Outgoing calls not possible after update

puneet1984 · March 4, 2023, 6:27pm

Hello
outbound calls were working normally for last 2-3 months without any issues.

after a recent update of the freepbx (both system and modules), my outgoing calls are restricted.
i have not changed any settings in the outbound routes/trunks.
Inbound calls are working fine and are being diverted to IVR as per the inbound rule.
in sngrep, earlier i used to get sip info - [email protected] called XYZ [email protected]_IP then call transferred to gateway_IP and then processed further…
now i only get is extension called the number and request reached freepbx… freepbx is not transferring the call to the gateway.

Also asterisk -rvvv is not showing any output of any incoming or outgoing calls.
Similarly sip set debug on does not show any debug info.
restarted asterisk using fwconsole, even rebooted the freepbx … still no outbound calls.

My setup

Pfsense+ firewall with NAT and firewall rules
Dynamic public IP allocation by ISP - attached to DDNS domain by pfsense.
softphone - GSwave Lite on android and on iOS
FreePBX - 192.168.1.81
BSNL WINGS SIP service (India) - registered on freePBX
CO/PSTN Matrix Gateway -192.168.1.240
4 FXO lines - all occupied
4 FXS lines - all occupied - namely 30,31,32,33
default CO lines landing destination - first FXS port (ext 30)
Matrix is registered with freepbx on peer-2-peer basis.

david55 · March 4, 2023, 6:47pm

The calls aren’t reaching Asterisk, or there is a deadlock.

Deadlocks are normally resolved, at least temporarily, after a restart.

The only things I can think of that would cause calls to stop reaching you are a change in the port number, or a firewall deciding you are hostile.

You should use sngrep, to see if the calls are reaching the machine itself.

puneet1984 · March 4, 2023, 6:56pm

hello

calls are reaching to the freepbx…as seen in the sngrep, but calls are not processed further as per the dial plan…and forwarded to the gateway.
extension to extension calling is working.
inbound calls from the gateway are coming on freepbx and IVR is played.

2023/03/05 00:26:39.896903 192.168.1.99:65265 -> 192.168.1.81:5160
INVITE sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:65265;rport;branch=z9hG4bKPjd69fdb41e42c40638e4e2e9eb80e7786
Max-Forwards: 70
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>
Contact: <sip:[email protected]:65265;ob>
Call-ID: f7901c8c2e664794a57597dbab8f078c
CSeq: 18227 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.21.3
Content-Type: application/sdp
Content-Length:   342

v=0
o=- 3886964793 3886964793 IN IP4 192.168.1.99
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 10018 RTP/AVP 8 0 101
c=IN IP4 192.168.1.99
b=TIAS:64000
a=rtcp:10019 IN IP4 192.168.1.99
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1306596301 cname:0a5e3dd676d41b58


2023/03/05 00:26:39.898574 192.168.1.81:5160 -> 192.168.1.99:65265
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.99:65265;rport=65265;received=192.168.1.99;branch=z9hG4bKPjd69fdb41e42c40638e4e2e9eb80e7786
Call-ID: f7901c8c2e664794a57597dbab8f078c
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>;tag=z9hG4bKPjd69fdb41e42c40638e4e2e9eb80e7786
CSeq: 18227 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1677956199/7fcb74151473e5e6d30f3073e22e951a",opaque="7e2a565b5ebd76f8",algorithm=MD5,qop="auth"
Server: FPBX-16.0.33(18.16.0)
Content-Length:  0



2023/03/05 00:26:39.899047 192.168.1.99:65265 -> 192.168.1.81:5160
ACK sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:65265;rport;branch=z9hG4bKPjd69fdb41e42c40638e4e2e9eb80e7786
Max-Forwards: 70
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>;tag=z9hG4bKPjd69fdb41e42c40638e4e2e9eb80e7786
Call-ID: f7901c8c2e664794a57597dbab8f078c
CSeq: 18227 ACK
Content-Length:  0



