nor the packet switchers in-between. How can I catch this OPTIONS packet to manipulate the value? It doesn’t appear to go through the REQUEST route or the ONSEND route. Thanks! ________________________________ Andrew White - Director uConnected Email: andrew(a)uconnected.com.au Web: www.uConnected.com.au

2 2

accounting and db_redis
by Daniel Tryba 27 Mar '19

27 Mar '19

Under https://www.kamailio.org/docs/modules/5.2.x/modules/db_redis.html#db_redis.… there is an example for using db_redis for accounting. This doesn't appear to work (5.2.2) modparam("db_redis", "keys", "acc=entry:callid,time_hires&cid:callid") results in: db_redis [redis_dbase.c:1886]: db_redis_insert(): inserting to prefix (table) 'acc' db_redis [redis_dbase.c:382]: db_redis_build_entry_keys(): build entry keys db_redis [redis_dbase.c:295]: db_redis_find_query_key(): checking for existence of ent db_redis [redis_dbase.c:306]: db_redis_find_query_key(): found key in entry key db_redis [redis_dbase.c:133]: db_redis_val2str(): converting str value '1db9c0bc57291e db_redis [redis_dbase.c:338]: db_redis_find_query_key(): entry key so far is 'acc:entr db_redis [redis_dbase.c:295]: db_redis_find_query_key(): checking for existence of entry key 'time_hires' in query db_redis [redis_dbase.c:346]: db_redis_find_query_key(): key 'time_hires' for type 'entry' not found, unable to use this type db_redis [redis_dbase.c:405]: db_redis_build_entry_keys(): Failed to create direct entry key, no matching key definition db_redis [redis_dbase.c:1892]: db_redis_insert(): failed to build entry keys db_redis [redis_dbase.c:1976]: db_redis_insert(): failed to do the insert What does work is "time" instead of "time_hires". Since I have no idea what values are available (I guess the table names?) and time only has a 1 sec resolution, what is a correct version of time_hires?

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