- sr-users - kamailio.org

Inserting a message into a transaction?
by Daniel Tryba 20 Feb '18

20 Feb '18

For a retarded endpoint (an old ISDN PBX behind a Smartnode) I need to insert a 180 reply before sending a 183 (with SDP), otherwise the PBX fails to playback the inband audio to the caller behind the PBX. I can't find hints on how to do this. uac_req_send() can be used to start a new transaction, but I only need to generate a 180 based on the 183 in the on_reply route. Current Q&D hack is to send a 180 on the initial INVITE, works but not ideal.

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Re: [SR-Users] lost mysql connections
by Jan Gaida 20 Feb '18

20 Feb '18

Hello juha, Sorry for getting back on this old topic. Can you confirm that with Kamailio v4.3 your problem was solved? Regards, Jan

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Create ISUP Body
by Duarte Rocha 20 Feb '18

20 Feb '18

Hey Everyone I'm setting up a load testing scenario for my Kamailio. In order to do that i've created another Kamailio to answer "200 OK" to an Invite in order to simulate a call being answered. One of the scenarios I need to test is reading values from the ISUP IAM or ACM body. Here's the code on the Kamailio that replies with 200 : $var(acm) = "06 02 04 01 2c 01 fb 0c 08 83 10 19 7c 82 12 15 00 36 01 19 40 01 00 39 06 2c c0 36 c0 40 c0 00"; set_reply_body("$var(acm)","application/isup;base=itu-t92+;version=etsi356"); if( is_method("INVITE") && !has_totag() ) { · xnotice("First Invite"); send_reply("200","OK"); } However, when i use this and i do a pcap i get a "Message Type: Overload", which doesn't support optional parameters. What am i doing wrong ? Are there other methods to setup a simple ISUP part without hex ? Thanks in advance

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Kamailio mod ALIAS_DB: multiple aliases
by Stefan Rüetschli 20 Feb '18

20 Feb '18

Hello Kamailio-Users I'm using kamailio 5.1 with Alias_DB module. My subscribers register with a username like 0000001, 0000002 etc... Every user has a list of multiple aliases which they can use on their SIP-Trunk. I want to avoid that the user can use any alias for outgoing calls, only the aliases which are boud to his username in the aliases-db. So, if the user uses an alias which does not belong to him, i want to overwrite it to the first alias for his username in the aliases-db. At the moment, the function alias_db_find returns me the last alias found in the db. Is there any way to do it? Or is there a way to extend the dbaliases with a new column marking the "primary" alias for an account?|||| This would be the perfect solution. Regards Stefan

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Info: Kamailio World 2018 - Preview With 15 Sessions
by Daniel-Constantin Mierla 20 Feb '18

20 Feb '18

Hello, less than 3 months till the start of the 6th edition of Kamailio World Conference, time if flying fast! About one week ago we published the details for a group of accepted speakers, today we made a selection of sessions at the Kamailio World 2018. We had more proposals than we could accommodate, we are trying hard to fit in as many as possible, taking also in consideration the feedback from the participants at the past editions. For now you can head to the Schedule page and see the details of 15 sessions, from both workshops and conference days: https://www.kamailioworld.com/k06/schedule/ A very divers range of topics, from using Kamailio for emergency services (112/911), scaling with Redis backend, deploying in a containerized environment with Docker and Kubernetes, how to migrate the SIP routing logic to rich KEMI languages such as Lua, Python or Javascript, unit testing for Kamailio and test driven deployments, to blockchains in telephony, using Kamailio and FreeSwitch together, or latest updates from Asterisk PBX. The IMS/VoLTE workshop is going to show what you can do with latest Kamailio in mobile networks. And, of course, we have the very popular two sessions that never missed a Kamailio World edition: Dangerous Demos with James Body and VUC Visions with Randy Resnick. The details for other speakers and sessions will be published in the near future, stay tuned! Do not miss Kamailio World Conference 2018, it is going to be another great edition! You can register now at: https://www.kamailioworld.com/k06/registration/ Looking forward to meeting many of you at the next Kamailio World Conference, during May 14-16, 2018, in Berlin, Germany! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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IMS: maybe a bug in S-CSCF: should we create an issue?
by Valentin Christoph 20 Feb '18