2023/03/05 00:26:39.899082 192.168.1.99:65265 -> 192.168.1.81:5160
INVITE sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:65265;rport;branch=z9hG4bKPje03b2fd5e68744b08643901e965e38d2
Max-Forwards: 70
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>
Contact: <sip:[email protected]:65265;ob>
Call-ID: f7901c8c2e664794a57597dbab8f078c
CSeq: 18228 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.21.3
Authorization: Digest username="999", realm="asterisk", nonce="1677956199/7fcb74151473e5e6d30f3073e22e951a", uri="sip:[email protected]:5160", response="fd3dcb433777d4a4fc8ced3945c83deb", algorithm=MD5, cnonce="b273995d00c9437193f0d67b11bf45bf", opaque="7e2a565b5ebd76f8", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   342

v=0
o=- 3886964793 3886964793 IN IP4 192.168.1.99
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 10018 RTP/AVP 8 0 101
c=IN IP4 192.168.1.99
b=TIAS:64000
a=rtcp:10019 IN IP4 192.168.1.99
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1306596301 cname:0a5e3dd676d41b58


2023/03/05 00:26:39.901516 192.168.1.81:5160 -> 192.168.1.99:65265
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.99:65265;rport=65265;received=192.168.1.99;branch=z9hG4bKPje03b2fd5e68744b08643901e965e38d2
Call-ID: f7901c8c2e664794a57597dbab8f078c
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>
CSeq: 18228 INVITE
Server: FPBX-16.0.33(18.16.0)
Content-Length:  0



2023/03/05 00:26:39.939401 192.168.1.81:5160 -> 192.168.1.99:65265
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.99:65265;rport=65265;received=192.168.1.99;branch=z9hG4bKPje03b2fd5e68744b08643901e965e38d2
Call-ID: f7901c8c2e664794a57597dbab8f078c
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>;tag=34ac5464-76c6-4efe-95ed-7c97cba8b676
CSeq: 18228 INVITE
Server: FPBX-16.0.33(18.16.0)
Contact: <sip:192.168.1.81:5160>
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   261

v=0
o=- 3886964793 3886964795 IN IP4 192.168.1.81
s=Asterisk
c=IN IP4 192.168.1.81
t=0 0
m=audio 13422 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


2023/03/05 00:26:39.948403 192.168.1.99:65265 -> 192.168.1.81:5160
UPDATE sip:192.168.1.81:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:65265;rport;branch=z9hG4bKPjcdd1d69f6a2c4c47a04c0693fe55e638
Max-Forwards: 70
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>;tag=34ac5464-76c6-4efe-95ed-7c97cba8b676
Contact: <sip:[email protected]:65265;ob>
Call-ID: f7901c8c2e664794a57597dbab8f078c
CSeq: 18229 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length:   318

v=0
o=- 3886964793 3886964794 IN IP4 192.168.1.99
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 10018 RTP/AVP 0 101
c=IN IP4 192.168.1.99
b=TIAS:64000
a=rtcp:10019 IN IP4 192.168.1.99
a=ssrc:1306596301 cname:0a5e3dd676d41b58
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv


2023/03/05 00:26:39.951101 192.168.1.81:5160 -> 192.168.1.99:65265
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.99:65265;rport=65265;received=192.168.1.99;branch=z9hG4bKPjcdd1d69f6a2c4c47a04c0693fe55e638
Call-ID: f7901c8c2e664794a57597dbab8f078c
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>;tag=34ac5464-76c6-4efe-95ed-7c97cba8b676
CSeq: 18229 UPDATE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:192.168.1.81:5160>
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: FPBX-16.0.33(18.16.0)
Content-Type: application/sdp
Content-Length:   237

v=0
o=- 3886964793 3886964796 IN IP4 192.168.1.81
s=Asterisk
c=IN IP4 192.168.1.81
t=0 0
m=audio 13422 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


