I have been working on a project with asterisk and Kamailio. I would prefer using Kamailio because i have personally met with the developers and it has more active users and rapid developments. The developers are also very friendly and helpful. And w..
Hi.Im having problems with the Dial() application when I use full SIP account details in it.Im looking at the OReilly book https://www.amazon.co.uk/dp/1449332420 on page 135, where it says The Dial() application also allows you to connect to a rem..
Im 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 ca..
Hello.Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have match_auth_username=yes and have nothing in pjsip.I have a lot of endpoints and registrations on same SIP server. And its problem in pjsip now. Is not it?I request..
Hi I have installed Asterisk 11.13.1 on Fedora running in VirtualBox. The VB network interface is configured to use NAT. The host machine is Windows 7 and is connected to a SIP server using a VPN connection. I have configured ‘externaddr’, ‘localnet..
Hi!My telco is Deutsche Telekom and they got about 30 SIP servers right now. Currently Ive set up a template for incoming calls in sip.conf and added each SIP server by its IP address like this:[DTAG-in-1](DTAG-in-template)host!184.108.40.206…[DTAG-in-30](DTAG-in-template)host!220.127.116.11..
folks,What I trying to do here is exactly this: http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt599.htmlMy provider given me a Huawei modem which have 2 phone jacks on it, but instead of using it I rather redirect my P..
all, I have a project for the 3G related, AMR and AMR-WB support. Im using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab ..
We are facing an issue with asterisk in the case of call-Transfer scenarios. Our requirement is to identify whether an incoming call is a fresh incoming call or a Transferred call from some other clients. We have a setup, where in the asterisk1.6 ..
I have been using Cisco 7960s with Asterisk for years.I am trying get a 7961 working and have a problem.In my configuration, not all of my line appearances register to the same Asterisk SIP server.I have an Asterisk server at home and another at work..