20 Feb '18

Hi all, We are using kamailio 5.1.0 as S-CSCF as follows: We have two machines and two kamailio instances, one on each machine, both working on the same database (together they build one highly available S-CSCF). After registering a user via the first kamailio instance, then the first instance is stopped (simulating an outage), and the HSS sends an RTR to the second instance. However, the second instance does not find the registered user in its cache and it does not access the database --> the deregistration fails. Configuration: kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "enable_debug_file", 0) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "matching_mode", 0) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_url", DB_URL) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_mode", 2) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "timer_interval", 10) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "timer_procs", 4) Does anyone immediately "see" a mis-configuration, or should we create an issue? Details in the mail below. Thanks, Christoph From: Friedrich Peter Sent: Friday, February 16, 2018 6:38 PM To: Valentin Christoph <Christoph.Valentin(a)kapsch.net> Cc: Onic Roman <Roman.Onic(a)kapsch.net>; Habashizadeh Mahmoud <Mahmoud.Habashizadeh(a)kapsch.net> Subject: AW: Kein RTR Message nachdem SCSCF-1 outage Hello, following scenario at REGISTER of catt subscriber: start with empty db and empty usrloc cache node1(.23): MariaDB [scscf]> select * from impu; Empty set (0.00 sec) [root@vict-ims ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net [root@vict-ims ~]# [root@vict-ims2 etc]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net [root@vict-ims2 etc]# After catt REGISTER: catt_subscriber visible in DB and usrloc cache of .23 but missing in usrloc cache of .24 MariaDB [scscf]> select * from impu; +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ | id | impu | barring | reg_state | ccf1 | ccf2 | ecf1 | ecf2 | ims_subscription_data | +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ | 196 | sip:catt178@vict-ims.net | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | | 197 | sip:+4350255872178@vict-ims.net | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | | 198 | tel:+4350255872178 | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ 3 rows in set (0.00 sec) MariaDB [scscf]> [root@vict-ims ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net { impu: sip:catt178@vict-ims.net state: registered barring: 0 ccf1: ccf2: ecf1: ecf2: subscription: { impi: catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> service profiles: { 1: { impus: { 1: sip:catt178@vict-ims.net 2: sip:+4350255872178@vict-ims.net 3: tel:+4350255872178 } } } } contacts: { sip:catt178@193.154.113.40:5060;transport=UDP;rinstance=317-catt178: { expires: 3542 client: n/a } } } [root@vict-ims ~]# [root@vict-ims2 etc]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net [root@vict-ims2 etc]# After kamctl stop + kamctl start of .24 catt_subscriber is visible in usrloc cache of (.24): preload_data loads data from db at module init [root@vict-ims2 ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net { impu: sip:catt178@vict-ims.net state: registered barring: 0 ccf1: ccf2: ecf1: ecf2: subscription: { impi: catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> service profiles: { 1: { impus: { 1: sip:catt178@vict-ims.net 2: sip:+4350255872178@vict-ims.net 3: tel:+4350255872178 } } } } contacts: { sip:catt178@193.154.113.40:5060;transport=UDP;rinstance=317-catt178: { expires: 2949 client: n/a } } } [root@vict-ims2 ~]# Now deregister catt_subscriber from HSS1: catt_subscriber disappears from db and usrloc cache of (.23) but persists on (.24) as registered in usrloc cache HSS >hss.deregister_subscriber catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> 0 [root@vict-ims ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net [root@vict-ims ~]# MariaDB [scscf]> select * from impu; Empty set (0.00 sec) MariaDB [scscf]> [root@vict-ims2 ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net { impu: sip:catt178@vict-ims.net state: registered barring: 0 ccf1: ccf2: ecf1: ecf2: subscription: { impi: catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> service profiles: { 1: { impus: { 1: sip:catt178@vict-ims.net 2: sip:+4350255872178@vict-ims.net 3: tel:+4350255872178 } } } } contacts: { sip:catt178@193.154.113.40:5060;transport=UDP;rinstance=317-catt178: { expires: 2571 client: n/a } } } [root@vict-ims2 ~]# Scenario of RTR from HSS after REGISTER of catt_subscriber at (.23) and outage of (.23): subscr. visible in db but not visible in usrloc cache of .24 HSS sends diameter RTR message to (.24) and deregister callback in (.24) searches in usrloc cache and does not find catt_subscriber and fails. MariaDB [scscf]> select * from impu; +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ | id | impu | barring | reg_state | ccf1 | ccf2 | ecf1 | ecf2 | ims_subscription_data | +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ | 196 | sip:catt178@vict-ims.net | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | | 197 | sip:+4350255872178@vict-ims.net | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | | 198 | tel:+4350255872178 | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ 3 rows in set (0.00 sec) Cfg file of (.24): kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "enable_debug_file", 0) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "matching_mode", 0) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_url", "cluster://cluster1") kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_url", DB_URL) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_mode", 2) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "timer_interval", 10) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "timer_procs", 4) lg P The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

