[OpenSIPS-Users] Problems with loose_route and ACK

D: 6d8b0c06-5b00-1238-3aa8-fa163e0144b5. CSeq: 10197616 ACK. Contact: . Content-Length: 0. This used to work in an older setup with Opensips running 2.1, so not sure if I'm doing something wrong here. Any hints are appreciated Thanks in advance / Olle ___

Re: [OpenSIPS-Users] UDP fragmentation in reply routes

be set when you open a socket, that’s why I wonders if Opensips might use those parameters or not, especially since we have so very different behaviour in different directions. BR/Olle Från: Users För Maxim Sobolev Skickat: den 18 maj 2020 22:03 Till: OpenSIPS users mailling list Ämne: Re

Re: [OpenSIPS-Users] UDP fragmentation in reply routes

now. My prime suspect is Centos since it send out the first part of the fragmented packet but not the following part that would complete the packet. But indeed it is a strange bug, since it does not always happen. BR/Olle Från: Users För Giovanni Maruzzelli Skickat: den 19 maj 2020

[OpenSIPS-Users] rtpengine and multiple instances

for redundancy, and this leads to my next question: Can you configure with fifo commands so a node is enabled but have weight 0 , and is only used in case the primary node fails? BR/Olle ___ Users mailing list Users@lists.opensips.org http

Re: [OpenSIPS-Users] rtpengine and multiple instances

Thanks, sounds like it's time to upgrade  BR/OLle -Original Message- From: Users On Behalf Of Razvan Crainea Sent: den 6 oktober 2023 13:13 To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] rtpengine and multiple instances Hi, Olle! Yes, the offer should be taken by one

[OpenSIPS-Users] OpenSIPS on Windows CE

there have tried OpenSIPS on Windows CE. BR/Olle ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Problem with event based routing

e[INSERT_CALL] { t_inject_branches("event","cancel"); } branch_route[1] { route(RELAY); exit; } BR / Olle PS I send another mail on the same subject since I missed your reply pls ignore that. Från: Bogdan-Andrei Iancu [mailt

[OpenSIPS-Users] Problems with EBR

the call we get this error: CRITICAL:tm:w_t_relay: unsupported route type: 8 I guess we have missed something in the realy so it would be great if you could share the full opensips.cfg file for the demo. We run the latest Opensips 2.3.2 from repo. Thanks in Advance /Olle

Re: [OpenSIPS-Users] Problem with event based routing

;transport=TLS;ob, injecting it in transaction One could of course check if this contact IP/port have already been injected into the call, but I just wondered if this is anything you have seen before. BR/Olle Från: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Skickat: den 26 januari

[OpenSIPS-Users] Username in DB URL

Mysql -h host.mysql.azure.com -u opensips@host -popensips opensips, which works fine In clustererar where we use this this translates to: modparam("clusterer", "db_url","mysql://opensips@host:opens...@host.mysql.azure.com/opensips") which fails Any idea

[OpenSIPS-Users] Problem with event based routing

: unsupported route type: 8 It would be great if you could share the configuration file that is used in the example mention in the blog post. BR/Olle ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman

[OpenSIPS-Users] Problem with force_send_socket

ideas? Thanks in advance / Olle ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Remove contacts frol location table with same SIP_instance ID

Hi thanks for the patch, I will test this once we have updated to 2.4 in the middle of that atm. Br/Olle Från: Users [mailto:users-boun...@lists.opensips.org] För Liviu Chircu Skickat: den 24 maj 2018 08:26 Till: users@lists.opensips.org Ämne: Re: [OpenSIPS-Users] Remove contacts frol

Re: [OpenSIPS-Users] rtpengine_offer error

Try replace comma with space rtp/avp rtp/save I’m not sure of you can offer both at the same time Br Olle Skickat från min iPhone > 30 juni 2019 kl. 09:43 skrev Dragomir Haralambiev : > > Hello, > > From rtpengine_offer manual I see: > > RTP/AVP, RTP/SAVP, RTP/AVPF, RT

