PJSIP Trunk unable to call out

FreePBX Endpoints
1

Ok so I have an odd scenario and problem.

I am running a FreePBX Distro 12.7.6-1904-1.sng7 and Asterisk 15.7.3.

Now my company uses Yealink Video Conferencing devices. I have a YMS server which allows me to centralize video conferencing. But my users want to be able to SIP call through YMS so working with yealink we setup a pjsip trunk between YMS and FreePBX. Now this works when i want to call extensions from YMS. My YMS connected device called yms called extension 6354 and the user was able to pick up the call and talk.

executing [s@macro-dial-one:54] Dial(“SIP/yms-00007cca”, “SIP/6354,30,HhTtrb(func-apply-sipheaders^s^1)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– SIP/6354-00007ccb Internal Gosub(func-apply-sipheaders,s,1) start
– Executing [s@func-apply-sipheaders:1] NoOp(“SIP/6354-00007ccb”, “Applying SIP Headers to channel SIP/6354-00007ccb”) in new stack
– Executing [s@func-apply-sipheaders:2] Set(“SIP/6354-00007ccb”, “TECH=SIP”) in new stack
– Executing [s@func-apply-sipheaders:3] Set(“SIP/6354-00007ccb”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“SIP/6354-00007ccb”, “0”) in new stack
– Jumping to priority 11
– Executing [s@func-apply-sipheaders:12] Return(“SIP/6354-00007ccb”, “”) in new stack
== Spawn extension (from-internal, 6354, 1) exited non-zero on ‘SIP/6354-00007ccb’
– SIP/6354-00007ccb Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called SIP/6354

But when I want to make outbound calls out of my FreePBX trunk and outbound route it fails with no service.

-- Executing [###6354@from-sip-external:1] NoOp("SIP/yms-00007d13", "Received incoming SIP connection from unknown peer to ###6354") in new stack

0x7f3b45e75690 – Strict RTP learning after remote address set to: 192.168.3.72:51010
– Executing [###6354@from-sip-external:1] NoOp(“SIP/yms-00007d13”, “Received incoming SIP connection from unknown peer to ###6354”) in new stack
– Executing [###6354@from-sip-external:2] Set(“SIP/yms-00007d13”, “DID=###6354”) in new stack
– Executing [###6354@from-sip-external:3] Goto(“SIP/yms-00007d13”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/yms-00007d13”, “1?setlanguage:checkanon”) in new stack
– Goto (from-sip-external,s,2)
– Executing [s@from-sip-external:2] Set(“SIP/yms-00007d13”, “CHANNEL(language)=en”) in new stack
– Executing [s@from-sip-external:3] GotoIf(“SIP/yms-00007d13”, “0?noanonymous”) in new stack
– Executing [s@from-sip-external:4] Goto(“SIP/yms-00007d13”, “from-trunk,###6354,1”) in new stack
– Goto (from-trunk,###6354,1)
– Executing [###6354@from-trunk:1] Set(“SIP/yms-00007d13”, “__FROM_DID=###6354”) in new stack
– Executing [###6354@from-trunk:2] NoOp(“SIP/yms-00007d13”, “Received an unknown call with DID set to ###6354”) in new stack
– Executing [###6354@from-trunk:3] Goto(“SIP/yms-00007d13”, “s,a2”) in new stack
– Goto (from-trunk,s,2)
– Executing [s@from-trunk:2] Answer(“SIP/yms-00007d13”, “”) in new stack
0x7f3b45e75690 – Strict RTP switching to RTP target address 192.168.3.72:51010 as source
– Executing [s@from-trunk:3] Log(“SIP/yms-00007d13”, “WARNING,Friendly Scanner from 192.168.3.72”) in new stack
[2019-11-18 09:28:23] WARNING[29092][C-000044e1]: Ext. s:3 @ from-trunk: Friendly Scanner from 192.168.3.72

And the call is dropped.

Now my pjsip trunk to YMS is using Context = from-pstn and i have Authentication set to None and Registration set to None.

I am unsure if it is picking up my outbound route because I use Extension Routing and since YMS isn’t an extension how to I set it up to allow calls from the PJSIP trunk?

Now I have been through this with Yealink and they have exhausted their ability so i’ve turned to the FreePBX community to see if someone can help.

Thank you in advance.

Ok so no one responded but me and I’ve been working other troubles but now 7 days and it’s closed.

2

I’m going to convert the trunk to a Chan sip trunk but I’m thinking something about extension routing.

More info and testing in the morning.

3

Ok Chan sip is no go. Yms uses port 5060 so I cannot setup Chan sip on 5060 so now to try a iax2 trunk. I’ll test tomorrow.

4

Ok YMS uses FreeSWITCH as their back end software to communicate video conferencing and SIP calling.

I’ll keep looking into this.

5

Ok so the big thing here is the ability for a PJSIP trunk sourced caller to be allowed to use another outbound route and ChanSIP trunk. I think from the above i’m getting dropped in a no service since it cannot find a suitable outbound route to call. Now i think this has to do with extension routing and the restriction that is putting in place. Anyone from FreePBX can you confirm that? Anyone?

Ok so I am back to the drawing board using port 5066 as a PJSIP trunk i can call extensions registered directly on the FreePBX server but i cannot call entities like Ring Groups or Conference Bridges. Now I’d love to be able to call Conference Bridges and then just direct all YMS VC units to SIP call a conference bridge and then calling parties would call into that bridge also. That would fulfill my current need but what happens when a user wants to SIP call a cell phone and doesn’t want the user to have call a bridge.

6

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