similar to: Multiple phones on same PJSIP account

On Monday 19 June 2023 at 15:09:44, TTT wrote: > I am creating a dialplan where a single user (Alice) has two offices. Both > of her phones should ring if her extension is called. > > I could use a ring group, but I'm wondering can both phones use the same > PJSIP extension account (username/secret)? Yes. This is one of the major advantages to using PJSIP instead of

Ok I've got multiple phone sets registered with the same extension/secret. However, this causes a strange problem. If I have 3 phone sets registered on extension 123, and I then call extension 123 (from extension 456), only a SINGLE phone set will ring. Is this by design or a bug? Does only the most recently registered phone set ring when I call the extension? Seems odd...is there a way

That begs another interesting question...with analog phones picking up two extensions on the same "line" allow multiple people to participate on the call (without a "conference" feature) Does this become possible with multiple phones on the same PJSIP account? Or would the first phone answered need to somehow conference in the other phone? -----Original Message----- From:

On Monday 19 June 2023 at 16:26:05, TTT wrote: > That begs another interesting question...with analog phones picking up two > extensions on the same "line" allow multiple people to participate on the > call (without a "conference" feature) > > Does this become possible with multiple phones on the same PJSIP account? No. Antony. -- There are 10 types of

On Wednesday 21 June 2023 at 17:52:16, TTT wrote: > Ok I've got multiple phone sets registered with the same extension/secret. > > However, this causes a strange problem. If I have 3 phone sets registered > on extension 123, and I then call extension 123 (from extension 456), only > a SINGLE phone set will ring. What values do you have for "max_contacts" and

I am following the instructions in https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am trying to make a call from extension Alice (6001) to extension for Bob (6002). When I make the call, I can hear the ringing on Alice's phone (caller), but Bob's phone (callee) doesn't ring, or show a call coming in from Alice. My setup and environment is as follows: Alice, Bob

I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: [Twilio] type=auth auth_type=userpass password=mysecret username=myun However, my calls using the trunk are rejected with a 403. Using pjsip logging I notice that the outgoing invite does not have an authentication line. Why is Asterisk not sending

The AGI debug command worked well, and I found the offending command: SetCallerPres(allowed) That worked in Asterisk 13, but from my google searching it looks like this command has disappeared in Asterisk 20 (actually everything after ver 13). I thought it was replaced with CALLERPRES(allowed) but this generated an error too in Asterisk 20. Is there a replacement command? -----Original

I tried that (only needed to add rewrite_contact=yes) but it didn't help. BTW, the CONTACT: line holds the correct ip! Only the FROM: line holds the wrong (private) IP. I'm still learning SIP...but I assume the FROM should also hold the rewritten public IP. Just don't know how to force Asterisk to do that. -----Original Message----- From: Eric Wieling [mailto:ewieling at

Hi, I am using Asterisk 13.6.0 and was wondering if I can programmatically add users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk server using API of some sort. Please do let me know. Thanks, Sonny. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I didn't use pjsip_wizard, I'm kind of crafting this by hand as I learn. I actually have a plain asterisk, and a FreePBX, system to help me learn. I sometimes create something in FreePBX to see what it does to the config files. So that's how I modelled my pjsip.X.conf files If I issue the command "pjsip show endpoint Twilio" it does show that outbound_auth=Twilio Does

Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is

    Dis you set "outbound_auth" in your endpoint configuration to Twilio? On 21/06/23 11:19, TTT wrote: > I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP > (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: > > [Twilio] > type=auth > auth_type=userpass > password=mysecret > username=myun > > However, my calls

On Wed, Jun 21, 2023 at 05:19:11PM +0000, TTT wrote: > I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP > (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: > > [Twilio] > type=auth > auth_type=userpass > password=mysecret > username=myun > > However, my calls using the trunk are rejected with a 403. Using pjsip >

type=endpoint rewrite_contact=yes force_rport=yes rtp_symmetric=yes On 6/21/23 14:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction: > > From: "MYNAME" <sip:16667778888 at

I’ve reread the documentation a few times, and what isn’t clear is whether I need an app=X parameter in the url. In other words, can I only get events for a single named statis app? Or can I get events for the entire Asterisk server? The command below (without app= parameter) results in no events being shown, but no error either. Thanks Brian (Ast newbie) From: asterisk-users

On Wed, Jun 7, 2023 at 10:46 AM TTT <lists at telium.io> wrote: > I’ve reread the documentation a few times, and what isn’t clear is whether > I need an app=X parameter in the url. In other words, can I only get > events for a single named statis app? Or can I get events for the entire > Asterisk server? > > > > The command below (without app= parameter) results in

I tried GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id) But it responds with "message": "Channel not in Stasis application" Since I want to get the call-id for a channel not in stasis I guess that won’t work. Similarly, I can’t force the channel through my own code in the dialplan, so the PJSIP_HEADER function won’t work. So it looks like I’ll

The uppercase command made a difference. I now get a call-id as show below. However, does the call-id look valid? The @0.0.0.0 seems strange. action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-3-In-00000028 Variable: CHANNEL(pjsip,call-id) Response: Success ActionID: act1 Variable: CHANNEL(pjsip,call-id) Value: 4decf884e3ae74595906283a74f7154e at 0.0.0.0 As well,

I think that’s getting me close. I’m trying to get (or recreate) the FROM and TO lines of the header, from a system running PJSIP. I think if I use CHANNEL to get local_uri and local_tag I can recreate a FROM line like: FROM=<URI>;tag=TAG And if I use CHANNEL to get remote_uri and remote_tag I can recreate a FROM line like: TO=<URI>;tag=TAG Would it be correct to assume