Archives : April-2015
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all,I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts with the same service provides. We have 8 phone numbers in total.Incoming calls from the public are all correctly directed to appropriate office handsets. However, the disp..
Im maintaining the FreeBSD ports for asterisk(With madpilot@FreeBSD.org as identity). Heres a link to the asterisk13 port for your reference:http://www.freshports.org/net/asterisk13/I performed some tests with RC1 and am doing some final tests with ..
I am running asterisk 13.1.0In pjsip.conf, the endpoint section has an aors and an auth field.I can name the auth field anything I want.The key is to set the auth=field accordingly. However, when I try this with the aors field, it never works.It se..
The Asterisk Development Team has announced the release of Asterisk 11.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 11.17.0 resolves several issues reported by ..
-I am trying to decide if I have stumbled across a bug in PJSIP or I am just missing something. My Asterisk has two interfaces, an internal eth0 and an external eth1. In pjsip.conf, I define the following transports:[trusted]type=transport protocol=..
all,since asterisk 11 (1.6 was okay) failed the ReceiveFax-Application when it called about Dial and a Local-Channel.Directly from external to FaxReceive is no problem.Cut from cli:[…][Apr1 11:12:31] — Executing [s@macro-redirection:85] Dial(SIP/access-trunk-000000..
everybody, Ive a matter with the queue annoucement with the thereare, because if I put just one member in my configuration (member => SIP/2098), the ivr gave me that I was the firt or second in the next at the queue. But the problem is, if I add ..
I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. Im getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection.Howev..
I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. Im getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection.Howev..
HelloI use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom Germany. We have sometimes problems with incoming and outgoing calls. I hope I can explain it understandable.For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de..