similar to: Kamallio registration

2017-02-16 14:27 GMT+01:00 Joshua Colp <jcolp at digium.com>: > On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote: > > Hello, > > > > I'm currently testing a so-called VQ RTCP-XR feature from a a SIP > > hardphone. > > > > When a phone has enabled this feature, it would send a SIP PUBLISH to its > > SIP Server letting this server dispatch to

Hello, I'm currently testing a so-called VQ RTCP-XR feature from a a SIP hardphone. When a phone has enabled this feature, it would send a SIP PUBLISH to its SIP Server letting this server dispatch to whatever is needs to. These messages are sent during calls but may also be sent when a call is over. At the moment, I'm using Asterisk to serve these SIP phones so my Asterisk box

From: Matthew Jordan <mjordan at digium.com> > > > >> If the INVITE request is not shown in the CLI with 'pjsip set logger > >> on', then Asterisk is not actually receiving the request. > >> > >> Does a pcap show the message being sent to the correct IP/port? If you > >> change the transports to bind to port 5060, does that change

Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username

For those that were interested I have attached the kamailio.cfg which we have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the following yum packages: kamailio.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-auth-ephemeral.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-bdb.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms

Can anyone recommend a good WebRTC phone to use with Asterisk? I do not mind if it is commercial or open source. Customers are starting to ask for web solutions and we need to start testing. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

Hi all Have recently watched Matt Jordan's session on Kamailio World 2014 On slides 26-29 of his presentation (http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf) he speaks about a (completely new, for me at least) approach to build scalable telephony systems, using N instances of Kamailio and N instances of Asterisk Are there any

Hi All; I am looking for a good Outbound Dialer and to be practical with possibility to do modification on it, the outbound dialer should send the calls to the agent when the agent is logged in as long the agent is belong to the queue (or let us say the skill group of this campaign). Any one can guide me? If I can build this using the AMI, so I appreciate if anyone did it before me so I can use

Do you know if this can be achieved with the standard sip stack in asterisk? Kind Regards Andrew Colin Converged Telecoms (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Switchboard: +27 (0)10?591 4600 Email:? andrew at convergedgroup.net Web: ?http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This

Hi all, I saw Matt Jordan's recent Kamailio world talk and was interested in the idea he proposed of stripping out authentication and registration from asterisk and letting Kamailio handle it. All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding registrations to asterisk. In order to do what Matt suggested would I be correct in assuming I would have to use the

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

Works for me.... Thanks, Hristo Benev -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexandru Oniciuc Sent: Monday, May 17, 2010 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration

I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration (MySQL database) so that kamailio authenticates and then forwards the registration to asterisk on localhost. The setup calls for asterisk to be

Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x

Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw

Hello, I was wondering if there is a tool that can create the realtime database structure for latest Asterisk version or a web resource/file containing the sql scripts. Hope I haven't missed obvious things, I had no luck searching on the web, in the wiki I found few pages with bits of sql or table structures, like:

Hello, I put together a new tutorial about asterisk realtime integration with kamailio (openser). This time the database used is the one of asterisk, also call routing logic is controlled by asterisk, here is the link: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Practically is an easier way to scale starting from existing asterisk installations. The other

Hi, I want to have Kamailio in front of one or more Asterisk boxes. I don't think it is necessary for Kamailio and Asterisk to register with one another. I'd like for PJSIP to recognise Kamailio by its IP address. I have two boxes, both have public IP addresses, they also have private IP addresses and can communicate with each other. I have a Snom phone accessing Kamailio via its

I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been heavily modified. Currently asterisk runs on localhost and only listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to

Hello all, I've been testing a Kamailio Asterisk Realtime integration, and found a strange situation. My problem is that when using the integration, everything seems ok but Asterisk does not see the clients as registered. Kamailio and the clients report registered clients. Also calls fail. In Asterisk cli sip show peers shows nothing but for example realtime load sipusers name 660 shows the