Can't dial trunk to trunk extensions

XcamaroX · October 31, 2018, 7:22pm

Hey guys,

I have an issue that prevents 2 different PBX’s dial internal extensions from trunk to trunk.

I have Trunk A at one location and Trunk B at another location.

Both are connected as I can see the link is healthy but when I dial internal extensions I get a message that the extension is not available.

For example in Trunk A, all extensions are in the 3000 range and in Trunk B are all on 4000.

If from Trunk B I try to call 3001 it says that, that “Your call cannot be completed as dialed, please check…”).

I have setup the Outbound Route but still have this issue.

This is what I find in the terminal, but I cant seem to find the Unkowns.

Reliably Transmitting (NAT) to providerIP:5060:

OPTIONS sip:voip.com SIP/2.0

Via: SIP/2.0/UDP publicIP:5160;branch=z9hG4bK283b84f3;rport

Max-Forwards: 70

From: &quot;Unknown&quot; &lt;sip:[email protected]:5160&gt;;tag=as304ae73c

To: &lt;sip:voipprovider.com&gt;

Contact: &lt;sip:[email protected]:5160&gt;

Call-ID: [email protected] XXX.XXX.XXX.XXX:5160

CSeq: 102 OPTIONS

User-Agent: FPBX-13.0.195.4(13.17.0)

Date: Wed, 31 Oct 2018 19:06:40 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0

---

&lt;--- SIP read from UDP:providerIP:5060 ---&gt;

SIP/2.0 200 OK

Via: SIP/2.0/UDP publicIP:5160;branch=z9hG4bK283b84f3;received= publicIP;rport=16237

From: &quot;Unknown&quot; &lt;sip:[email protected] XXX.XXX.XXX.XXX:5160&gt;;tag=as304ae73c

To: &lt;sip:voipprovider.com&gt;;tag=as1ecbff6c

Call-ID: [email protected] XXX.XXX.XXX.XXX:5160

CSeq: 102 OPTIONS

Server: voip.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Contact: &lt;sip:providerIP:5060&gt;

Accept: application/sdp

Content-Length: 0

&lt;-------------&gt;

--- (12 headers 0 lines) ---

Really destroying SIP dialog '[email protected]:5160' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.XX4.5:5060:

OPTIONS sip:192.168.XX4.5 SIP/2.0

Via: SIP/2.0/UDP 192.168.XX3.5:5160;branch=z9hG4bK35c747b5;rport

Max-Forwards: 70

From: &quot;Unknown&quot; &lt;sip:[email protected]:5160&gt;;tag=as6d1317a5

To: &lt;sip:192.168.XX4.5&gt;

Contact: &lt;sip:[email protected]:5160&gt;

Call-ID: [email protected]:5160

CSeq: 102 OPTIONS

User-Agent: FPBX-13.0.195.4(13.17.0)

Date: Wed, 31 Oct 2018 19:06:40 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0

---

&lt;--- SIP read from UDP:192.168.XX4.5:5060 ---&gt;

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.XX3.5:5160;branch=z9hG4bK35c747b5;received=192.168.XX3.5;rport=5160

From: &quot;Unknown&quot; &lt;sip:[email protected]:5160&gt;;tag=as6d1317a5

To: &lt;sip:192.168.XX4.5&gt;;tag=as40aed762

Call-ID: [email protected]:5160

CSeq: 102 OPTIONS

Server: FPBX-13.0.195.13(13.23.1)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Contact: &lt;sip:192.168.XX4.5:5060&gt;

Accept: application/sdp

Content-Length: 0

cynjut · November 1, 2018, 1:27pm

You can’t send a local extension number to your SIP provider. You need to set up an Inter Asterisk trunk (using SIP or AIX) between the two and then set up dial rules that tells your local Asterisk server (on each end) how to get to the other server.

XcamaroX · November 1, 2018, 1:52pm

Sorry for the confusion, but both trunks are merged by SIP trunk and that link is healthy.

The Outbound Route points to this intra-trunk and with the correct dial pattern.

Something so simple and it doesn’t work, is what I’m trying to figure out.

What else should I look for?

cynjut · November 1, 2018, 1:58pm

The words you use here tell me you have two Asterisk servers, one at location “3000” and one at location “4000”. If that’s the case, then the server at location 3000 can only dial local extensions for the server at location 3000. In order to dial the extensions at location 4000, you need another trunk that points between the two servers directly (not using your POTS provider).

The /var/log/asterisk/full log would actually be a good place to look to figure this out. It will tell you why the call is failing.