Re: [OpenSIPS-Users] rtpengine_offer error

We solved this by adding info in Register Another option could be if you know if avp/ savp is tied to transport protocol So TLS implies savp Br Olle Skickat från min iPhone > 1 juli 2019 kl. 09:11 skrev Dragomir Haralambiev : > > > Hi Alexej, > > Yes. You a

[OpenSIPS-Users] UDP fragmentation in reply routes

sue. BR/Olle ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Reloading TLS certificates without restart

Hi but them in a DB then you can reload afaik / Olle Skickat från min iPhone > 20 maj 2020 kl. 19:06 skrev William Simon : > >  > Is it possible to reload the TLS certificates without a restart? My TLS > domains are defined in the config script. I just want to repla

Re: [OpenSIPS-Users] UDP fragmentation in reply routes

routes Br Olle Skickat från min iPhone > 20 maj 2020 kl. 17:14 skrev junkmail : > > Hello, I had run into the same issue. One thing I was a bit mistaken > because I was using tcpdump and doing a capture filter of port 5060 or the > such. So I was missing the Fragment in my s

Re: [OpenSIPS-Users] UDP fragmentation in reply routes

Hi thanks for the tip, how dit you find it? I just capture 3 ports in my tcpdump. Br Olle Skickat från min iPhone > 20 maj 2020 kl. 19:18 skrev junkmail : > > Sorry that is what I was trying to let you know. Is that I had thought the > same thing that the Fragment was n

Re: [OpenSIPS-Users] UDP fragmentation in reply routes

Thanks for the tips will give it a try to see what happens, but I guess TCP is the solution. Br Olle Skickat från min iPhone > 21 maj 2020 kl. 07:41 skrev junkmail : > > Yea that is it. > > so if you are doing something like tcpdump udp port 5060 or udp port 5080 > et

Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

10 feb 2009 kl. 13.10 skrev Iñaki Baz Castillo: 2009/2/10 Johansson Olle E o...@edvina.net: If both devices are on private IP's, there's going to be two RTP proxys involved if they're on different SIP networks. Each SIP service needs an RTP proxy for supporting their local users. Hi, I

Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

10 feb 2009 kl. 13.44 skrev Iñaki Baz Castillo: 2009/2/10 Johansson Olle E o...@edvina.net: alice --- (NAT A) --- ProxyA RtpProxyA --- ProxyB RtpProxyB --- (NAT B) --- bob In this case, when alice calls bob, ProxyA will apply RtpProxyA so the SDP will contain a public IP. Since ProxyB

Re: [OpenSIPS-Users] Pickup of a ringing extension under OpenSIPS

24 feb 2009 kl. 13.39 skrev Yehavi Bourvine: Hello, I am in the process of duplicating my Asterisk system into OpenSIPS in order to allow for a future growth. I need to do directed pickup when another extension rings. How do I do that? (assuming I know who wants to pickup what).

Re: [OpenSIPS-Users] XMPP Issue

for how to set up component connections in your XMPP server. Regards, /Olle --- * Olle E. Johansson - o...@edvina.net * Asterisk/OpenSER Training http://edvina.net/training/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi

Re: [OpenSIPS-Users] [NEW] SDP codec manipulation feature

http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8

Re: [OpenSIPS-Users] [NEW] SDP codec manipulation feature

28 jul 2009 kl. 16.53 skrev Alex Balashov: It's worth pointing out that no member of the OpenSER project stack has been a pure SIP proxy for very long; they have certain UAS features (registrar, PUA, NAT ping, etc.) As Bogdan said, a pure proxy would not be terribly useful in most

Re: [OpenSIPS-Users] Opensips and/or Asterisk 1.4 with directrtpsetup=yes

10 aug 2009 kl. 18.31 skrev Ricardo Martins: Hi all! I found some performance reports about asterisk handling a 250 calls per second limit with a 2.8GHz processor what makes it quite bigger than transnexus reports for openser performance (60 calls per second per GHz). Did opensips make

Re: [OpenSIPS-Users] Opensips and/or Asterisk 1.4 with directrtpsetup=yes

12 aug 2009 kl. 16.18 skrev Klaus Darilion: Olle E. Johansson schrieb: 10 aug 2009 kl. 18.31 skrev Ricardo Martins: Hi all! I found some performance reports about asterisk handling a 250 calls per second limit with a 2.8GHz processor what makes it quite bigger than transnexus reports