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issue with running kamailio
by justin barnes 20 Feb '18

20 Feb '18

I am installing from githum kamailio 5.0 on my ubuntu-16.04-x86_64 VPS. when I start the kamailio server I get this error : /# /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceJob for kamailio.service failed because the control process exited with error code. See "systemctl status kamailio.service" and "journalctl -xe" for details. failed! Ilooking up the status, I get this kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/etc/systemd/system/kamailio.service; disabled; vendor preset: enabled) Active: failed (Result: exit-code) since Mon 2018-02-19 15:23:41 EST; 4h 10min ago Process: 18969 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GR Feb 19 15:23:41 khynepassion systemd[1]: Starting Kamailio (OpenSER) - the Open Source SIP Server... Feb 19 15:23:41 khynepassion systemd[18969]: kamailio.service: Failed at step EXEC spawning /usr/sbin/kamailio: No such file or directory Feb 19 15:23:41 khynepassion systemd[1]: kamailio.service: Control process exited, code=exited status=203 Feb 19 15:23:41 khynepassion systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Feb 19 15:23:41 khynepassion systemd[1]: kamailio.service: Unit entered failed state. Feb 19 15:23:41 khynepassion systemd[1]: kamailio.service: Failed with result 'exit-code'.

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LCR options as of 2018
by Francisco Valentin Vinagrero 19 Feb '18

19 Feb '18

Hello, I am configuring a Kamailio node that will interface with 3 carriers and will implement Least Cost Routing for around 900 destinations. I have already implemented a quick prototype with the LCR module and I have been reading the CarrierRoute doc and the CGrates site. I don't really need advanced functionality, just: - Re-routing to a backup carrier when the main one is not available. - Ping with OPTIONS to all carriers so I can detect if any of them is not available (à la Dispatcher) - Translate prices from a central DB to routing priorities. - Get some accounting post-call info about the routing decision (acc module) So far, while testing the LCR module I have had some issues with the rule IDs being integers that auto-increment, so the LCR rules need to be referenced from the rule_target table using this numeric ID. I can see some issues when scaling this to 900 lcr_rules and 2700 lcr_rule_targets with 3 priorities. In addition to that, I think the OPTIONS mechanism is just used by the lcr module when a gateway is marked as defunct, but the gateways are not actively monitored in the normal status. Is this correct? I have been considering CarrierRoute, which has a slightly simpler table structure, but I can't find any other advantage for my use case. Regarding CGrates, it looks great but it looks like it exceeds my current needs. For those who have implemented LCR in your kamailios, am I missing something regarding performance, scalability, etc? Are there any newer modules that can implement this, or maybe just a dispatcher-on-steroids setup will do the trick? Cheers, Francisco.

