The strange thing is its only sometimes my dial string is as follows
exten = s,1, Dial (SIP/200,, tT)
For that particular route but obviously s,3 as have Ringing () first etc.
After she pushes ## 6 times it will go thru sometimes.
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div Original message
On 16/2/15 4:13 pm, Andrew Colin wrote:
The strange thing is its only sometimes my dial string is as follows
exten = s,1, Dial (SIP/200,, tT)
For that particular route but obviously s,3 as have Ringing () first etc.
After she pushes ## 6 times it will go thru sometimes.
Are you sure it's a
RFC2833
The strange thing is how asterisk is not registering she has pushed ## on
those Rare occiasions
On Mon, Feb 16, 2015 at 10:13 AM, Andrew Colin and...@vsave.co.za wrote:
The strange thing is its only sometimes my dial string is as follows
exten = s,1, Dial (SIP/200,, tT)
For that
On Mon, Feb 16, 2015 at 10:13 AM, Andrew Colin and...@vsave.co.za wrote:
The strange thing is its only sometimes my dial string is as follows
exten = s,1, Dial (SIP/200,, tT)
For that particular route but obviously s,3 as have Ringing () first etc.
After she pushes ## 6 times it will go
Hello, I am working with 1.8.32.2 which I have patched with t38-gateway and
PRACK. t38 is tested and working fine with Zoiper client but I can't get the
t.38 software from Biscom (FAXCOM) to talk. In my first attempts I found
FAXCOM announces that it supports 100rel so I added the PRACK patch
It looks as if that is more of a question/issue with your router, rather than
Asterisk.
I have SIP devices working on my LAN, all hardwired, and have no need to open
any ports or have the router address SIP in any way
My switch is not managed, and the router ports on the LAN side are all
I'm reading the O'Reilly Asterisk the definitive guide, 4th ed, with a
starfish on it. In some ways, astonishing that it's not really that
definitive, it's more general -- and it only clocks in at one ream of
paper!
In any event, I'm having some port problems on my home network:
On Mon, 16 Feb 2015 16:12:04 -0500, John Novack wrote:
It looks as if that is more of a question/issue with your router, rather
than Asterisk.
I have SIP devices working on my LAN, all hardwired, and have no need to
open any ports or have the router address SIP in any way My switch is
not
Hi,
I copied a setup from an older 1.8.5 installation to an 11.15 installation, and
I'm having problems getting call files to work. Here is the extension setup
I'm using:
[outbound-swift]
exten = _[a-zA-Z].,1,Answer
exten = _[a-zA-Z].,n,Playback(AAA/check_ip_failure)
;exten =
On 16 February 2015 at 11:49, Igor Pavlov i...@izhnet.ru wrote:
Hi, list.
We have a problem with loss peers after ‘sip reload’, our configuration:
Asterisk 11.6-cert1, SIP realtime peers, sip.conf:
- rtcachefriends=yes
- rtsavesysname=yes
- rtupdate=yes
- rtautoclear=yes
When we
Hi Guys
We have a client running on a polycom vvx400 IP phone on our asterisk
1.8.18 system
The issue we have is the switchboard lady uses ## to transfer calls but
sometimes it just does not work and just plays the DTMF tone to the
calling party.
Is there any way to adjust the
Hi Guys
We have a client running on a polycom vvx400 IP phone on our
asterisk 1.8.18 system
The issue we have is the switchboard lady uses ## to transfer calls
but sometimes it just does not work and just plays the DTMF tone to
the calling party.
Is there any way to adjust the
Hi, list.
We have a problem with loss peers after 'sip reload', our configuration:
Asterisk 11.6-cert1, SIP realtime peers, sip.conf:
- rtcachefriends=yes
- rtsavesysname=yes
- rtupdate=yes
- rtautoclear=yes
When we do 'sip reload' , peers are removing from available.
Before `sip
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