2023/03/05 00:26:39.980278 192.168.1.81:5160 -> 192.168.1.99:65265
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.99:65265;rport=65265;received=192.168.1.99;branch=z9hG4bKPje03b2fd5e68744b08643901e965e38d2
Call-ID: f7901c8c2e664794a57597dbab8f078c
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>;tag=34ac5464-76c6-4efe-95ed-7c97cba8b676
CSeq: 18228 INVITE
Server: FPBX-16.0.33(18.16.0)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Contact: <sip:192.168.1.81:5160>
Content-Type: application/sdp
Content-Length:   261

v=0
o=- 3886964793 3886964795 IN IP4 192.168.1.81
s=Asterisk
c=IN IP4 192.168.1.81
t=0 0
m=audio 13422 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


2023/03/05 00:26:39.986591 192.168.1.99:65265 -> 192.168.1.81:5160
UPDATE sip:192.168.1.81:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:65265;rport;branch=z9hG4bKPjd3083a5c706f41ffa213a232cccc98f3
Max-Forwards: 70
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>;tag=34ac5464-76c6-4efe-95ed-7c97cba8b676
Contact: <sip:[email protected]:65265;ob>
Call-ID: f7901c8c2e664794a57597dbab8f078c
CSeq: 18230 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Length:   318

v=0
o=- 3886964793 3886964796 IN IP4 192.168.1.99
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 10018 RTP/AVP 0 101
c=IN IP4 192.168.1.99
b=TIAS:64000
a=rtcp:10019 IN IP4 192.168.1.99
a=ssrc:1306596301 cname:0a5e3dd676d41b58
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv


2023/03/05 00:26:39.987882 192.168.1.81:5160 -> 192.168.1.99:65265
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.99:65265;rport=65265;received=192.168.1.99;branch=z9hG4bKPjd3083a5c706f41ffa213a232cccc98f3
Call-ID: f7901c8c2e664794a57597dbab8f078c
From: <sip:[email protected]8.1.81>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>;tag=34ac5464-76c6-4efe-95ed-7c97cba8b676
CSeq: 18230 UPDATE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:192.168.1.81:5160>
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: FPBX-16.0.33(18.16.0)
Content-Type: application/sdp
Content-Length:   237

v=0
o=- 3886964793 3886964797 IN IP4 192.168.1.81
s=Asterisk
c=IN IP4 192.168.1.81
t=0 0
m=audio 13422 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


2023/03/05 00:26:46.826659 192.168.1.99:65265 -> 192.168.1.81:5160
CANCEL sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:65265;rport;branch=z9hG4bKPje03b2fd5e68744b08643901e965e38d2
Max-Forwards: 70
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>
Call-ID: f7901c8c2e664794a57597dbab8f078c
CSeq: 18228 CANCEL
User-Agent: MicroSIP/3.21.3
Content-Length:  0



2023/03/05 00:26:46.827463 192.168.1.81:5160 -> 192.168.1.99:65265
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.99:65265;rport=65265;received=192.168.1.99;branch=z9hG4bKPje03b2fd5e68744b08643901e965e38d2
Call-ID: f7901c8c2e664794a57597dbab8f078c
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>;tag=34ac5464-76c6-4efe-95ed-7c97cba8b676
CSeq: 18228 CANCEL
Server: FPBX-16.0.33(18.16.0)
Content-Length:  0



2023/03/05 00:26:46.827659 192.168.1.81:5160 -> 192.168.1.99:65265
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.99:65265;rport=65265;received=192.168.1.99;branch=z9hG4bKPje03b2fd5e68744b08643901e965e38d2
Call-ID: f7901c8c2e664794a57597dbab8f078c
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>;tag=34ac5464-76c6-4efe-95ed-7c97cba8b676
CSeq: 18228 INVITE
Server: FPBX-16.0.33(18.16.0)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Content-Length:  0



