We are looking for professional help to get this issue working: ARI to snoop on call Asterisk APIs We’re using the GitHub - asterisk/asterisk-external-media project as a basis for doing real-time call transcription. It works fine if the dialstring goes to a context which does ChanSpy on the call to be transcribed. This is a bit messy though, and we would like to use ARI to snoopChannel on the call to be transcribed instead. Presumably the following part of lib/ari-controller.js from that project needs to change, but I’m not sure how it should be changed to use snoopChannel. Can anyone point… I ..read more

Hello guy’s, i make a call to an external number but neither party can hear the other. after 30s the call is automatically disconnected. I have this in the logs: Disconnecting channel for lack of audio RTP activity in 30 seconds //rtp_custum.conf [general] rtpstart=10000 rtpend=20000 icesupport=true stunaddr=stun.l.google.com:19302 //pjsip.transport.conf [0.0.0.0-udp] type=transport protocol=udp bind=0.0.0.0:5060 allow_reload=no tos=cs3 cos=3 local_net=10.38.43.0/24 [0.0.0.0-wss] type=transport protocol=wss bind=0.0.0.0 allow_reload=no tos=cs3 cos=3 local_net=10.38.43.0/24 //pjsip.endpoin ..read more

The Asterisk Development Team would like to announce the release of asterisk-21.3.0. The release artifacts are available for immediate download at GitHub Release Asterisk Release 21.3.0 · asterisk/asterisk The Asterisk Development Team would like to announce the release of asterisk-21.3.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/... and https://downloads.asterisk.org/pub/telephony/asterisk Repository: GitHub - asterisk/asterisk: The official Asterisk Project repository. Tag: 21.3.0 This release resolves issues reported by ..read more

The Asterisk Development Team would like to announce the release of asterisk-20.8.0. The release artifacts are available for immediate download at GitHub Release Asterisk Release 20.8.0 · asterisk/asterisk The Asterisk Development Team would like to announce the release of asterisk-20.8.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/... and https://downloads.asterisk.org/pub/telephony/asterisk Repository: GitHub - asterisk/asterisk: The official Asterisk Project repository. Tag: 20.8.0 This release resolves issues reported by ..read more

The Asterisk Development Team would like to announce the release of asterisk-18.23.0. The release artifacts are available for immediate download at GitHub Release Asterisk Release 18.23.0 · asterisk/asterisk The Asterisk Development Team would like to announce the release of asterisk-18.23.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag... and https://downloads.asterisk.org/pub/telephony/asterisk Repository: GitHub - asterisk/asterisk: The official Asterisk Project repository. Tag: 18.23.0 This release resolves issues reported ..read more

Could you guide how to configure outgoing context in Asterisk for making calls to real phone numbers? I’m interested in setting up Asterisk to make outbound calls, and I’d like to know what configurations are needed to achieve this effectively. Additionally, I’m exploring alternative methods to configure the extension.conf file in Asterisk. Are there any other approaches or best practices I should consider? 1 post - 1 participant Read full topic ..read more

def stasis_start_cb(channel_obj, ev): “”“Handler for StasisStart”“” channel = channel_obj.get(‘channel’) channel_name = channel.json.get(‘name’) args = ev.get(‘args’) if not args: print "Error: {} didn't provide any arguments!".format(channel_name) return if args and args[0] != 'inbound': # Only handle inbound channels here return if len(args) != 2: print "Error: {} didn't tell us who to dial".format(channel_name) channel.hangup() return user_id = channel.json.get('caller').get('number') destination = args[1] # Check user credit credit_balance = check_user_cred ..read more

We’re using the GitHub - asterisk/asterisk-external-media project as a basis for doing real-time call transcription. It works fine if the dialstring goes to a context which does ChanSpy on the call to be transcribed. This is a bit messy though, and we would like to use ARI to snoopChannel on the call to be transcribed instead. Presumably the following part of lib/ari-controller.js from that project needs to change, but I’m not sure how it should be changed to use snoopChannel. Can anyone point me in the right direction? Thank you very much in advance. // Call the phone or confbrid ..read more

i have asterisk 11.25 version system behind firewall, and there are two external IP ,one is pulic ,one is vpn set in firewall, and the two IP had done NAT to the asterisk’s 5060 port. my question is i can set only one external_ip ,when i set public ip in system, the vpn users make phone and can not hear voice. is there any thing i can do to handle the problem? 1 post - 1 participant Read full topic ..read more

Hello. We have a call flow application that works with Asterisk and the ARI. I’d like to be able to create a mixing bridge once the inbound channel is created, and record the bridge. When the outbound channel is added to the mixing bridge, the recording is stopped. Is this a known situation, when going from 1 to 2 channels on a mixing bridge, that the existing recording will end? We also create inbound and outbound snoop channels and record those, so in total we’ll have 3 recordings at the end of the call. Thank you Steve 1 post - 1 participant Read full topic ..read more