similar to: Relay Register

Displaying 20 results from an estimated 1000 matches similar to: "Relay Register"

2009 Mar 23
0
Issue with no change of SIP call ID
Good afternoon everybody. I first would like you to excuse me for my english. I have an issue with a SIP call ID which is not changed in the call configuration described bellow : I have an Asterisk Server A using only SIP protocol. Behind A there are 2 distant clients (using softphone X-lite) C1 and C2 and a proxy server OpenSIPS (ex OpenSER) P. The idea is that when C1 want to call C2,
2009 May 15
1
Spiral SIP Request problem
Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the
2015 Nov 20
2
SIP calls dropping at 15 minutes
I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP
2020 Oct 28
4
PJSIP tight loop on auth failure
Hi, We're using Asterisk 13.17.0 with PJSIP 2.8 bundled. I've found an issue when Asterisk tries to make a SIP call out using auth, but has the wrong credentials and keeps getting returned a SIP 407, in this example to an OpenSIPs server requiring user auth. Basically this happens: 1. Asterisk sends plain INVITE to OpenSIPs 2. OpenSIPs responds with SIP 407 auth required with a
2013 Mar 10
1
Register Free Opensips/Asterisk Integration
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 OpenSIPS/Asterisk integration. Most approaches have OpenSIPS relay the UA's REGISTER request to Asterisk which has "host=dynamic" set for the
2010 Apr 19
2
OpenSIPS with Asterisk Backend
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2011 Mar 04
3
OT: OpenSIPS vs Kamailio -- which do you use and why?
I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :)
2009 Mar 20
3
OpenSIPS on CentOS
Hello, I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via
2012 Jan 09
1
Asterisk as register server through OpenSIPS
Hi all, I've been trying to register a SIP user agent to an Asterisk server using OpenSIPS as SIP router. The functionality is working fine. However, Asterisk uses the IP address of the OpenSIPS server as the peer IP address. How can I use the original IP address of the peer without changing the peer's nat=yes? I appreciate any kind of help. Thanks! Regards, Ronald -------------- next
2016 Jul 05
2
OpenSIPS or Kamailio based fronting for Asterisk?
Hello, I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for either. Is there any place I can get a very simple HA configuration (telling me where the config files are, for starters, is a good thing) for OpenSIPS or Kamailio with the following features: (a) Support an arbitrarily large number of
2009 Feb 26
1
incoming call problem
Dear All, I have created an inbound context in SIP .conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with OpenSIPS and cal
2009 Mar 01
1
Help T.38
Dear All, I have created an inbound context in sip.conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes under General context...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with
2009 Feb 11
2
OPTIONS packets
Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: 1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060 2. OPTIONS sip:OPENSIPS_IP
2009 Dec 30
2
multiple instances of asterisk on same machine
hi all, I have a little problem.... I'm using asterisk with opensips as opensips dispatches calls to asterisk. I have to use multiple asterisk servers but since for the time being im using 1 machine for testing i want to run different instances of asterisk running on 1 pc listening to different ports. Can someone please guide me how to do this? I'll be very thankful Regards, Saeed
2010 Mar 17
3
SIP codec negotiation / manipulation
We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second
2009 Aug 13
1
Thoughts
Hi, After using a little bit oVirt, here's my thoughts: On the Network part : * In order to start the bridges, you have to reboot the node. It should be good to create the bridges "on the fly". * The VLAN configuration seems not to work (you can create it but it's impossible to assign it to an interface...). You can also assign this kind of interface on a opennet On the VM part:
2009 Aug 05
1
iSCSI questions and VM Creation questions
Hi, I finally managed to have one node working. I added a iSCSI LUN (ok if selinux is set to 0) and wanted to create my first VM. So I created a VM and had some issues: * I can choose boot from HD but I'm just able to choose the LUN, not a specific size in the LUN like you can do in vmware ESX. So, do I have to create a LUN per VM? If yes, it's quite complicated... * If I choose boot from
2009 Mar 20
1
Asterisk + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk
Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg. Contact" is the IP
2009 Feb 24
1
Incoming call
Dera All, I have the following scenario, A customer dial a DID number...The call is routed to a PSTN GW that send the call to asterisk... On asterisk I created an AGI Script that send the call to an extension registered on OpenSIPS server... The extension is ringing successfully, but as soon as I accept the call on OpenSIPS side the call is hangd up... I checked rhe SIP debug and it seems that I
2008 Dec 11
2
Multi OS boot and shared secret trouble
Hello We are facing a boring trouble with multi OB boot machines that access to our samba server. All machines authenticate to samba PDC ( linux debian etch + windows ) It appears when machines windows's bootup the shared secret is changed by windows if we reboot the same machine under Linux it cannot authenticate anymore because Linux does NOT change the shared secret ... The machines