To be sure you could setup a soft phone and see if the caller ID name comes in
correctly.
On Mar 10, 2015, at 8:41 AM, Jordan Cook - Gyron Networks
jordan.c...@gyron.net wrote:
Hi,
In my dialplan I have the following line.
same = n,Set(CALLERID(name)=Support)
I am expecting
Anyone know where it’s gone?.. Appears to have been down all day.
Steve
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On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes
steve-li...@geekinter.net wrote:
Anyone know where it’s gone?.. Appears to have been down all day.
The hamsters should be running in their wheels again now.
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Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW -
hello list,
i use chanspy with the code below
[app-chanspy]
exten = _007.,1,Macro(user-callerid,)
exten = _007.,n,Answer
exten = _007.,n,Authenticate()
exten = _007.,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = _007.,n,Hangup
i have a question related to chanspy
i have created extension from
Are the phones exposed to the internet (even using NAT)? If so there is a good
chance these calls are not coming through your PBX but are coming in direct
from someone, usually scammers.
Polycom has a config option to disable accepting calls from unknown devices.
No idea if Cisco has
On 3/11/15 12:48 PM, Salaheddine Elharit wrote:
hello list,
i use chanspy with the code below
[app-chanspy]
exten = _007.,1,Macro(user-callerid,)
exten = _007.,n,Answer
exten = _007.,n,Authenticate()
exten = _007.,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = _007.,n,Hangup
i have a question
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Hash: SHA1
Hi,
I made some tests with asterisk-13.2.0 and chan_pjsip this weekend
myself, and came to the same conclusion: some peerstatus events are
missing (eg. when contacts become unreachable / unavailable, IIRC), and
I could not find a way to get contacts
Hello.
Asterisk 13.2, PJSIP.
Problem: I do not get any AMI events when changing the status of the
contact.
When using chan_sip I got peerstatus event.
When using res_pjsip and devices (endpoint configuration) I got
peerstatus event.
When using res_pjsip.so and OUTBOUND REGISTRATION WITH
The main goal here is to be able to make a video call between two WebRTC
endpoints registered on asterisk 13 it is a feature that definitely asterisk 13
should support .
the problems that i faced with this is the following and i hope i could get an
advise here.
asterisk 13 vanilla version has
Please favorite the following bug and express desire to have wav49
VoiceMails play natively in Google Desktop Chrome HTML5.
https://code.google.com/p/chromium/issues/detail?id=465431
.WAV already plays in Android Chrome and Chromebook and others, but not the
desktop version.
--
Should I unload or rename the res_format_attr_h264.soH.264 Format
Attribute Module
The asterisk server 13.2.0 does not break anymore upon calls towards GXV3175
grandstream, however only downstream video displayed on the GXV3175 is very
slow (1 frame per 10 seconds)
This problem only
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