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!188.8.131.52…[DTAG-in-30](DTAG-in-template)host!184.108.40.206..
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..
Maybe I am missing something or am a little blind at the moment, but I
didnt find out how asterisk can place a call on hold when acting as user
agent client to another SIP server.