Re: [asterisk-users] Pass through registration / proxy

Hi You could use kamailio +asterisk On Tue, Apr 10, 2018, 9:25 PM Telium Technical Support wrote: > I need to create a SIP proxy to be placed in front of a legacy PBX. When > a phone registers with the proxy, I would like Asterisk to register with > the PBX behind it. (To

[asterisk-users] Pass through registration / proxy

I need to create a SIP proxy to be placed in front of a legacy PBX. When a phone registers with the proxy, I would like Asterisk to register with the PBX behind it. (To tell the PBX to send calls to the proxy and then to the SIP phone). Can I use Asterisk to create a proxy like this? Is

Re: [asterisk-users] PJSip CallerID Question

On 4/7/2018 5:50 AM, Daniel Tryba wrote: On Fri, Apr 06, 2018 at 02:27:31PM -0500, Brent Davidson wrote: I have multiple Asterisk instances set up in different locations and would like to modify the callerID of inbound calls to identify which instance the call is coming from.  I knew how to do

Re: [asterisk-users] Asterisk behind NAT Early Media Video

I just noticed, the calling device isn't even sending the early media video stream. It just sends an early media audio stream. Is there propably a change in the signaling needed? (On another P2P SIP Server the early media video works.) 2018-04-10 12:29 GMT+02:00 Benjamin Marty

Re: [asterisk-users] withheld caller id

so any ideas, please? On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux wrote: > after adding the ww: > root@Pbx: /etc/asterisk $ asterisk -rvvv > Asterisk 11.25.3, Copyright (C) 1999 - 2013 D == Using SIP RTP TOS bits > 184 > == Using SIP RTP CoS mark 5--

Re: [asterisk-users] withheld caller id

after adding the ww: root@Pbx: /etc/asterisk $ asterisk -rvvv Asterisk 11.25.3, Copyright (C) 1999 - 2013 D == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5-- Executing [9211123456@AllCalls:1] Goto("SIP/500-0003", "DefaultPlan,9211123456,1") in new stack

Re: [asterisk-users] withheld caller id

>>> My suggestion would be to add a pause or two before dialing the phone number Looks like using w for a pause is no longer supported. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

Re: [asterisk-users] withheld caller id

>>> > exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT) My suggestion would be to add a pause or two before dialing the phone number exten => _9X.,1,Dial(Dongle/dongle800/#31#ww${EXTEN:1},120,KT) D(digits): After the called party answers, send digits as a DTMF stream, then connect

Re: [asterisk-users] Asterisk behind NAT Early Media Video

Hi Florian I already have the external_media_address set in the PJSIP setup. Also the external_signaling_address is set to the Public IP. If I make a call from an Early Media (video) capable device to an Early Media capable device (also video) the Early Media audio works perfectly. But no video.

Re: [asterisk-users] withheld caller id

On 2018-04-10 10:19, Atux Atux wrote: exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT) exten => _9X.,n,Hangup(${HANGUPCAUSE}) What am i doing wrong in asterisk? unless i'm missing something your config looks OK. Do you have any logs / debugs of what number is actually being

Re: [asterisk-users] withheld caller id

thanks a lot for the reply. i thought of that and i did try to send *exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT)exten => _9X.,n,Hangup(${HANGUPCAUSE})* but the provider replies back that it is a wrong number. Then i inserted the sim to an ordinary mobile phone and dialed #31#

Re: [asterisk-users] withheld caller id

On 2018-04-10 08:46, Atux Atux wrote: 9+#31#+destination_number. Unfortunately, zoiper did stop on 9#31# and it dialled one of my recent numbers. The same result happened with haven't used zoiper at all, so can't comment on its features of parsing numbers. I'd recommend 'hiding' this

[asterisk-users] withheld caller id

Hi. I am running asterisk 11 and i have usb 3g dongles to make my gsm calls with the following config in extensions.conf exten => _9X.,1,Dial(Dongle/dongle800/${EXTEN:1},120,KT) exten => _9X.,n,Hangup(${HANGUPCAUSE}) By dialing 9 it opens the dongle to make a call. I would like to restrict my

Re: [asterisk-users] Asterisk behind NAT Early Media Video

I applied the patch to my Asterisk 13.20. But it seems that it still doesn't forward the early media video stream. Do I need to put something special into the extensions.conf? I basically just make a Dial. The calling Client sends the 183 protocol. [public] exten => 6001,1,Dial(SIP/${EXTEN})

Re: [asterisk-users] Asterisk behind NAT Early Media Video

Hi Benjamin! You're obviously using a similar scenario that I have in place for testing. I initially had issues with early media (not only video also audio) as well in that scenario. What I had to do was to additionally set external_media_address= in pjsip.conf Also, as I wrote the patch for