2023/03/05 00:26:46.828152 192.168.1.99:65265 -> 192.168.1.81:5160
ACK sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:65265;rport;branch=z9hG4bKPje03b2fd5e68744b08643901e965e38d2
Max-Forwards: 70
From: <sip:[email protected]>;tag=5d93182319b24c35905dace78f6632a3
To: <sip:[email protected]>;tag=34ac5464-76c6-4efe-95ed-7c97cba8b676
Call-ID: f7901c8c2e664794a57597dbab8f078c
CSeq: 18228 ACK
Content-Length:  0



2023/03/05 00:26:51.436964 192.168.1.81:5160 -> 192.168.1.99:65265
OPTIONS sip:[email protected]:65265;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.81:5160;rport;branch=z9hG4bKPjbf5b89e6-1190-4be8-acca-89cc2b185a00
From: <sip:[email protected]>;tag=042c27e0-fc85-4e9b-88bd-0dc2e34c74bf
To: <sip:[email protected];ob>
Contact: <sip:[email protected]:5160>
Call-ID: 9d275f38-c7e6-4309-9332-fcd6528230c3
CSeq: 42696 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.33(18.16.0)
Content-Length:  0



2023/03/05 00:26:51.437833 192.168.1.99:65265 -> 192.168.1.81:5160
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.81:5160;rport=5160;received=192.168.1.81;branch=z9hG4bKPjbf5b89e6-1190-4be8-acca-89cc2b185a00
Call-ID: 9d275f38-c7e6-4309-9332-fcd6528230c3
From: <sip:[email protected]>;tag=042c27e0-fc85-4e9b-88bd-0dc2e34c74bf
To: <sip:[email protected];ob>;tag=z9hG4bKPjbf5b89e6-1190-4be8-acca-89cc2b185a00
CSeq: 42696 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub, trickle-ice
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.21.3
Content-Length:  0

david55 · March 4, 2023, 7:23pm

The request is on 5160. Does this mean you have both channel drivers loaded and are using the legacy driver? If so, note it is unlikely that bugs in chan_sip will get fixed.

Apart from that, it looks like deadlock, but you wouldn’t expect the first call to always deadlock.

Debugging deadlocks is difficult on FreePBX, as the standard binaries aren’t built in a way that makes debugging easy.

I notice it is forcing a change in the direction of session refresh, but don’t see how that would cause problems.

puneet1984 · March 4, 2023, 7:55pm

all my extensions are on PJSIP
my gateway is not chan_sip in peer-peer mode.

5160 port is for the PJSIP
5060 port is for the chansip on the gateway

what i dont understand is that why all of a sudden everything stopped working??

what should i do??
fresh install??
i have installed freepbx on VM on proxmox

BlazeStudios · March 4, 2023, 7:59pm

We need to see the actual trunk side of the call. All we’re seeing is the client side (microsip) of the call. As well you should do asterisk -rvvvvvvvvv and get that output for us when making an outbound call.

puneet1984 · March 4, 2023, 8:05pm

there is no output with asterisk -rvvv command…
for outbound as well for inbound…
but inbound calls are getting through and i can hear IVR but calls are not diverted to designated extensions on press of digits.

david55 · March 4, 2023, 10:34pm

Verbose 3 should be enough, but if one direction is working but the other isn’t, but neither produced dialplan logging, your first problem is the logging.

