Re: [asterisk-users] A bit OT - Configure GoIP for Asterisk

2017-10-02 Thread Antony Stone
On Monday 02 October 2017 at 20:58:33, Steve Edwards wrote: > I recently received a GoIP-32 for a client project -- primarily outbound > calling. > > How should a GoIP be configured for Asterisk? Have you tried http://www.hybertone.com/en/solutionsClass.asp?Id=78 Antony. -- Police have

[asterisk-users] Connect Two Existing Channels and Stop Listening

2017-10-02 Thread Joseph Smith
Hello all, In my scenario I have two channels connected to Asterisk and in a stasis app. I can put them both in a bridge and audio between them works as expected. However, I would like to free up the resource and no longer have Asterisk involved in the call if possible. I'm currently

Re: [asterisk-users] PJSIP add header not working

2017-10-02 Thread Andre Gronwald
Thanks all for the help, I got a step ahead. But in this scenario I am not able to deliver call-id of call-leg a to call-leg b. Extension A is going to make an outbound trunk call: 1. extension calls asterisk (call leg a, call-id 1234567890) 2. asterisk makes outbound trunk call (call leg b,

[asterisk-users] A bit OT - Configure GoIP for Asterisk

2017-10-02 Thread Steve Edwards
I recently received a GoIP-32 for a client project -- primarily outbound calling. How should a GoIP be configured for Asterisk? No fancy shmancy Elastix or FPBX GUI -- just using the configuration files. Single Server Mode, Config By Line, and Trunk Gateway Mode all seem likely suspects.

Re: [asterisk-users] PJSIP add header not working

2017-10-02 Thread Bryant Zimmerman
Andre For this to work we have had to go to using the b() option in the dial legs for the calls that are pasting up. You call a context that gets run before the calls are made on each channel. This allows you to add headers to the new pjsip channels. It works well. You can also set

Re: [asterisk-users] PJSIP add header not working

2017-10-02 Thread Joshua Colp
On Mon, Oct 2, 2017, at 12:06 PM, Andre Gronwald wrote: > Hi, > I am trying to add a custom header to my calls to map several call-legs > into a global call for viewing. > > For this to work I read the call-id from pjsip-channel and write it into > X-CID: > > ## > -- Executing

Re: [asterisk-users] PJSIP add header not working

2017-10-02 Thread Loic Chabert
Hi, Following some new behaviour on PJSIP, adding SIP header must be done using a subrouting. Please find below my working configuration: *[subroutine]exten => caller_handler,1,NoOp()same =>n,Set(PJSIP_HEADER(add,X-CID)=${ARG1})same => n,Return()* and then, add new parameters on Dial

[asterisk-users] PJSIP add header not working

2017-10-02 Thread Andre Gronwald
Hi, I am trying to add a custom header to my calls to map several call-legs into a global call for viewing. For this to work I read the call-id from pjsip-channel and write it into X-CID: ## -- Executing [s@macro-dialout-trunk-predial-hook:4] Set("PJSIP/10-0006",

Re: [asterisk-users] Gerrit usage?

2017-10-02 Thread Daniel Tryba
On Fri, Sep 29, 2017 at 12:27:53PM -0300, Joshua Colp wrote: > > "git checkout -b 13" appears to fix this. > > This did not create a branch from 13. This created a branch named "13" > from the branch you were on, which was most likely master. That is why > your "git review" is not working as you