XcamaroX · November 1, 2018, 2:31pm

OK, so trunk A and B are merged by an intra-trunk (SIP) and that trunk connects fine. The Outbound Routes created are pointing to the respective trunks.

Now, could it be that on Trunk A the extensions are PJSIP, and on Trunk B the extensions are SIP?

Do I need to configure something extra so that when the PJSIP extension calls a SIP extension, they can talk?

cynjut · November 1, 2018, 2:37pm

…

Stewart1 · November 1, 2018, 2:42pm

To direct dial extensions from one office to the other, set the context for the intracompany trunk (at both ends) to from-internal.

However, note that this assumes that the systems trust each other (office A could use office B’s system to call anywhere in the world, and vice-versa). If that’s a problem, you’ll need to create a custom context or take other security measures.

XcamaroX · November 6, 2018, 8:18pm

New development.

Both trunks on these 2 PBX’s show as registered.

PBX A is registered to PBX B and viceversa via a SIP trunk.

To-B 192.168.XXB.XXX Yes Yes 5060 OK (43 ms)
To-A 192.168.XXA.XXX Yes Yes 5060 OK (32 ms)

Now, PBX B can dial ALL extensions from A and B and it works great.
PBX A can dial all extensions of PBX A but when I dial PBX B extensions I get that “the extension just dialed is not in service” and I get this in the FULL asterisk log for PBX B:

2018-11-06 15:10:00] VERBOSE[15082][C-00000036] pbx.c: Executing [[email protected]:6] Log(“SIP/192.168.XXA.XXX-00000071”, “WARNING,“Rejecting unknown SIP connection from 192.168.XXA.XXX””) in new stack
[2018-11-06 15:10:00] WARNING[15082][C-00000036] Ext. s: “Rejecting unknown SIP connection from 192.168.XXA.XXX”

They are connected via a VPN tunnel and both subnets can ping each other.

These is the inter-trunk on PBX A pointing to B (outgoing and blank in incoming)

type=friend
host=192.168.XXB.XXX
context=from-trunk
insecure=invite
allow=all
qualify=yes

This is the inter-trunk of PBX B pointing to PBX A (outgoing and blank in incoming)

type=friend
host=192.168.XXA.XXX
context=from-trunk
insecure=invite
allow=all
qualify=yes

Why is it that there is a one-way working trubnk and not the other? When EVERY setting is the same back and forth?

Any help will be appreciated!

arielgrin · November 6, 2018, 10:45pm

Try context=from-internal

BlazeStudios · November 6, 2018, 11:54pm

There is actually a bit wrong with this.

You are peering IP-to-IP, so having type=friend is wrong because it makes both a user and a peer. Meaning it will want to look at the username (trunk name in Outgoing Settings) for a match. To give you an example, all your Chan_SIP extensions are “friend”. So it should be type=peer

2.) The insecure setting is missing the “port” option, unless you are specifically matching ports but then you don’t have any ports defined. So it should be insecure=invite,port

3.) You’re using a VPN so I’m going to take a stab that there is not NAT. So you need nat=no

) As other’s have pointed out, context=from-internal

Part of your issue is #1, you are using type=friend and it’s trying to validate against the user. Make the updates and try again.

XcamaroX · November 7, 2018, 2:44am

Hey Tom, I tried all your suggestions and I’m still with the same problem.

This is what my Trunk looks, and it still connects, but I cant dial the extensions from A to B, but I can from B - A.

I have tried wiht and without nat, with and without port and peer and friend.

What gives?

type=peer
host=192.168.XXA.XXX
context=from-internal
port=5060
insecure=port,invite
allow=all
qualify=yes
nat=no

BlazeStudios · November 7, 2018, 3:26am

You made changes to both peers? And you need to show a failed call from both the PBX making the call and receiving the call. So 1 call, logs from the two sides.

Let’s see what is actually happening versus guessing.

asterisk -rvvvvvvvvv (on both machines)
Make call
copy and past logs here or to pastebin and give links here.

XcamaroX · November 7, 2018, 2:50pm

I made the changes to both, yes.

