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46 participants
37553 discussions

DMQ_USRLOC module and calls
by Social Boh 13 Jul '21

13 Jul '21

Hello, I'm testing DMQ_USRLOC modulo between two Kamailio servers (A) and (B) The two servers share the same domain and using Amazon Route53 to distributes 50% requests each Server. The problem is when I try to make a call from user A (registered on kamailio A) and the user B (registered on Kamailio B). The call follow this flow: User A --> KamailioA --> UserB go directly to userB because KamailioA know, via DMQ_USRLOC IP and port of UserB. With some Softphone the call work with audio without problem; with other Softphone USERB never answer the INVITE sends from Kamailio. Is it possible use this flow? USER A --> KamailioA --> KamailioB --> User B to resolve this kind of problem or is there other available option? Thank you -- --- I'm SoCIaL, MayBe

5 15

Change routing for Register
by H Yavari 13 Jul '21

13 Jul '21

Hi to all, In a scenario, we need to change the destination of users' registration requests when they send the first 'Register' to Kamailio-A and it replies 401 to the user, next time the user sends 'Register' to Kamailio-B. User------Register-------> Kamailio-A User<-----401--------------Kamailio-A User-------Register--------> Kamailio-B RFC says we can't use the Record-Route for register because it doesn't create a dialog.Is there any option, header that Kamailio-A puts in the 401, and after that users send register to the Kamailio-B? Thank you.BR,Hossein

3 3

Kamailio v5.4.6 Released
by Daniel-Constantin Mierla 13 Jul '21

13 Jul '21

Hello, Kamailio SIP Server v5.4.6 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.6. Note that 5.4 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.5, with v5.5.0 being release out of it. For more details about version 5.4.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/06/kamailio-v5-4-6-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 Kamailio Advanced Training - Online - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

3 2

Using the dialog module. dialog was not destroyed with BYE message
by woenho＠gmail.com 08 Jul '21

08 Jul '21

According to the documentation a dialog is automatically destroyed when a BYE is received. But, dialog was not destroyed. What can I do? I am testing with kamailio 5.5 version

3 2

Subscriber Management using JSONRPC over HTTP
by Surabhi Garg 08 Jul '21

08 Jul '21

Hi Team, Greetings! I was trying to use HTTP to do the subscriber management (CRUD operations) & for that I am using JSONRPC-S module (JSONRPC over HTTP). But like other modules e.g. dispatcher, location I am unable to find any RPC commands for doing subscriber management (checked in auth_db) module. So is there a way to achieve subscriber management using JSONRPC over HTTP. Few HTTP examples which I am using for dispatcher module are given below. Similar commands I am looking for subscriber management as well. curl http://localhost:8081/RPC -H "Content-Type: application/json" -d " {\"jsonrpc\":\"2.0\",\"method\":\"dispatcher.add\", \"params\":[2, \"sip:127.0.0.1:5061\"], \"id\": \"1\"}" curl http://localhost:8081/RPC -H "Content-Type: application/json" -d " {\"jsonrpc\":\"2.0\",\"method\":\"dispatcher.remove\”, \"params\":[2, \"sip:127.0.0.1:5061\"], \"id\": \"1\"}" Surabhi Garg Mobile - 91 783 888 7523

2 1

How to check If Alias is already Set in Contact Header
by Juha Heinanen 08 Jul '21

08 Jul '21

Transformations (e.g. nameaddr.uri and uri.param.name) on $ct are your friends. -- Juha

2 2

How to check If Alias is already Set in Contact Header
by Terry Tailor 08 Jul '21

08 Jul '21

Hi, I'm having the following scenario UA <---> Prox1 <----> Proxy2 <-----> B2BUA <------> UA Proxy2 and B2BUA are connected on LAN and Proxy1 is adding his own Contact Alias and Proxy 2 is adding his own Contact Alias I'm adding Alias via using the following function set_contact_alias("1") as it will remove other contact alias but is causing the problem in the case of BYE coming from B2BUA. Is there is a way to find if the contact alias is already set? How I can replace the local IP alias with domain name when request is coming from B2BUA and alias is added by Proxy2. Here are logs to show problem in 200 sent by B2BUA and proxy2 add alias 10.44.44.2~6000~1 Now in ACK it is causing the problem SIP/2.0 200 OK Via: SIP/2.0/UDP sip.prox1.example.com:5060 ;rport=5060;received=3.18.1.21;branch=z9hG4bK2d66.a89039bedfe3d1113132f04c94b8a757.0 Via: SIP/2.0/UDP 192.168.1.168:50279 ;received=109.8.2.1;rport=45607;branch=z9hG4bKPj84875d08eafc48bfa6c29de5fe245cea Record-Route: <sip:sip.proxy2.example;lr;did=e84.87e2;rtp=bridge> Record-Route: <sip:sip.prox1.example.com;lr;did=e84.7452;rtp=bridge> From: "8001" <sip:8001@sip.example.com>;tag=0b9915abae654e21845fd4d7f577ede9 To: <sip:2501@sip.example.com>;tag=as1f4f6b1c Call-ID: a0095b9868924b0681225a8884a06c22 CSeq: 3687 INVITE Server: Voip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:2501@10.44.44.2:6000;alias=10.44.44.2~6000~1> Content-Type: application/sdp Require: timer Content-Length: 281 Ack for 200 ACK sip:2501@10.44.44.2:6000 SIP/2.0 Via: SIP/2.0/UDP sip.proxy2.example.com:5060 ;branch=z9hG4bK2d66.3c10a3b45e5c8caacbc5853d5e6418b2.0 Via: SIP/2.0/UDP sip.proxy1.example.com:5060;rport=5060;received= 3.18.1.21 ;branch=z9hG4bK2d66.788312006d5f7aedc058435cd203fed3.0 Via: SIP/2.0/UDP 192.168.1.168:50279;received= 109.8.2.1 ;rport=45607;branch=z9hG4bKPjf003c3bfe4ba423e9a0dedd647511dd3 Max-Forwards: 68 From: "8001" <sip:8001@sip.example.com>;tag=0b9915abae654e21845fd4d7f577ede9 To: <sip:2501@sip.example.com>;tag=as1f4f6b1c Call-ID: a0095b9868924b0681225a8884a06c22 CSeq: 3687 ACK Content-Length: 0 Thanks in advance

1 0

SECSIPID Identity Decode
by Daniel W. Graham 06 Jul '21

06 Jul '21

I am unable to base64url decode the json payload in identity header generated by secsipid. (Using python for test) decoded_payload = url64.decode(‘payload’) UnicodeDecodeError: 'utf-8' codec can't decode byte 0xc2 in position 27: invalid continuation byte Header decodes fine this way but not payload. Is this an issue with the payload encoding? Kamailio 5.5 Daniel W. Graham, CTO CMSInter.net LLC DIRECT (989) 400-4230 INTERNET | TELEPHONE | MANAGED IT

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I found bug in sipdump module.
by woenho 06 Jul '21

06 Jul '21

2 1

Re: [SR-Users] SECSIPID Identity Decode
by Daniel W. Graham 03 Jul '21

03 Jul '21

This was already pointed out, $mb. Thanks, -dan On Jul 2, 2021, at 4:40 PM, Daniel W. Graham <dan(a)cmsinter.net> wrote:

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