If, for some reason, you are still using chan_sip on the outbound side, you will need to use “sip set debug on” as well before you can see anything happening on the B leg of the “outbound” call. It is, however, unlikely that there is a valid reason for using chan_sip, and most provider provided chan_sip configurations are somewhat faulty, so only having a chan_sip one from the provider is a questionable reason.

puneet1984 · March 5, 2023, 5:55am

logs are back…it was my mistake… disabled console log…
but calls are still not working

  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
    -- Executing [[email protected]:1] Macro("PJSIP/999-00000009", "user-callerid,LIMIT") in new stack
    -- Executing [[email protected]:1] Set("PJSIP/999-00000009", "TOUCH_MONITOR=1677995656.40") in new stack
    -- Executing [[email protected]:2] Set("PJSIP/999-00000009", "CHANCONTEXT=") in new stack
    -- Executing [[email protected]:3] Progress("PJSIP/999-00000009", "") in new stack
    -- Executing [[email protected]:4] Set("PJSIP/999-00000009", "CHANCONTEXT=") in new stack
    -- Executing [[email protected]:5] Set("PJSIP/999-00000009", "CHANEXTENCONTEXT=999-00000009") in new stack
    -- Executing [[email protected]:6] Set("PJSIP/999-00000009", "CHANEXTEN=999-00000009") in new stack
    -- Executing [[email protected]:7] Set("PJSIP/999-00000009", "CALLERID(number)=999") in new stack
    -- Executing [[email protected]:8] Set("PJSIP/999-00000009", "AMPUSER=999") in new stack
    -- Executing [[email protected]:9] Set("PJSIP/999-00000009", "HOTDESCKCHAN=999-00000009") in new stack
    -- Executing [[email protected]:10] Set("PJSIP/999-00000009", "HOTDESKEXTEN=999") in new stack
    -- Executing [[email protected]:11] Set("PJSIP/999-00000009", "HOTDESKCALL=0") in new stack
    -- Executing [[email protected]:12] ExecIf("PJSIP/999-00000009", "0?Set(HOTDESKCALL=1)") in new stack
    -- Executing [[email protected]:13] ExecIf("PJSIP/999-00000009", "0?Set(CALLERID(name)=)") in new stack
    -- Executing [[email protected]:14] GotoIf("PJSIP/999-00000009", "0?report") in new stack
    -- Executing [[email protected]:15] ExecIf("PJSIP/999-00000009", "1?Set(REALCALLERIDNUM=999)") in new stack
    -- Executing [[email protected]:16] Set("PJSIP/999-00000009", "AMPUSER=999") in new stack
    -- Executing [[email protected]:17] GotoIf("PJSIP/999-00000009", "0?limit") in new stack
    -- Executing [[email protected]:18] Set("PJSIP/999-00000009", "AMPUSERCIDNAME=microsip") in new stack
    -- Executing [[email protected]:19] ExecIf("PJSIP/999-00000009", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
    -- Executing [[email protected]:20] GotoIf("PJSIP/999-00000009", "0?report") in new stack
    -- Executing [[email protected]:21] Set("PJSIP/999-00000009", "AMPUSERCID=999") in new stack
    -- Executing [[email protected]:22] Set("PJSIP/999-00000009", "__DIAL_OPTIONS=HhTtr") in new stack
    -- Executing [[email protected]:23] Set("PJSIP/999-00000009", "CALLERID(all)="microsip" <999>") in new stack
    -- Executing [[email protected]:24] ExecIf("PJSIP/999-00000009", "0?Set(CUSDIAL=)") in new stack