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User state
by Magnus Löfqvist 19 Feb '18

19 Feb '18

Hi, We are running for a customer a kamailio server for user registration and routing of calls. Also using Asterisk for queue management and IVR. Now we are in the progress of writing a web portal where the customers users should be able to see if there collueges are busy in phone or availible. In asterisk I would use hints and listing in events in AMI. But what way are the best in Kamailio? Best Regards Magnus Sent from Nine<http://www.9folders.com/>

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PR1447: first attempt at app_python3
by Anthony Alba 18 Feb '18

18 Feb '18

Hi Daniel and kamailio contributors, I have submitted a PR for app_python3 here: https://github.com/kamailio/kamailio/pull/1447. It is very rough, but would appreciate your feedback. Notes: 1. Almost a verbatim port of app_python; just changing obvious semantics. String->Unicode is practically the only large change. No change to symbol names and export functions so the two modules cannot coexist. (Don't loadmodule "app_python.so" and loadmodule "app_python3.so) 2. Module initialization semantics in Py3 have changed: add the top level modules to the inittab before calling Py_Initialize(). I have done this for KSR and Router. I.e. PyImport_AppendInittab() is called for Router and KSR *before* Py_Initialize() 3. Makefile looks for a python3 binary (PYTHON3 in the Makefile) to get include paths and link libraries. Here is an example compile line for IUS Python 3.6 on CentOS 7 (no native Python3 from RHEL) ## default: needs to find python3 and python3-config ## needs to find both python3.6 and python3.6-config make -C src/ PYTHON3=python3.6 SKIP_MODULES="${SKIP_MODULES}" quiet=verbose module_group_py="app_python3" group_include=py every-module 4. The semantics for GIL and thread management are too different for one-to-one replacements. I have therefore decided to comment out all threading calls. Fortunately, kamailio does not used threads at the moment. This is more complicated than it seems: even with Py2+thread management: there would be deadlocks if kamailio->python2 script->Cython extension with threads(e.g. grpcio). For an MVP it would be best to get it in without thread management. 5. It should backport to 5.1. 6. Tested locally, with a routing scripting making many python_exec() calls Thank you. Anthony Alba

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PUBLISH and ETag
by Cyrille Demaret 16 Feb '18

16 Feb '18

Hi, I'm using Kamailio with presence enabled and Asterisk PJSIP and outbound-publish. My problem is happening when I place 2 consecutive calls from Asterisk : When I make a first call Asterisk sent the following: PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e69723a610 82 From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10697 PUBLISH Event: dialog Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 247 <?xml version="1.0" encoding="UTF-8"?> early.. Kamailio replies : SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e6972 3a61082;received=192.168.100.37 From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-ff39 Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10697 PUBLISH Expires: 180 SIP-ETag: a.1518775074.19863.16.0 Server: kamailio (5.0.5 (x86_64/linux)) Content-Length: 0 When the call is done, Asterisk sent another PUBLISH telling that the call if terminated : PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97ab8c7 52 From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10698 PUBLISH Event: dialog SIP-If-Match: a.1518775074.19863.16.0 Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 230 <?xml version="1.0" encoding="UTF-8"?> terminated.. And Kamailio replies : SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97 ab8c752;received=192.168.100.37 From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-48b4 Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10698 PUBLISH Expires: 180 SIP-ETag: a.1518775074.19873.18.1 Server: kamailio (5.0.5 (x86_64/linux)) Content-Length: 0 Here, the SIP ETag is a.1518775074.19873.18.1. The problem is if I make a new call before the expiration of the previous SUBSCRIBE, Asterisk reuse this SIP ETag according to the RFC : PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj9d13bb82-31d9-48db-9672-bd4b6b4f22 f0 From: sip:201@mydomain.com;tag=33e6b028-0444-4b3a-8bc2-4a987a291528 To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10699 PUBLISH Event: dialog SIP-If-Match: a.1518775074.19873.18.1 Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 247 <?xml version="1.0" encoding="UTF-8"?> early. Kamailio refuse it with this error : "Trying to update an already terminated state. Skipping update." because the call is considered as terminated. The RFC is stating : When updating previously published event state, PUBLISH requests MUST contain a single SIP-If-Match header field identifying the specific event state that the request is refreshing, modifying or removing. This header field MUST contain a single entity-tag that was returned by the ESC in the SIP-ETag header field of the response to a previous publication. Why Kamailio is acting like that? Best regards, Cyrille

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Problem with "transport=tcp" and rtpproxy
by Pau Giralt 16 Feb '18

16 Feb '18

Hi! I'm working with kamailio and rtpproxy. When I configure the clients to send sip messages through TCP adding the ";transport=tcp" in the sip message the rtpproxy is not working anymore. The code is never passing the point pasted below, inside route[NATMANAGE] in kamailio.cfg if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; Therefore, rtpproxy_manage("co"); is never reached. Could it be a parsing problem? ( check_route_param("nat=yes") ) The config file is exactly the same as the one provided in the kamailio repository (the default one) Thanks!