This is the log from the extension (650) making the call and on the PBX that does not work properly to Extension (504)

> == Using SIP RTP TOS bits 184

== Using SIP RTP CoS mark 5

> 0x7f34f0031730 – Strict RTP learning after remote address set to: 192.168.XXA.251:38392

– Executing [[email protected]:1] Macro (" SIP/650-00000063 ", " user-callerid,LIMIT,EXTERNAL, ") in new stack

– Executing [[email protected]:1] Set (" SIP/650-00000063 ", " TOUCH_MONITOR=1541601318.99 ") in new stack

– Executing [[email protected]:2] Set (" SIP/650-00000063 ", " AMPUSER=650 ") in new stack

– Executing [[email protected]:3] GotoIf (" SIP/650-00000063 ", " 0?report ") in new stack

– Executing [[email protected]:4] ExecIf (" SIP/650-00000063 ", " 1?Set(REALCALLERIDNUM=650) ") in new stack

– Executing [[email protected]:5] Set (" SIP/650-00000063 ", " AMPUSER=650 ") in new stack

– Executing [[email protected]:6] GotoIf (" SIP/650-00000063 ", " 0?limit ") in new stack

– Executing [[email protected]:7] Set (" SIP/650-00000063 ", " AMPUSERCIDNAME=650 ") in new stack

– Executing [[email protected]:8] ExecIf (" SIP/650-00000063 ", " 0?Set(__CIDMASQUERADING=TRUE) ") in new stack

– Executing [[email protected]:9] GotoIf (" SIP/650-00000063 ", " 0?report ") in new stack

– Executing [[email protected]:10] Set (" SIP/650-00000063 ", " AMPUSERCID=650 ") in new stack

– Executing [[email protected]:11] Set (" SIP/650-00000063 ", " __DIAL_OPTIONS=Ttr ") in new stack

– Executing [[email protected]:12] Set (" SIP/650-00000063 ", " CALLERID(all)="650" <650> ") in new stack

– Executing [[email protected]:13] GotoIf (" SIP/650-00000063 ", " 0?limit ") in new stack

– Executing [[email protected]:14] ExecIf (" SIP/650-00000063 ", " 1?Set(GROUP(concurrency_limit)=650) ") in new stack

– Executing [[email protected]:15] NoOp (" SIP/650-00000063 ", " Macro Depth is 1 ") in new stack

– Executing [[email protected]:16] GotoIf (" SIP/650-00000063 ", " 1?report2:macroerror ") in new stack

– Goto (macro-user-callerid,s,17)

– Executing [[email protected]:17] GotoIf (" SIP/650-00000063 ", " 1?continue ") in new stack

– Goto (macro-user-callerid,s,35)

– Executing [[email protected]:35] Set (" SIP/650-00000063 ", " CALLERID(number)=650 ") in new stack

– Executing [[email protected]:36] Set (" SIP/650-00000063 ", " CALLERID(name)=650 ") in new stack

– Executing [[email protected]:37] GotoIf (" SIP/650-00000063 ", " 0?cnum ") in new stack

– Executing [[email protected]:38] Set (" SIP/650-00000063 ", " CDR(cnam)=650 ") in new stack

– Executing [[email protected]:39] Set (" SIP/650-00000063 ", " CDR(cnum)=650 ") in new stack

– Executing [[email protected]:40] Set (" SIP/650-00000063 ", " CHANNEL(language)=en ") in new stack

– Executing [[email protected]:2] Gosub (" SIP/650-00000063 ", " sub-record-check,s,1(out,504,dontcare) ") in new stack

– Executing [[email protected]:1] GotoIf (" SIP/650-00000063 ", " 0?initialized ") in new stack

– Executing [[email protected]:2] Set (" SIP/650-00000063 ", " __REC_STATUS=INITIALIZED ") in new stack

– Executing [[email protected]:3] Set (" SIP/650-00000063 ", " NOW=1541601318 ") in new stack

– Executing [[email protected]:4] Set (" SIP/650-00000063 ", " __DAY=07 ") in new stack

– Executing [[email protected]:5] Set (" SIP/650-00000063 ", " __MONTH=11 ") in new stack

– Executing [[email protected]:6] Set (" SIP/650-00000063 ", " __YEAR=2018 ") in new stack

– Executing [[email protected]:7] Set (" SIP/650-00000063 ", " __TIMESTR=20181107-093518 ") in new stack

– Executing [[email protected]:8] Set (" SIP/650-00000063 ", " __FROMEXTEN=650 ") in new stack

– Executing [[email protected]:9] Set (" SIP/650-00000063 ", " __MON_FMT=wav ") in new stack

– Executing [[email protected]:10] NoOp (" SIP/650-00000063 ", " Recordings initialized ") in new stack

– Executing [[email protected]:11] ExecIf (" SIP/650-00000063 ", " 0?Set(ARG3=dontcare) ") in new stack

