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count of registered users or count of tls connections
by Karthik Srinivasan 24 May '18

24 May '18

Hello, Via the routing file i am trying to find access two items:

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Core Value : myself
by Duarte Rocha 23 May '18

23 May '18

Greetings, I'm doing some work with the AUTH routes and modules, and sometimes i get a "403 Not relaying" answer when the INVITE has a parameter in the R-URI. That parameter was not included in the REGISTER. I think it's this part of the code that throws this reply : # 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; } Does a different or new parameter in the R-URI trigger the "uri!=myself" ? if so, can I set it up to check R-URI but not check the parameters? Thanks

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Any modules with LRN dipping capacity?
by KamDev Essa 22 May '18

22 May '18

Are there any modules with LRN dipping capacity?. If not, whats the best way we can implement it as a bolt on. KD

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Best way to implement custom LCR
by KamDev Essa 22 May '18

22 May '18

I apologize for asking unit questions today and decided to just present the big picture and address what I am trying to get @ holistically. I want to implement a custom LCR only solution on kamailio. INVITE in 300 out. Nothing else. I expect awesome cps stats that makes we lean towards kamailio rather than throwing up a home grown SIP server and worrying about performance. I have the guts of the LCR logic down pat, but todays its manual. In my process I do the following : - Check LRN of the requested user. I use dig doing a ENUM call. - Use the requesting user and LRN of requested user to mediate call type. Local or Interstate using LRN check via ENUM call (above) and LCAD data lookup. I use sql query. - Mediate call type Intrastate or International if not Local or Interstate using LERG lookup. SQL query too. - dip into database with requesting user and call type and get N lists by costs ascending. SQL query. - Construct Contact info in db or kamailio. - Send 300 reply will contact having the various routes. I can automate 2 through 6 using Kamailio. Basically xquery with supplied data and deal with the Contact header with data received. 404 or default route if no data received. I cannot handle the ENUM in kamailio or mysql. The provider does not provide SIP redirect. What are my options, besides changing LRN providers? KD

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300 Multiple Choices response engine
by KamDev Essa 22 May '18

22 May '18

Whats the best route to implement a basic 300 Multiple Choices response engine. INVITE in 300 Multiple Choices out with choices in the response. I have a logic to generate my list choices, so a basic process would be better. H

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2 pull requests: issue #1168 and another small issue
by Valentin Christoph 22 May '18

22 May '18

Hi all, Issued two pull requests. Any comment welcome, in particular, whether it will be taken for master branch...... https://github.com/kamailio/kamailio/pull/1538 https://github.com/kamailio/kamailio/pull/1539 Kr Christoph 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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I think I found issue to Kamailio not forwarding ACK sent to it by Asterisk
by Wilkins, Steve 22 May '18

22 May '18

Hello All, When Kamailio sends a '200 OK' to Asterisk, it is putting its Public IP into the Record-Routes. In Asterisk 14.6, it would send the 'ACK' back to the Private IP Address of Kamailio, but Asterisk 15.x is using the Public IP Address that Kamailio placed in the Record-Routes so... Is there a way to force Kamailio to set the Record-Route to its Private IP address of the first Record-Route in messages forwarded to Asterisk? Thank you?

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rtpp_test() in kamailio log
by Giovanni Tommasini - evosip 22 May '18

22 May '18

Hi guys, I've a question on using rtpengine with kamailio *scenario*: kamailio istance with rtpengine enabled and configured in localhost (but with a remote rtpengine is the same) modparam("rtpengine", "rtpengine_sock","udp:localhost:22222") kamailio version 4.4.6 + rtpengine version 6.3.0.0 running on Ubuntu 16.04 when I start kamailio I see a lot of INFO log rows like that May 22 12:13:12 my-dev /usr/sbin/kamailio[3388]: INFO: rtpengine [rtpengine.c:2546]: rtpp_test(): rtp proxy <udp:localhost:22222> found, support for it enabled May 22 12:13:12 my-dev /usr/sbin/kamailio[3397]: INFO: rtpengine [rtpengine.c:2546]: rtpp_test(): rtp proxy <udp:localhost:22222> found, support for it enabled ... more to the point. I see always 24 rows for each rtpengine_sock configured i n rtpengine.c I find the rtpp_tes function and that's fine, but why 24 times? thanks a lot -- Giovanni Tommasini | evosip web: evosip.cloud

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Kamailio sends SIP messages to old websocket after reconnection during call
by Idris AVCI 22 May '18

22 May '18

Hi, We use kamailio as a backend for JSSIP based WebRTC UAs. We use SIP over Weboscket as transport and everythings work fine generally. Some websocket connections to kamailio drop during an active call and JSSIP reconnects immediately. New websocket connection has a different port than the dropped one but kamailio tries to send to old websocket port which is not active anymore. We use GRUU and configured all the related parameters in registrar, dialog, tm, websocket, tls, xhttp and userloc modules both for versions 4.4 and 5.1 but couldn't find a solution yet. For a typical call below are the ordered events: 1. UA register to kamailio over wss (Code: WS1) 2. Kamailio send an INVITE to the UA (Code: Call1) 3. UA answer and media starts 4. UA drops WS1 to kamailio and reconnect (Code: WS2) 5. UA has an active websocket (WS2) to kamailio 6. Kamailio tries to send BYE message of Call1 to WS1 which is no longer active and complains that websocket could not be found 7. Kamailio send an INVTE to the UA (Code: Call2) over WS2 8. UA answer and media starts 9. Kamailio send BYE message of Call2 to UA over WS2 and everything work as expected (No websocket drop) Error log: WARNING: <core> [msg_translator.c:2761]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found I think we need a mechanism that will update the dialog information of Call1 to use WS2 after reconnection. Any help greatly appreciated. Thanks.

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htable - unable to update entry when event_route[htable:expired:<table>] is executed
by David Heath 22 May '18

22 May '18

Hi, Looking for some help/ advice on updating the expiration timer/ values in a hash table when when [htable:expired:<table>] occurs. On startup my script successfully reads in a set of entries from a DB; I also know from testing/ tracing that I can add new entries to the hash table in request_route. My issue occurs when the expiration timer kicks in - my goal was to re-read from the DB to ensure either the entry still exists or hasn’t changed and either delete, update or refresh the value until the next time the expiration event occurs. This way entries in the DB could be modified overtime. At the moment all that’s occurring is that in the event_route[htable:expired:cli] block the entries are being drained one-by-one. I re-read the row from the from the DB and I reset the value using $sht(cli=>$shtrecord(key)) = $xavp(cli=>MAPPED_CLI); I’ve also tried updating the expiration timer against the key. I’ve tried both $shtex(cli=>$shtrecord(key)) = 60 and $shtex(cli=>$shtrecord(key)) = $TS+60 [the latter based on the values shown from sht_print()] but the entries still get deleted. I’ve tried this against both 5.1.0 & 5.1.2. Any help appreciated. Thanks David Heath

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