Hi Matt,
Thanks for prompt reply.
Le 30/01/2019 à 22:38, Matthew Fredrickson a écrit :
> Right now, chan_pjsip does not properly handle tel: URIs. If you need
> them you might need to use chan_sip.
ok, I'm back on chan_sip, but I still do not see how I can send outgoing
calls with tel: uri
Hello I use this dial paln:
[o2-in]
exten => o2,1,Answer
exten => o2,n,Playback(hello-world)
exten => o2,n,Ringing
exten => o2,n,Dial(SIP/10/20/s@no-op,25,rt)
exten => o2,n,Playtones(425/1000,0/4000)
exten => o2,n,Wait(30)
exten => o2,n,Hangup()
All is fine. Hello world is Playback and I hear a
With softphone I mean linphone csipsimple or whatever.
How should a dialplan lokks like?
On 31.01.19 11:26, Antony Stone wrote:
> On Thursday 31 January 2019 at 10:59:01, basti wrote:
>
>> Hello I use this dial paln:
>>
>> [o2-in]
>> exten => o2,1,Answer
>> exten => o2,n,Playback(hello-world)
On 7/3/18 3:57 PM, D'Arcy Cain wrote:
> On 2018-06-13 07:45 AM, D'Arcy Cain wrote:
>> On 2018-06-13 07:20 AM, James Cloos wrote:
D'Arcy Cain writes:
>>>
> Ie after both sides select t38, until they agree on the t38 terms.
>>>
OK, so does that mean that setting it to 25000 should
On Thursday 31 January 2019 at 11:36:05, basti wrote:
> With softphone I mean linphone csipsimple or whatever.
I know what you mean by "a softphone"; I just wasn't sure how you were calling
your softphone and what you were saying (didn't) happen.
> How should a dialplan lokks like?
Have you
On Thursday 31 January 2019 at 10:59:01, basti wrote:
> Hello I use this dial paln:
>
> [o2-in]
> exten => o2,1,Answer
> exten => o2,n,Playback(hello-world)
> exten => o2,n,Ringing
> exten => o2,n,Dial(SIP/10/20/s@no-op,25,rt)
> exten => o2,n,Playtones(425/1000,0/4000)
> exten => o2,n,Wait(30)
>
Hi list,
Using Asterisk 16.1.1, with PJSIP, I'm asked to build a SIP trunk to a
system that uses exclusively tel: uri on inbound and outbound calls. I
could not find documentation or sample config about tel:uri. Is this
doable? If not possible with PJSIP, is chan_sip a better option? Any
pointer
On Thu, Jan 31, 2019, 9:24 AM Jean-Denis Girard Hi list,
>
> Using Asterisk 16.1.1, with PJSIP, I'm asked to build a SIP trunk to a
> system that uses exclusively tel: uri on inbound and outbound calls. I
> could not find documentation or sample config about tel:uri. Is this
> doable? If not
Asterisk 16.1
This statement appears in the features.conf doc: "Note that the DTMF
features listed below only work when two channels have answered and are
bridged together. They can not be used while the remote party is ringing
or in progress. If you require this feature you can use
Thanks antony. Now it works.
Just for the docu:
[o2-in]
exten => o2,1,Ringing
exten => o2,n,Playtones(425/1000,0/4000)
exten => o2,n,Dial(SIP/10/20/s@no-op,130,rt)
exten => o2,n,StopPlaytones()
exten => o2,n,Hangup()
[no-op]; just hang up
exten => s,1,Hangup(130)
On 31.01.19 11:40, Antony
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