– Executing [[email protected]:12] Set (" SIP/650-00000063 ", " REC_POLICY_MODE_SAVE= ") in new stack

– Executing [[email protected]:13] ExecIf (" SIP/650-00000063 ", " 0?Set(REC_STATUS=NO) ") in new stack

– Executing [[email protected]:14] GotoIf (" SIP/650-00000063 ", " 3?checkaction ") in new stack

– Goto (sub-record-check,s,17)

– Executing [[email protected]:17] GotoIf (" SIP/650-00000063 ", " 1?sub-record-check,out,1 ") in new stack

– Goto (sub-record-check,out,1)

– Executing [[email protected]:1] NoOp (" SIP/650-00000063 ", " Outbound Recording Check from 650 to 504 ") in new stack

– Executing [[email protected]:2] Set (" SIP/650-00000063 ", " RECMODE=dontcare ") in new stack

– Executing [[email protected]:3] ExecIf (" SIP/650-00000063 ", " 1?Goto(routewins) ") in new stack

– Goto (sub-record-check,out,7)

– Executing [[email protected]:7] Gosub (" SIP/650-00000063 ", " recordcheck,1(dontcare,out,504) ") in new stack

– Executing [[email protected]:1] NoOp (" SIP/650-00000063 ", " Starting recording check against dontcare ") in new stack

– Executing [[email protected]:2] Goto (" SIP/650-00000063 ", " dontcare ") in new stack

– Goto (sub-record-check,recordcheck,3)

– Executing [[email protected]:3] Return (" SIP/650-00000063 ", "") in new stack

– Executing [[email protected]:8] Return (" SIP/650-00000063 ", "") in new stack

– Executing [[email protected]:3] Set (" SIP/650-00000063 ", " MOHCLASS=default ") in new stack

– Executing [[email protected]:4] Set (" SIP/650-00000063 ", " _NODEST= ") in new stack

– Executing [[email protected]:5] Macro (" SIP/650-00000063 ", " dialout-trunk,1,504,off ") in new stack

– Executing [[email protected]:1] Set (" SIP/650-00000063 ", " DIAL_TRUNK=1 ") in new stack

– Executing [[email protected]:2] ExecIf (" SIP/650-00000063 ", " 0?Set(DIAL_OPTIONS=tr) ") in new stack

– Executing [[email protected]:3] GosubIf (" SIP/650-00000063 ", " 0?sub-pincheck,s,1() ") in new stack

– Executing [[email protected]:4] GotoIf (" SIP/650-00000063 ", " 0?disabletrunk,1 ") in new stack

– Executing [[email protected]:5] Set (" SIP/650-00000063 ", " DIAL_NUMBER=504 ") in new stack

– Executing [[email protected]:6] Set (" SIP/650-00000063 ", " DIAL_TRUNK_OPTIONS=Ttr ") in new stack

– Executing [[email protected]:7] Set (" SIP/650-00000063 ", " OUTBOUND_GROUP=OUT_1 ") in new stack

– Executing [[email protected]:8] Set (" SIP/650-00000063 ", " DIAL_TRUNK_OPTIONS=T ") in new stack

– Executing [[email protected]:9] GotoIf (" SIP/650-00000063 ", " 1?nomax ") in new stack

– Goto (macro-dialout-trunk,s,11)

– Executing [[email protected]:11] GotoIf (" SIP/650-00000063 ", " 0?skipoutcid ") in new stack

– Executing [[email protected]:12] Macro (" SIP/650-00000063 ", " outbound-callerid,1 ") in new stack

– Executing [[email protected]:1] ExecIf (" SIP/650-00000063 ", " 0?Set(CALLERPRES(name-pres)=) ") in new stack

– Executing [[email protected]:2] ExecIf (" SIP/650-00000063 ", " 0?Set(CALLERPRES(num-pres)=) ") in new stack

– Executing [[email protected]:3] ExecIf (" SIP/650-00000063 ", " 0?Set(REALCALLERIDNUM=650) ") in new stack

– Executing [[email protected]:4] ExecIf (" SIP/650-00000063 ", " 0?Set(AMPUSER=650) ") in new stack

– Executing [[email protected]:5] GotoIf (" SIP/650-00000063 ", " 1?normcid ") in new stack

– Goto (macro-outbound-callerid,s,9)

– Executing [[email protected]:9] Set (" SIP/650-00000063 ", " USEROUTCID=7865742525 ") in new stack

– Executing [[email protected]:10] Set (" SIP/650-00000063 ", " EMERGENCYCID= ") in new stack

