For what it's worth I'm also using HylaFAX 6.0.3 with IAXmodem to talk to my
1.6.0.17 box to send faxes across my SIP provider (they only support SIP at
this time). I can't really speak to the reliability for high-volume loads, but
I haven't had any problems with the dozen or so that I've sent
Hi All,
I'm running 1.6.0.17 and wanted to adjust the output from the command sip show
channels. When there is a current call in progress the User/ANR field shows 10
numbers. The problem I'm having is that it is including a 1 such as 1949555121
which is truncating the last number. Is there a
Hi Jerry,
I use the built-in function queue_member
http://www.asterisk.org/docs/asterisk/trunk/functions/queue_member?type=functionsvalue=QUEUE_MEMBER
and check with a GotoIf statement to check if the number is equal to zero. If
it is not I send the call to the queue, if it is I pass the
Hello all,
Do you know if it IS possible to use multiple lines/extensions on SIP with a
Cisco 7960 or other phone models? My boss wanted to have 1 physical phone but
have it register to a couple of different extensions, then use different
ringtones to identify which line was ringing when a
Hello Everyone,
I'm looking for help/ideas on how to do the following:
I have a couple of people out of many (the couple of people randomly change)
who log into an on-call queue. A call comes in and it rings the on-call
extensions, but no one answers. I would like the call to then try the
] Queues
It should be realistic, but have you considered just using followme to add the
cell phones to the queue list?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry
Sent: Monday, November 16, 2009 3:25 PM
this instead of having to code a lot of dialplan, but that’s just me…
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry
Sent: Monday, November 16, 2009 3:50 PM
To: Asterisk Users Mailing List - Non-Commercial
