Try to set fromuser=number in your sip provider peer configuration
On Tue, May 8, 2018, 11:05 PM Jeff LaCoursiere wrote:
>
> Thats till doesn't change the SIP header. Basically they want to send a
> RE INVITE and authenticate my DID number. But my DID number does not
On Tue, May 8, 2018, at 5:48 PM, Dovid Bender wrote:
> Hi,
>
> It is my understanding that while Hebrew is supported by Asterisk the sound
> files are not shipped with it as they are no longer being maintained. Can
> anyone advise on what's needed to maintain a specific sound package? We are
>
Hi,
It is my understanding that while Hebrew is supported by Asterisk the sound
files are not shipped with it as they are no longer being maintained. Can
anyone advise on what's needed to maintain a specific sound package? We are
considering to support Hebrew and possibly Yiddish.
--
Thats till doesn't change the SIP header. Basically they want to send a
RE INVITE and authenticate my DID number. But my DID number does not
have a peer or user entry in sip.conf. Perhaps I am answering my own
question, but is that the only way this is going to work?
Thanks,
j
On
try adding a + sign for the number
same => n,Set(CALLERID(all)=17864089672 <+17864089672>)
On Tue, May 8, 2018, 8:51 PM Jeff LaCoursiere wrote:
>
> I *am* doing that, as I assumed it would be required just for the 911
> mapping we have provided, but that doesn't change
I *am* doing that, as I assumed it would be required just for the 911
mapping we have provided, but that doesn't change the SIP header.
Cheers,
j
On 05/08/2018 02:41 PM, Khalil Khamlichi wrote:
try setting the callerid with
same => n,Set(CALLERID(all)=17864089672 <17864089672>)
ofcourse
try setting the callerid with
same => n,Set(CALLERID(all)=17864089672 <17864089672>)
ofcourse for each customer you will need to provide his own did.
On Tue, May 8, 2018, 8:37 PM Jeff LaCoursiere wrote:
> Hi,
>
> We have been using Voxbone for some time for origination,
Hi,
We have been using Voxbone for some time for origination, and they now
offer E911 services. We are trying to set this up and having trouble
meeting their authentication requirements.
I setup a peer as I normally would, with user/pass as they supplied
("lacoursj", "pass"), but my calls
Use a script to redirect the ringing call into an extension that returns the
proper sip result, and hangup.
You could also add logic to alert or log that call.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, May
Hi,
I'm looking for a way to reject a call remotely using the Asterisk
dialplan.
For example, phone A is ringing - I'm at the other end of the room next to
phone B, and I want to reject the call to Phone A by dialing an extension.
I'm basically trying to reproduce the Polycom "reject"
Hi all
I need to pass a parameter in a thread-safe manner to the Queue pickup
macro. This is to know when (and who) picked up an incoming call to a queue
and log that to my back-office system with a CURL to a HTTP endpoint.
However, the Queue application does not appear to allow passing of
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