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Raspberry Pi deb packages -> built
by Mititelu Stefan 15 Feb '18

15 Feb '18

Hi people! After a bit of research and lib installations, I've managed to build the full kamailio deb packages for a raspberry pi 3 (version B) running raspbian stretch (lite) that can be found at [1] (with a really few module exceptions that might be used in really specific scenarios -> I think) Just for fun, I'm planning to do some CPS stats tomorrow (if I got the time), with a basic config and also share here, but, if someone else tried it before and is able to share some stats/benchmarks, would be greatly appreciated (and interesting to compare, given my "decent" rpi overclock), because I'm also curious about VoIP stuff on raspberry pi! :D [1] https://github.com/smititelu/rpi/tree/master/kamailio/5.1.1 Thank you, Stefan

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KEMI
by Aleksandar Sosic 15 Feb '18

15 Feb '18

Hi everyone, we're valuating on rewriting our kamailio routes in python. We're doing a lot of string manipulation and GraphQL API queries so the kamailio scripting language is a little bit limited for this use case. Can someone tell us if the KEMI framework is stable enough for production? Kind regards, -- Aleksandar Sosic mail: alex.sosic(a)timenet.it skype: alex.sosic cell: +385 91 2505 146

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Re: [SR-Users] RTPEngine Transcoding
by Social Boh 15 Feb '18

15 Feb '18

Hello, I'm trying to use this configuration: Softphone A audio codec g722 and Opus Softphone B audio codec PCMA and PCMU Kamailio with rtpengine_manage with codec-transcode-PCMU codec-transcode-PCMA in the request Softphone A call Softphone B The RTPEngine DEBUG show many Decoder Error: https://pastebin.com/k1Zhw8HS Thank you Regards --- I'm SoCIaL, MayBe El 13/02/2018 a las 20:05, Richard Fuchs escribió: > On 2018-02-13 07:22 PM, Social Boh wrote: >> Thank you. >> >> which is the best way to implement ir >> >> codec-strip-all and the add codecs? >> >> Is there a document about how work the transcoding? > It's really up to you and depends on what you want to achieve. Doing a > strip-all and then adding codecs you want to offer is definitely one > way. Another is to just add codecs that you know or think the peer may > want to see. > > Cheers >

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Invalid payload location
by Soltanici Ilie 15 Feb '18

15 Feb '18

Hi All, Can someone explain why i'm receiving such errors in syslog: [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:228]: set_sdp_payload_fmtp(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location This is how the SIP Packet is looking: 2018/02/15 10:51:40.916040 46.166.139.7:30189 -> X.X.X.X:5060 INVITE sip:01031203697179@X.X.X.X:5060 SIP/2.0 Via: SIP/2.0/UDP 46.166.139.7:30189;branch=z9hG4bK-5342365299746297160962993236828333618949;rport=30189 From: <sip:88893855391@46.166.139.7:30189>;tag=0vkn019bur To: <sip:01031203697179@X.X.X.X:5060> Contact: <sip:88893855391@46.166.139.7:30189> Call-ID: 0889633580294714035892119863939070195463 CSeq: 101 INVITE User-Agent: Hello Annonymous Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 356 v=0 o=SIP-Session 950029666 1307322426 IN IP4 46.166.139.7 s=SIP c=IN IP4 46.166.139.7 t=0 0 m=audio 5075 RTP/AVP 18 0 8 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=fmtp:9 bitrate=64000 a=rtpmap:104 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv Thank You.