    -- Executing [[email protected]:25] ExecIf("PJSIP/999-00000009", "0?Set(CALLERID(all)="microsip" <999>)") in new stack
    -- Executing [[email protected]:26] GotoIf("PJSIP/999-00000009", "0?limit") in new stack
    -- Executing [[email protected]:27] ExecIf("PJSIP/999-00000009", "1?Set(GROUP(concurrency_limit)=999)") in new stack
    -- Executing [[email protected]:28] ExecIf("PJSIP/999-00000009", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [[email protected]:29] NoOp("PJSIP/999-00000009", "Macro Depth is 1") in new stack
    -- Executing [[email protected]:30] GotoIf("PJSIP/999-00000009", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,31)
    -- Executing [[email protected]:31] GotoIf("PJSIP/999-00000009", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,50)
    -- Executing [[email protected]:50] Set("PJSIP/999-00000009", "CALLERID(number)=999") in new stack
    -- Executing [[email protected]:51] Set("PJSIP/999-00000009", "CALLERID(name)=microsip") in new stack
    -- Executing [[email protected]:52] GotoIf("PJSIP/999-00000009", "0?cnum") in new stack
    -- Executing [[email protected]:53] Set("PJSIP/999-00000009", "CDR(cnam)=microsip") in new stack
    -- Executing [[email protected]:54] Set("PJSIP/999-00000009", "CDR(cnum)=999") in new stack
    -- Executing [[email protected]:55] Set("PJSIP/999-00000009", "CHANNEL(language)=en") in new stack
    -- Executing [[email protected]:2] Set("PJSIP/999-00000009", "ROUTEUSER=999") in new stack
    -- Executing [[email protected]:3] Set("PJSIP/999-00000009", "ROUTEUSER=999") in new stack
    -- Executing [[email protected]:4] GotoIf("PJSIP/999-00000009", "1?notblind") in new stack
    -- Goto (from-internal,9827222070,7)
    -- Executing [[email protected]:7] GotoIf("PJSIP/999-00000009", "1?restrictedroute-cfcd208495d565ef66e7dff9f98764da,9827222070,2:outbound-allroutes,9827222070,2") in new stack
    -- Goto (restrictedroute-cfcd208495d565ef66e7dff9f98764da,9827222070,2)
    -- Channel 'PJSIP/999-00000009' sent to invalid extension: context,exten,priority=restrictedroute-cfcd208495d565ef66e7dff9f98764da,9827222070,2
    -- Executing [[email protected]:1] Goto("PJSIP/999-00000009", "bad-number,s,1") in new stack
    -- Goto (bad-number,s,1)
    -- Executing [[email protected]:1] Goto("PJSIP/999-00000009", "11,1") in new stack
    -- Goto (bad-number,11,1)
    -- Executing [[email protected]mber:1] ResetCDR("PJSIP/999-00000009", "") in new stack
    -- Executing [[email protected]:2] NoCDR("PJSIP/999-00000009", "") in new stack
    -- Executing [[email protected]:3] Progress("PJSIP/999-00000009", "") in new stack
    -- Executing [[email protected]:4] Wait("PJSIP/999-00000009", "1") in new stack
    -- Executing [[email protected]:5] Playback("PJSIP/999-00000009", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
    -- <PJSIP/999-00000009> Playing 'silence/1.ulaw' (language 'en')
    -- <PJSIP/999-00000009> Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
    -- <PJSIP/999-00000009> Playing 'check-number-dial-again.ulaw' (language 'en')

