Ive 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 ;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4 ..
Author : TTT
Home » Articles posted by TTT (paged 2)
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 ..
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 Im wondering can both phones use the same PJSIP extension account (username/secret)?ThanksBr..
Based on postings it should be possible to get the SIP Call-ID header value from the ARI.At what point is this value available ?As well, how do Iretrieve that value – something like GET /channels/{channelId}/pjsip_..
Im trying to join a user (at SIP/99) into a conference via REST/ARI.Iwant the PBX to call the user, and then join him into an existing conference. I have created a conference in FreePBX with number 1234, and name conf. Conceptually the steps I have..
Im monitoring the ARI, and if extension 1 calls extension 2, it seems that extension 2 enters the bridge first, then extension 1 enters the bridge. Can I safely (always) determine who initiated the call by who is the latest endpoint to enter the bridge..
I have the ARI enabled on my Asterisk test box, and want to listen to all events.I cant find the syntax to do that.Can I only listen to events related to a stasis app? I was hoping that a simple wscat command like this would show me all events: ws..
You dont say what happens when you start Asterisk, but Ill assume your registration with your provider is failing.If you turn on SIP debug from CLI you can watch your registration attempts, and see the exact reason for failure.(eg: unreachable vs credentials).P..
Ive noticed on several occasions that if Asterisk starts without a network connection, then even if the network connection is restored, DNS lookups fail. After the connection is restored I can successfully do NSLOOKUPs from the command line, but ..
I want to reload the dialplan via the AMI, and I found the documentation showing the command: Action: ReloadActionID: Module:And the module names are* cdr* dnsmgr* extconfig* enum* acl* manager* http* logger* features* dsp* udptl* indications* cel* plc..