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0 answers
64 views

I am currently facing an issue with the call conference feature while creating a meeting. Basically, I am trying to create a meeting after both extensions have been successfully connected. After the ...
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0 answers
40 views

I am using 3CX with call forwarding to enable Single Number Reach (SNR) to my Cisco phone. The numbers are identical, but to route the calls through CUCM, my Cisco phone has a prefix 5 added. Setup ...
0 votes
1 answer
155 views

i got two grandstream ip phones that once in a while losing the pbx registration, only solution i came by is rebooting them (surprise surprise) my idea for the simplest way to do it is by using ssh ...
amit_m's user avatar
  • 1
2 votes
0 answers
186 views

I need your support that guide step by step. Cause there is no explanation in it's documentation. I have worked with Webrtc in web through following classes such as (RTCpeerconnection) but, in flutter ...
0 votes
1 answer
567 views

Our outbound calls through Twilio are all being dropped after 32 seconds (inbound calls are fine). The error we receive from Twilio is Error 32022 "Ack not received from your SIP endpoint." ...
bruce57's user avatar
1 vote
1 answer
3k views

I am trying to make calls between extensions 101 and 102 and is NOT going through. I have the following extensions.conf file [general] autofallthrough=no priorityjumping=yes static=yes writeprotect=no ...
1 vote
1 answer
217 views

I'm trying to get rid of controlling audio played via session.streamFile() in Freeswitch. For this I tried the 3rd example of this documentation. Almost everything here is working, but the DTMF 1 (...
0 votes
1 answer
1k views

I need to execute AGI scripts when following events occur: An incoming call (it is simple just call AGI() function). When a call is "Ringing" (I cannot figure it out!). <-- Problem, how ...
0 votes
2 answers
1k views

I have many SIP servers, but none of them have an external network. Can I use a server with an external network to proxy many SIP servers without an external network
曹姣月's user avatar
-1 votes
1 answer
325 views

We need to integrate our WebApp with Avaya PBX. Until now we used 3cx PBX and there is it very simple. We can launch the webclient passing the calling number in this way: https://blalbait:5001/...
-1 votes
1 answer
653 views

I used Asterisk PBX (Yeastar) s100 series . I need to save events with AMI in DB . It does not matter what the database is. Is there a solution under php? How do I do that?
0 votes
1 answer
887 views

Dear colleagues and visitors, I am trying to manage the Audiocodes Mediant VE SBC, in order to connect Microsoft Teams tenant to my PBX. I have already configured all the settings on SIP side (S+C), ...
0 votes
1 answer
616 views

I need help I have an asterisk and I need to play music on hold and a parallel execut macro same => n,Dial(SIP/${pbx}/#5147582218943,60,M(booms),m(mymoh)) M(booms) identify if he answered the call ...
1 vote
3 answers
1k views

The problem goes like this: value match: 218\d{3}(\d{4})@domain.com replace with 101円 to get 10 followed by last 4 digits for example 2181234567 would become 104567 value match: 332\d{3}(\d{4})@...
Mr. Popo's user avatar
0 votes
2 answers
1k views

I use Asterisk 16.5 and sip trunk. If known sip channel can i find uniqueid of call? Note: I want do it with Asterisk AMI actions and events.

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