– Executing [[email protected]:11] Set (" SIP/650-00000063 ", " TRUNKOUTCID= ") in new stack

– Executing [[email protected]:12] GotoIf (" SIP/650-00000063 ", " 1?trunkcid ") in new stack

– Goto (macro-outbound-callerid,s,17)

– Executing [[email protected]:17] ExecIf (" SIP/650-00000063 ", " 0?Set(CALLERID(all)=) ") in new stack

– Executing [[email protected]:18] ExecIf (" SIP/650-00000063 ", " 1?Set(CALLERID(all)=7865742525) ") in new stack

– Executing [[email protected]:19] ExecIf (" SIP/650-00000063 ", " 0?Set(CALLERID(all)=) ") in new stack

– Executing [[email protected]:20] ExecIf (" SIP/650-00000063 ", " 0?Set(CALLERPRES(name-pres)=prohib_passed_screen) ") in new stack

– Executing [[email protected]:21] ExecIf (" SIP/650-00000063 ", " 0?Set(CALLERPRES(num-pres)=prohib_passed_screen) ") in new stack

– Executing [[email protected]:22] Set (" SIP/650-00000063 ", " CDR(outbound_cnum)=7865742525 ") in new stack

– Executing [[email protected]:23] Set (" SIP/650-00000063 ", " CDR(outbound_cnam)= ") in new stack

– Executing [[email protected]:13] GosubIf (" SIP/650-00000063 ", " 0?sub-flp-1,s,1() ") in new stack

– Executing [[email protected]:14] Set (" SIP/650-00000063 ", " OUTNUM=504 ") in new stack

– Executing [[email protected]:15] Set (" SIP/650-00000063 ", " custom=SIP/PBXB-out ") in new stack

– Executing [[email protected]:16] ExecIf (" SIP/650-00000063 ", " 0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T) ") in new stack

– Executing [[email protected]:17] ExecIf (" SIP/650-00000063 ", " 0?Set(DIAL_TRUNK_OPTIONS=TM(confirm)) ") in new stack

– Executing [[email protected]:18] Macro (" SIP/650-00000063 ", " dialout-trunk-predial-hook, ") in new stack

– Executing [[email protected]:1] MacroExit (" SIP/650-00000063 ", "") in new stack

– Executing [[email protected]:19] GotoIf (" SIP/650-00000063 ", " 0?bypass,1 ") in new stack

– Executing [[email protected]:20] ExecIf (" SIP/650-00000063 ", " 1?Set(CONNECTEDLINE(num,i)=504) ") in new stack

– Executing [[email protected]:21] ExecIf (" SIP/650-00000063 ", " 1?Set(CONNECTEDLINE(name,i)=CID:7865742525) ") in new stack

– Executing [[email protected]:22] ExecIf (" SIP/650-00000063 ", " 0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)XXXXXXXXX) ") in new stack

– Executing [[email protected]:23] GotoIf (" SIP/650-00000063 ", " 0?customtrunk ") in new stack

– Executing [[email protected]:24] Dial (" SIP/650-00000063 ", " SIP/fl-out/504,300,T ") in new stack

== Using SIP RTP TOS bits 184

== Using SIP RTP CoS mark 5

– Called SIP/PBXB-out/504

> 0x7f34f4007c80 – Strict RTP learning after remote address set to: 192.168.XXB.XXX:14048

– SIP/PBXB-out-00000064 answered SIP/650-00000063

– Channel SIP/fl-out-00000064 joined ‘simple_bridge’ basic-bridge <ae92b068-e115-4423-9733-d3495e631031>

– Channel SIP/650-00000063 joined ‘simple_bridge’ basic-bridge <ae92b068-e115-4423-9733-d3495e631031>

> 0x7f34f0031730 – Strict RTP switching to RTP target address 192.168.XXA.XXX:38392 as source

> 0x7f34f4007c80 – Strict RTP switching to RTP target address 192.168.XXB.XXX:14048 as source

– Channel SIP/650-00000063 left ‘simple_bridge’ basic-bridge <ae92b068-e115-4423-9733-d3495e631031>

== Spawn extension (macro-dialout-trunk, s, 24) exited non-zero on ‘SIP/650-00000063’ in macro ‘dialout-trunk’

== Spawn extension (from-internal, 504, 5) exited non-zero on ‘SIP/650-00000063’

– Executing [[email protected]:1] Macro (" SIP/650-00000063 ", " hangupcall ") in new stack

– Executing [[email protected]:1] GotoIf (" SIP/650-00000063 ", " 1?theend ") in new stack