1 0

python_exec return 0 causes script hangs; return -1 is treated like 0
by Anthony Alba 15 Feb '18

15 Feb '18

5.1.1: if python_exec method returns 0, then the script hangs If the python_exec method return -1, then expressions treat it as if it returned zero. Somewhere there seems to be an offset by 1 issue. if (!python_exec("method")) { # if you return -1, the script does not hang but treated like zero } ################## Example 1: class kamailio(object): def hello(self, msg): KSR.info("hello, world\n") return 0 Calling: python_exec("hello") will cause script hang ################### Example 2: class kamailio(object): def hello(self, msg): KSR.info("hello, world\n") return -1 Calling python_exec("hello") if (!python_exec("hello")) { ## method returns -1 ## why did we get here? xlog("L_INFO","method return zero!\n"); } shows INFO: <script>: method return zero!

2 1

[IGNORE] 5.1.1 python_exec, extra chars ' 0' appended to args
by Anthony Alba 15 Feb '18

15 Feb '18

Apologies for the noise; it is working now. Not sure how I got the extra " 0" appended to the method arguments for python_exec(). AAlba

1 0

[BUG?] 5.1.1 python_exec, extra chars ' 0' appended to args
by Anthony Alba 15 Feb '18

15 Feb '18

Using 5.1, the python_exc args string is getting an extra [_0' appended When I call python_exec("method", "string"); In python: def method(self, msg, args): KSR.info("DBG: {} {} {}".format(msg, args)) This will show 'string 0'; instead of string. AAlba

1 0

RTPEngine Transcoding
by Social Boh 14 Feb '18

14 Feb '18

Hi List, I just read on the RTPEngine page the Proxy RTP now support transcoding: https://github.com/sipwise/rtpengine#transcoding is there a plan to support it? Regards -- --- I'm SoCIaL, MayBe

5 10

RTPEngine QoS statistics and leg labelling
by Alex Balashov 14 Feb '18

14 Feb '18

Hello, I am excited about the new media statistics from RTPEngine, exposed conveniently into PVs, e.g. jitter, packet loss, MOS, time offset of lowest and highest MOS scores, etc. Great work! I am, however, having a bit of trouble understanding how to properly apply this new functionality, as it is not clearly documented (IMHO): 1. What is the meaning of "global" statistics which do not take into account an "A" and a "B" side? That is to say, if I have a packet loss or a jitter or MOS score statistic using just the generic toplevel values, e.g. https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… What does that mean? A functional call ordinarily consists of an inbound and outbound RTP stream direction relative to the endpoints on both sides of RTPEngine. Inbound A --> server --> Outbound A Outbound B <-- server <--- Inbound B Where do the "global" stats situate themselves in relation to all this? 2. I am given to understand that the statistics for the "A" and "B" sides of a bidirectional RTP stream can be separated based on an arbitrary label, using these PVs: https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… How do I use it properly? My intuition was to try something like this: rtpengine_offer("replace-origin replace-session-connection ICE=remove label=outbound"); And in the answer/manage handler on the SDP answer: rtpengine_answer("replace-origin replace-session-connection ICE=remove label=inbound"); Later, when processing a BYE and obtaining subsequent media statistics, I tried this: if(check_route_param("proxy_media=yes")) { $avp(rtpengine_a_label) = 'outbound'; $avp(rtpengine_b_label) = 'inbound'; rtpengine_delete(); route(LOG_RTPENGINE_STATS); } But this only gets me stats for the "A" leg in the JSON object I emit: { "server_addr": "udp:127.0.0.1:5050", "a_label": "outbound", "b_label": "inbound", "a_leg": { "min": { "mos": "4.4", "at": "0:04", "packetloss": "0", "jitter": "0", "roundtrip": 0 }, "max": { "mos": "4.4", "at": "0:04", "packetloss": "0", "jitter": "0", "roundtrip": 0 }, "avg": { "mos": "4.4", "packetloss": "0", "jitter": "0", "roundtrip": 0, "samples": 2 } }, "b_leg": { "min": { "mos": "<null>", "at": "<null>", "packetloss": "<null>", "jitter": "<null>" }, "max": { "mos": "<null>", "at": "<null>", "packetloss": "<null>", "jitter": "<null>" }, "avg": { "mos": "<null>", "packetloss": "<null>", "jitter": "<null>" } } } I also tried rtpengine_delete("label=inbound") (and outbound) individually. Some clarity on how this all fits together would be appreciated. :-) Many thanks to Sipwise for great work on it! -- 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/