Re: [OpenSIPS-Users] Ucarp (new topic)

2 sep 2009 kl. 20.46 skrev Brett Nemeroff: UCARP is pretty simple as well: http://www.ucarp.org/project/ucarp Similar to the heartbeat (linuxHA) stuff, but a lot more lightweight from my experience. Can you tell us a bit more about your experience of this? In production? Requirements

Re: [OpenSIPS-Users] RTPPROXY

3 nov 2009 kl. 12.36 skrev Raúl Alexis Betancor Santana: On Tuesday 03 November 2009 10:15:31 michel freiha wrote: Dear All, I would like to ask you please if there is a way to use OpenSIPs with a proxy for rtp other than rtpproxy? Mean if i can use a propriatery rtp proxy All you

Re: [OpenSIPS-Users] Any word on the B2B bug?

of software for free, maybe it's time to contribute as well? :-) /Olle ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] LDAP authentification

by the auth module. Good luck! /Olle ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] re-REGISTER with To tag gets 404

31 dec 2009 kl. 12.12 skrev Victor Pascual Avila: Please, see RFC 3261 - Section 10.2: A REGISTER request does not establish a dialog Right, but there are many servers that request that you reuse the same dialog identifiers as the challenged transaction when you authenticate. /O

Re: [OpenSIPS-Users] New features in BLF and presence

Anca, This sounds like a very good start. My experience is that this is something that different people will want to use in different ways. Some people don't want phone calls to affect presence, where others say differently In the future, there might be need for a per-uri configuration

Re: [OpenSIPS-Users] opensips and asterisk

. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users --- * Olle E Johansson - o...@edvina.net

Re: [OpenSIPS-Users] DNS issues

list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users --- * Olle E Johansson - o...@edvina.net * Cell

Re: [OpenSIPS-Users] How to bill SIP session time correctly?

29 jul 2010 kl. 16.36 skrev Brett Nemeroff: On Thu, Jul 29, 2010 at 9:25 AM, Alejandro Recarey alexreca...@gmail.com wrote: When using a pure SIP solution like OpenSIPS, and session timers are not enough, how do you bill your customers? I have done a number of configurations where

Re: [OpenSIPS-Users] [OpenSIPS-Devel] Presence Subscriptions from External Domains

26 aug 2010 kl. 12.46 skrev Adrian Georgescu: Hello, I have a question maybe someone can help or comment. How can one protect in the real world against faking the identity of presence subscriptions originating from foreign domains? The scenario is: Once us...@domaina accepts

Re: [OpenSIPS-Users] Munin monitoring plugin

22 mar 2011 kl. 09.01 skrev Henning Holtschneider: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello everybody, I wrote a Munin monitoring plugin which collects data via the 'opensipsctl fifo get_statistics' command the other day. The plugin is available at

Re: [OpenSIPS-Users] NOTIFY authentication for UACs

14 feb 2012 kl. 17:40 skrev Matt Hamilton: Hi, I'm using Opensips to authenticate/authorize REGISTER and SUBSCRIBE messages for UACs before relaying them to Asterisk. secret field for sip peers is empty so Asterisk doesn't require authentication again for those. This works fine, but

Re: [OpenSIPS-Users] Call parking with loadbalancing

://lists.opensips.org/cgi-bin/mailman/listinfo/users --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo

Re: [OpenSIPS-Users] OpenSIPS Asterisk Integration in a new way

2 jul 2012 kl. 13:34 skrev aamir chougule: Wanted Scenario: Calls comes in to OpenSIPS server == Authentication Proxying part will be done by OpenSIPS == Call is relayed to Asterisk Server == Asterisk Server provides the IVR services to fetch the number from the customer == Asterisk

Re: [OpenSIPS-Users] OpenSIPS Asterisk Integration in a new way

2 jul 2012 kl. 16:08 skrev aamir chougule: Hi Olle, Thanks for the genuine suggestion and I really appreciate your answer. I understand the complications now after hearing the answers but is there a way before answering a call fetching the digits and then sending the digits back