– Goto (macro-hangupcall,s,3)

– Executing [[email protected]:3] ExecIf (" SIP/650-00000063 ", " 0?Set(CDR(recordingfile)=) ") in new stack

– Executing [[email protected]:4] Hangup (" SIP/650-00000063 ", "") in new stack

== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/650-00000063’ in macro ‘hangupcall’

== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/650-00000063’

– Channel SIP/PBXB-out-00000064 left ‘simple_bridge’ basic-bridge <ae92b068-e115-4423-9733-d3495e631031>

Now this is the log from PBXB on extension (504)

Connected to Asterisk 13.23.1 currently running on FL (pid = 2945)

== Using SIP RTP TOS bits 184

== Using SIP RTP CoS mark 5

> 0x7f2e60024540 – Strict RTP learning after remote address set to: 192.168.XXA.XXX:19994

– Executing [[email protected]:1] NoOp (" SIP/192.168.XXA.XXX-00000075 ", " Received incoming SIP connection from unknown peer to 504 ") in new stack

– Executing [[email protected]:2] Set (" SIP/192.168.XXA.XXX-00000075 ", " DID=504 ") in new stack

– Executing [[email protected]:3] Goto (" SIP/192.168.XXA.XXX-00000075 ", " s,1 ") in new stack

– Goto (from-sip-external,s,1)

– Executing [[email protected]:1] GotoIf (" SIP/192.168.XXA.XXX-00000075 ", " 1?setlanguage:checkanon ") in new stack

– Goto (from-sip-external,s,2)

– Executing [[email protected]:2] Set (" SIP/192.168.XXA.XXX-00000075 ", " CHANNEL(language)=en ") in new stack

– Executing [[email protected]:3] GotoIf (" SIP/192.168.XXA.XXX-00000075 ", " 1?noanonymous ") in new stack

– Goto (from-sip-external,s,5)

– Executing [[email protected]:5] Set (" SIP/192.168.XXA.XXX-00000075 ", " TIMEOUT(absolute)=15 ") in new stack

– Channel will hangup at 2018-11-07 09:35:33.357 EST.

– Executing [[email protected]:6] Log (" SIP/192.168.XXA.XXX-00000075 ", " WARNING,"Rejecting unknown SIP connection from 192.168.XXA.XXX" ") in new stack

[2018-11-07 09:35:18] WARNING [9329][C-0000003a]: Ext. s : 6 @ from-sip-external : "Rejecting unknown SIP connection from 192.168.XXA.XXX"

– Executing [[email protected]:7] Answer (" SIP/192.168.XXA.XXX-00000075 ", "") in new stack

> 0x7f2e60024540 – Strict RTP switching to RTP target address 192.168.XXA.XXX:19994 as source

– Executing [[email protected]:8] Wait (" SIP/192.168.XXA.XXX-00000075 ", " 2 ") in new stack

– Executing [[email protected]:9] Playback (" SIP/192.168.XXA.XXX-00000075 ", " ss-noservice ") in new stack

– <SIP/192.168.XXA.XXX-00000075> Playing ‘ss-noservice.ulaw’ (language ‘en’)

– Executing [[email protected]:1] Hangup (" SIP/192.168.XXA.XXX-00000075 ", "") in new stack

== Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/192.168.XXA.XXX-00000075’

XcamaroX · November 7, 2018, 2:51pm

If I make the call from 504 to 650 (or any other extension from the respective PBX’s) the call works fine as intended.

BlazeStudios · November 7, 2018, 3:29pm

The trunk on the PBX that 504 lives on is not correct. It doesn’t think the IP the call is coming from is valid.

XcamaroX · November 7, 2018, 3:49pm

This is the inter-trunk that connects A to B:

type=friend
host=192.168.XXB.XXX
context=from-internal
port=5060
insecure=invite,port
allow=all
qualify=yes
nat=no

And this is the link as connected
PBXB 192.168.XXB.XXX No No 5060 OK (18 ms)

This is the inter-trunk that connects B to A

type=friend
host=192.168.XXA.XXX
context=from-internal
port=5060
insecure=invite,port
allow=all
qualify=yes
nat=no

And this is the link as connected
PBXA 192.168.XXA.XXX No No 5060 OK (41 ms)

What do you think?

XcamaroX · November 7, 2018, 8:39pm

Found the issue, somehow the NIC card was getting a static and DHCP IP, after hardwiring to manual, the problem went away.

Thanks so much for all your guys help!

system · November 14, 2018, 8:39pm

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