2 14

How to add a new module to existing Kamailio Server?
by Arun NV 14 Feb '18

14 Feb '18

I am using a *Kamailio 4.4* server. And I would like to install *http_async_client* module to this server. I have looked in the module directory(*/usr/local/lib64/kamailio/modules*) to check whether this *http_async _client* module already exists there, but it was not there. And I searched for a solution and only found <http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-modular-install> the method to include new module at the compile time. I could find the modules source code on GitHub <https://github.com/sipwise/kamailio/tree/master/modules/http_async_client>. But I don't know the steps to install a new module to the existing Kamailio Thank you, Arunbalan

2 1

Use RTPProxy in 302 scenario
by Fabian Borot 13 Feb '18

13 Feb '18

I have this scenario Customer --> Kamailio --> Provider1 The route where l route the call to Provider1 is called route["INCOMING"] and I "arm" the replies and failures to : t_on_reply("INCOMING") and t_on_failure("INCOMING"). Provider1 always replies with a 302 Redirect and in the failure_route["INCOMING'] I have this code: if (t_check_status("302")) { get_redirects("*"); t_on_branch("REDIRECT_BRANCH"); t_load_contacts(); t_next_contacts(); t_on_failure("SERIAL_FORKING"); t_relay(); exit; } In the t_on_failure("SERIAL_FORKING") block I do t_relay() as long as there are more contacts to be processed. Lets say I need to try Provider_2 and Provider_3 failure_route["SERIAL_FORKING"] { xlog("L_INFO","mylog: In Failure Route SERIAL_FORKING. Reply [$rs]\n"); if (!t_next_contacts()) { exit; } xlog("L_INFO","mylog: In Failure Route SERIAL_FORKING. Trying next Contact.\n"); t_on_failure("SERIAL_FORKING"); t_relay(); } I want to use RTPProxy in this scenario and I am using the function rtpproxy_manage() to let kamailio handle it automatically. RTPProxy ↑ | | Customer --> Kamailio --> Provider1 | | ↓ Provider N After the 302 response is processed I would like to have the signaling flow "Customer -> Kamailio --> ProviderN" and the RTP flow "Customer --> RTPProxy --> ProviderN" I have tried inserting the function in several places but can't make it work and what I mean is that sometimes the rtpproxy creates the 1st session and then deletes it as soon as I call it the 2nd time with a message similar to this one (log of rtpproxy) : "INFO:416f20b74a58fb75168175e61b2edece@192.168.1.61:5060:rtpp_cmd_delete_ematch: forcefully deleting session 1 on ports 63502/0". Sometimes the kamailio log shows this message: "ERROR: rtpproxy [rtpproxy.c:2536]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy" I think I must call rtpproxy_manage() before I send the INVITE to each of the Providers in the Contact list so they receive a SDP offer with an IP:Port combination provided by the RTPProxy (instead of the ones coming from the Customer). That would be in my route["INCOMING'] Then once Provider_N replies with a 183 with SDP or a 200 OK I would call it again to get a new session to be passed on the reply to my customer. But is the part I am struggling with. Should I add a new reply route inside the failure_route["INCOMING'] only if I am dealing with 302 and do it there ? I am out of ideas guys and any idea is welcome

2 2

Can rtpengine change network interface of estabished call?
by Yufei Tao 13 Feb '18

13 Feb '18

Hi, Is it possible to define two pairs of network interfaces for rtpengine, e.g. int/ext and up/down? If so during a call established between interfaces 'int' and 'ext', is it possible to send a re-INVITE from the calling party on interface 'ext', to update the RTP destination to an IP on a different interface 'up', for example? Or does the updated RTP destination have to be on the same interface as when the call was established? I have some success updating the RTP destination if it is on the same interface (different IP:port) however haven't managed to divert it to a different interface. I wonder if it is possible at all or me missing something? Cheers, Yufei

