Hello, We recently upgraded to Asterisk 1.8/DAHDI 2.4/WANPipe 3.5.16. This system is connected to a PRI where the provider requires long distance codes. Normally, you dial, see progress and hear a tone (call is still "unanswered" at this point), enter your code, and it starts ringing as a normal call. Since upgrading, we aren't getting the tone anymore and aren't able to enter any digits at the "right place". The only workaround we've seen is to use something like below (which works). Does anyone know what changed between 1.4 and 1.8 in regards to early audio (both hearing it and…
Hi I have a problem that I can't pass.
I have asterisk and cisco 7970 phones with 8.0.3 sip firmware.
I registered two extensions: Line1: 260
Line2: 160 Regardless of which extension I call, always Line 1 on cisco is blinking.
This makes impossible to recognize which extension is calling.
Also, I've set Line 2 to be automatically answered with speaker phone
(intercom). Even though I call extension 160 from Line 2 it is never
automatically answered. Can anybody help me with this issue? I've been searching Internet to find
clue on what…
A few days I have problems connecting to the Internet on my house and
since then my local SIP extensions are no longer registered against the
local Asterisk server. I'm using Asterisk 126.96.36.199. I was researching on the Internet and I
found that it can be related to a bug of chan_sip, can it be? In this
case, is there a possible workaround? Thanks in advance for your reply. Regards,
I've been experiencing trouble with my DAHDI channels for some time and have
finally decided to try and resolve the issue. Essentially, the problem I am having is that when a call comes in, and my
DAHDI phones therefore ring, Asterisk thinks that one of the handsets has
picked up to answer the incoming call - whereas in actual fact it is still
on hook. The call then gets instantly dropped (the phone is on-hook, after
all), and the caller has to redial. Sample log (this is an incoming call from SIP/5555, that was…
I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out those settings and included them below. I was using a Linksys PAP2T-NA with firmware version 3.1.15(LS), but I would assume that these settings would be similar on…
Please help me in configuring asterisk for the below scenario:
I want to make calls to some mobile users then
My query is simple, my agent attends an incoming call from Queue and I
want my agent to play a recording and request the user to input 4 digit
input from his keypad and that input be recorded somewhere. Now I want to know technically if thats possible or not with asterisk? If
its possible than how? Regards, Abeer
When Mobile user attends an incoming call from asterisk then I want to play
i have this configuration , An Asterisk server connected to my private LAN 192.168.10.0/24 when i do port forwarding for port 5060 so that i make a call from Internet into Asterisk wireshark show the message "destintion port unrechable"
i configured sip.conf for "nat=yes" and "qualify=yes" and "externip="my public IP"
did i forget some other ports to forward otherthan 5060?
did i forget any other configurations?
i even tried the "virtual server" function in my D-Link 2640U ADSL router with no hope
appreciate your help