Re: [OpenSIPS-Users] [OpenSIPS-Devel] Presence using RLS support in upcoming Blink release

21 sep 2012 kl. 15:21 skrev Adrian Georgescu a...@ag-projects.com: Hello, Here is some technical information about the support for Presence in upcoming Blink release. This is work in progress and if you have comments or suggestions you may attach them to the link below: I do look forward

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

Asterisk 11 has some early support for SIP over websockets, but that's far from being compatible with WebRTC. The standards for WebRTC are still evolving and require much more. It's a good step forward, but the ASterisk team is not there yet... :-) SIP over websockets is currently a draft that

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

31 okt 2012 kl. 18:17 skrev Ali Pey ali...@gmail.com: Which one sounds simpler? Having a new layer of proxies and extra hardware on different software packages with their own set of configurations, limitations and bugs than having WebSocket enabled on opensips and control your routing

Re: [OpenSIPS-Users] Issue with From domain coming from Asterisk

28 feb 2013 kl. 17:08 skrev Bogdan-Andrei Iancu bog...@opensips.org: Well, do not know much on Asterisk, so cannot comment :). What I wanted to point out is that we have the option to do it on opensips in an easy way - this will make quite irrelevant what Asterisk can do. In new versions of

Re: [OpenSIPS-Users] [asterisk-users] Register Free Opensips/Asterisk Integration

10 mar 2013 kl. 03:04 skrev Nick Khamis sym...@gmail.com: Hello Everyone, I have gone through a few really good tutorials from the OpenSIPS site, Asterisk resources etc.. The unanswered question (and final piece of our puzzle) is if it's possible to have a register free environment in an

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration

13 apr 2013 kl. 21:43 skrev Nick Khamis sym...@gmail.com: Make sure that you have host=dynamic on both the general level (i.e., sip.conf) and at the peer level (i.e., extensions, sip_peers in the database etc...) host=dynamic has no effect whatsoever in the general section of sip.conf. You

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration

13 apr 2013 kl. 22:08 skrev sjs205 sjs205.li...@gmail.com: Hello N, Thanks for getting back to me on this. This is one of the issues with this tutorial, one can not set the asterisk sip_peers to dynamic since the tutorial creates a view from the 'subscriber' table and uses the 'domain'

Re: [OpenSIPS-Users] Proxy everything to asterisk

17 apr 2013 kl. 13:04 skrev Bogdan-Andrei Iancu bog...@opensips.org: Hello Arthur, The OpenSIPS script allows you to implement whatever logic you want, so the answer is : yes, you can do that. Reuse the part for handling the sequential requests from the default opensips script and

Re: [OpenSIPS-Users] dns srv question

On 11 Feb 2014, at 14:47, Miha m...@softnet.si wrote: Hi Now I have one DNS SRV domain (sip.domain.com) which points to two A record inputs with different weight (sip1.sss.domain.com and sip2.sss.domain.com). If I add for sip.domain.com A record which will pont to the same IP as

Re: [OpenSIPS-Users] Encrypt and Decrypt sip signals

On 23 Jun 2014, at 09:12, kaushik parmar androidj...@gmail.com wrote: Hello All, My Android mobile SIP Dialer is sending Encrypted SIP messages Is it actually using S/MIME to decrypt on a per-message basis or do you mean it's using TLS as a transport? /O and i want to decrypt that SIP

Re: [OpenSIPS-Users] Dispatcher algorithm question

> On 29 Feb 2016, at 18:58, Gunjan Korlekar wrote: > > Hello, > > I have a dispatcher table with about 10 destinations and was using the > round-robin algorithm to route to destinations since we wanted to have an > even spread of traffic across the destinations. However I

Re: [OpenSIPS-Users] OpenSIPS in the market.

> On 18 Oct 2016, at 08:46, Bogdan-Andrei Iancu wrote: > > Hello Rodrigo, > > The questions your are asking are hard to answer and the Open Source world is > most of the times opaque when comes to who is using and how much is used. As > anyone can simply download and