1 0

(no subject)
by Tihomir Culjaga 13 Feb '18

13 Feb '18

Hi, Im new to Kamailio so im sorry if some of my questions will sound a bit wierd :=) i have a setup like this: (Kamailio version 5.1) UA > Kamailio > Freeswitch Kamailio forwards registration to freeswitch without saving location. I have configured sip over websockets and im able to register to Freeswitcha via kamailio. Also, i use add_path_received() so kamailio adds a path header to register messages recv 945 bytes from udp/[192.168.50.60]:5060 at 22:38:07.283809: ------------------------------------------------------------------------ REGISTER sip:192.168.50.60 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435 Max-Forwards: 69 To: <sip:1001@192.168.50.60> From: <sip:1001@192.168.50.60>;tag=vq30modpgd Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Authorization: Digest algorithm=MD5, username="1001", realm="192.168.50.60", nonce="744aaa14-4a86-4ce3-b6d6-419b0a34ee90", uri="sip:192.168.50.60", response="9b592b874474d4ffe5769f2410baceef", qop=auth, cnonce="sf7ff0fp8tm5", nc=00000001 Contact: <sip:1001@192.0.2.110 ;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, 100rel, outbound User-Agent: SIP.js/0.7.0 BB Content-Length: 0 Path: <sip:192.168.50.60;lr;received=192.168.200.77~63296~6> ------------------------------------------------------------------------ send 719 bytes to udp/[192.168.50.60]:5060 at 22:38:07.329444: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435 From: <sip:1001@192.168.50.60>;tag=vq30modpgd To: <sip:1001@192.168.50.60>;tag=Spya88DKU9gSQ Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Contact: <sip:1001@192.0.2.110;transport=ws>;expires=600 Date: Fri, 09 Feb 2018 21:38:07 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Path: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6 Content-Length: 0 ------------------------------------------------------------------------ Once, registered, im trying to originate a call to the registered endpoint from freeswitch this results in an invite like this: send 1757 bytes to udp/[192.168.50.60]:5060 at 22:46:21.252549: ------------------------------------------------------------------------ INVITE sip:1001@192.0.2.110;transport=ws SIP/2.0 Via: SIP/2.0/UDP 192.168.50.65:15060;rport;branch=z9hG4bKXHcX316pDgXZB Route: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6 Max-Forwards: 70 From: "" <sip:1002@192.168.50.65>;tag=tZQ393yprj7BK To: <sip:1001@192.0.2.110;transport=ws> Call-ID: 8212b700-8885-1236-08b3-000c293ae94e CSeq: 118770326 INVITE Contact: <sip:mod_sofia@192.168.50.65:15060> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 891 X-FS-Support: update_display,send_info Remote-Party-ID: <sip:1002@192.168.50.65 >;party=calling;screen=yes;privacy=off Freeswitch appends a route header to the invite specifying kamailio as route and a received with original uri. well, here is the issue ... im failing to hit the SIP over websocket client... Kamailio my RELAY route is like this route[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_ALERT", "Its a Branch, go manage that !!! <$ru>\n"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_ALERT", "its onreply_route, go manage !!! <$ru>\n"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_ALERT", "Its failure_route, go manage !!! <$ru>\n"); t_on_failure("MANAGE_FAILURE"); } } $fs = "udp:192.168.50.60:5060"; xlog("L_ALERT", "Trying to rely !!! <$ru>\n"); if (!t_relay()) { xlog("L_ALERT", "failed to rely !!! <$ru>\n"); sl_reply_error(); } exit; } here is a part of the log: Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced udp:192.168.50.60:5060, to ws:192.0.2.110:5060) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: <core> [core/msg_translator.c:424]: clen_builder(): content-length: 916 (916) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/msg_translator.c:2765]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: <core> [core/msg_translator.c:1980]: build_req_buf_from_sip_req(): could not create Via header Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -2 (-2) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:352]: t_relay_to(): -2 error reply generation delayed Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=744 a=26 n=xlog Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ALERT: <script>: failed to rely !!! <sip:1001@192.0.2.110;transport=ws> is that because i set a $fu before t_relay() ? any idea how to approach the issue ?

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