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 => _[a-zA-Z].,1,S
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
>
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 unmana
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:
http://secu
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 wrote:
>
>> The strange thing is its only sometimes my dial string is as follows
>>
>> exten => s,1, Dial (SIP/200,, tT)
>>
>> For that pa
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 DTM
On Mon, Feb 16, 2015 at 10:13 AM, 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.
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.
Sent from Samsung Mobile
Original message Fr
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 hopin
> 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
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 sensitivi
On 16 February 2015 at 11:49, Igor Pavlov 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
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 r
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