Displaying 6 results from an estimated 6 matches for "12125551212".
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12025551212
2007 Dec 21
0
Incoming CID change
Quick and dirty too hungover to think/search... ;) Sorry list
AsteriskBoxA -> SBC/PSTN voodoo -> world wide Interweb -> AsteriskBoxB
Account on BoxA UsernameJohn<12125551212>
Account on BoxB UsernameJohn<102>
They're both the same users, had to do some funky trunking (managed
firewall provider is playing not in my backyard games)...
So anyway, I need to take specific calls from BoxA with the CID of
UsernameJohn<12125551212> and switch it to Usernam...
2004 Jun 29
1
* Busy-Redial ??
I was wondering if anyone knew of a way to create a busy-redial feature in
the * dialplan? For example, you try to call 12125551212 but the number is
busy, so you hang up and dial *XX12125551212 and hangup again, then * would
continue to retry calling the number until either it rings or a timeout is
reached, if it rings * then calls back the exten that made the *XX call and
bridges the two channels (maybe even with a distinctiv...
2009 May 27
1
setting CDR values on failed calls
Hi All,
I am relatively new to Asterisk. I have CDR enabled and successfully writing
to MS SQL server. In my cdr table I am setting the userfield value with a
line in my dialplan. If a call is placed to an invalid number (e.g.
12125551212), I see a cdr record created, however, my userfield value never
gets set since the call never made it into the context of my dialplan. I am
using AMI with the Originate command to invoke the call. How can I set this
value before the call is actually passed to my voip provider (of whom
quickly respo...
2005 Jan 02
1
Call Queue Question
...oth numbers at the same time and wait for a #
confirmation before bridging the call. This will be repeated with 1..
2 etc for different family members....
If I use AddQueueMember to add the members to the queue - or define
the members directly in queues.conf (like member =>
IAX2/username@NuFone/12125551212) - I do not get the ability to use #
to confirm before the call is bridged. Every other feature works
perfectly though. It seems the only way I can get the # is to use
AgentCallbackLogin.
Problem is - I don't want anyone to have to do anything to log in. The
agents should always be on... Can...
2009 Jun 26
1
Calls dropping
Hi,
I am using a call file formated like this:
Channel: local/12125557891 at outbound/n
Callerid: 12125551212
Context: detect
Extension: s
Priority: 1
This sends the call into the dialplan at the [outbound] context. In
[outbound], I have:
[outbound]
exten => _1.,1,Dial(SIP/${EXTEN}@flowroute,43)
If the call is answered, it move on to the [detect] context.
When using this method, it appears that the cal...
2006 Jan 26
5
Skype-to-Asterisk(SIP): progress
I'm sitting in the Emerging Telephony Conference, so this seems a
particularly apt place to pre-announce this...
I've wanted to be able to gateway calls between Skype and Asterisk
for a while, which of course would require some type of protocol
converter (IAX or SIP to Skype, probably.) This of course is
directly not in Skype's interest, since they would like to keep the
network