similar to: Setting different caller-id for second leg of the Originate

Hi We have the following test .call file and test dialplan: I can't set a custom CDR var to a value on one channel leg, and another value on the connected channel leg? Is there a way I can woraround this issue? ## test call file Channel: Local/queue at TiagoGeada CallerID: teste-geada:0:210332450: MaxRetries: 0 RetryTime: 1 WaitTime: 8640 Account: teste-geada Context: TiagoGeada

Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the

We have several customers we need to place outbound calls for (in a single system). May have to place calls for thousands of different caller ids. Customer signs a contract guaranteeing the caller id they provide is owned by them. I have everything setup for AMI Originate and can place the calls. However, I'm encountering a problem with the caller id. The system I'm dialing through

I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hello friends, I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can access the caller Call ID (fbasename field in voipmonitor cdr) looking at the SIPCALLID variable in asterisk, but how can I access from within asterisk the Call ID of the second leg of the call (the one originating from asterisk to the destination peer)? is there a variable holding this value? Thank you

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

Hi, Here is my confing: [out] Exten => _X.,1,Noop() Exten => _X.,2,Dial(SIP/${EXTEN}@peer,60,gcU(do_dtmf_cc-take-call,s,1)) Exten => _X.,3,Playback(tt-monkeys) Exten => _X.,4,Playback(tt-monkeys) Exten => _X.,5,Playback(tt-monkeys) Exten => h,1,Noop(ABCDEFGHIJKLMNOPQRSTUVWXYZ) [do_dtmf_cc-take-call] Exten => s,1,AGI(agi://127.0.0.1:4579/update_call_status?status=60) Exten

I am using AMI Originate to perform a new outbound call. The SIP Provider we send the call to wants us to pass the caller id of the person we are calling for in the Contact header. For the AMI Originate, I pass the caller id information data in the CallerID field. However, this is never being passed through the PJSIP INVITE header Action: Originate ActionID: S598 Channel: PJSIP/133 at 1002

An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users at lists.digium.com' <asterisk-users at lists.digium.com> Subject: [asterisk-users] With ARI,

I am trying to make a data channel using ISDN and i need to set the caller id num field. Can any body tell me how i can set the caller id field since i notice in chan_dahdi.conf callerid field doesn't work with channel originate. Thanks, Pawel -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, When implementing click2dial feature, I can trigger an Aastra phone to auto-answer using statement like : SIPAddHeader(Alert-Info: info=alert-autoanswer); This is very convenient when trying to reach a distant party (ie through PSTN) The trouble is when 2 Aastra are calling each other over the LAN, this single statement is memorized somehow and both phones (caller and callee) auto-answer.

I have Meetme conferencing compiled for Debian as per http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the outgoing dir, and it intitiates a call to a local extension as a channel, but the call seems to block on the Meetme() command. That extension completes the outgoing Dial(SIP) command to my phone, announcing that leg is the only member of the conference, and just waits. If I

Hi Jöran, Would it be possible to see an example using curl of how you are passing the PAI Header through ARI create? Dan From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Jöran Vinzens Sent: Friday, August 7, 2020 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] With

Is there a way to specify the outbound leg Call-ID? -- Eric Chamberlain

Hi Dan, i did it wrong, sorry: curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,

When I make a regular call from my SIP phone connected to my Asterisk server I have no issues, however when I make a call using Originate : 'Channel'=>"SIP/1XXXXXXXXXX@sip.broadvoice.com", 'Context'=>'mycontext', 'Exten'=>'899', 'Priority'=>1, 'Callerid'=>'whatever')); It creates a screech sound when the

I'm trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at

Hi All, I have been running a environment with asterisk 1.4.20.1 for some time now with no issue but have recently added some extra functionality (enabled call recording via MixMonitor) and ran into some deadlock issues which seem to be well documented with earlier 1.4.x releases so have decided to take the plunge and upgrade. I decided to start testing with 1.6.2 but have run into a couple

I am using AMI to Originate a call. I have been able to get the caller id number to be passed through. However, I can't get the name to be passed through. A person I'm working with has a Freeswitch that is able to pass the caller id name and number through for their call. Comparing the Asterisk SIP messages to the Freeswitch SIP messages, I have narrowed the problem down to a single