what is restricted route??

  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
    -- Executing [[email protected]:1] Macro("PJSIP/999-0000000b", "user-callerid,LIMIT") in new stack
    -- Executing [[email protected]:1] Set("PJSIP/999-0000000b", "TOUCH_MONITOR=1677995863.42") in new stack
    -- Executing [[email protected]:2] Set("PJSIP/999-0000000b", "CHANCONTEXT=") in new stack
    -- Executing [[email protected]:3] Progress("PJSIP/999-0000000b", "") in new stack
    -- Executing [[email protected]:4] Set("PJSIP/999-0000000b", "CHANCONTEXT=") in new stack
    -- Executing [[email protected]:5] Set("PJSIP/999-0000000b", "CHANEXTENCONTEXT=999-0000000b") in new stack
    -- Executing [[email protected]:6] Set("PJSIP/999-0000000b", "CHANEXTEN=999-0000000b") in new stack
    -- Executing [[email protected]:7] Set("PJSIP/999-0000000b", "CALLERID(number)=999") in new stack
    -- Executing [[email protected]:8] Set("PJSIP/999-0000000b", "AMPUSER=999") in new stack
    -- Executing [[email protected]:9] Set("PJSIP/999-0000000b", "HOTDESCKCHAN=999-0000000b") in new stack
    -- Executing [[email protected]:10] Set("PJSIP/999-0000000b", "HOTDESKEXTEN=999") in new stack
    -- Executing [[email protected]:11] Set("PJSIP/999-0000000b", "HOTDESKCALL=0") in new stack
    -- Executing [[email protected]:12] ExecIf("PJSIP/999-0000000b", "0?Set(HOTDESKCALL=1)") in new stack
    -- Executing [[email protected]:13] ExecIf("PJSIP/999-0000000b", "0?Set(CALLERID(name)=)") in new stack
    -- Executing [[email protected]:14] GotoIf("PJSIP/999-0000000b", "0?report") in new stack
    -- Executing [[email protected]:15] ExecIf("PJSIP/999-0000000b", "1?Set(REALCALLERIDNUM=999)") in new stack
    -- Executing [[email protected]:16] Set("PJSIP/999-0000000b", "AMPUSER=999") in new stack
    -- Executing [[email protected]:17] GotoIf("PJSIP/999-0000000b", "0?limit") in new stack
    -- Executing [[email protected]:18] Set("PJSIP/999-0000000b", "AMPUSERCIDNAME=microsip") in new stack
    -- Executing [[email protected]:19] ExecIf("PJSIP/999-0000000b", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
    -- Executing [[email protected]:20] GotoIf("PJSIP/999-0000000b", "0?report") in new stack
    -- Executing [[email protected]:21] Set("PJSIP/999-0000000b", "AMPUSERCID=999") in new stack
    -- Executing [[email protected]:22] Set("PJSIP/999-0000000b", "__DIAL_OPTIONS=HhTtr") in new stack
    -- Executing [[email protected]:23] Set("PJSIP/999-0000000b", "CALLERID(all)="microsip" <999>") in new stack
    -- Executing [[email protected]:24] ExecIf("PJSIP/999-0000000b", "0?Set(CUSDIAL=)") in new stack
    -- Executing [[email protected]:25] ExecIf("PJSIP/999-0000000b", "0?Set(CALLERID(all)="microsip" <999>)") in new stack
    -- Executing [[email protected]:26] GotoIf("PJSIP/999-0000000b", "0?limit") in new stack
    -- Executing [[email protected]:27] ExecIf("PJSIP/999-0000000b", "1?Set(GROUP(concurrency_limit)=999)") in new stack
    -- Executing [[email protected]:28] ExecIf("PJSIP/999-0000000b", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [[email protected]:29] NoOp("PJSIP/999-0000000b", "Macro Depth is 1") in new stack
    -- Executing [[email protected]:30] GotoIf("PJSIP/999-0000000b", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,31)
    -- Executing [[email protected]:31] GotoIf("PJSIP/999-0000000b", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,50)
    -- Executing [[email protected]:50] Set("PJSIP/999-0000000b", "CALLERID(number)=999") in new stack
    -- Executing [[email protected]:51] Set("PJSIP/999-0000000b", "CALLERID(name)=microsip") in new stack
    -- Executing [[email protected]:52] GotoIf("PJSIP/999-0000000b", "0?cnum") in new stack
    -- Executing [[email protected]:53] Set("PJSIP/999-0000000b", "CDR(cnam)=microsip") in new stack
    -- Executing [[email protected]:54] Set("PJSIP/999-0000000b", "CDR(cnum)=999") in new stack
    -- Executing [[email protected]:55] Set("PJSIP/999-0000000b", "CHANNEL(language)=en") in new stack
    -- Executing [[email protected]:2] Set("PJSIP/999-0000000b", "ROUTEUSER=999") in new stack
    -- Executing [[email protected]:3] Set("PJSIP/999-0000000b", "ROUTEUSER=999") in new stack
    -- Executing [[email protected]:4] GotoIf("PJSIP/999-0000000b", "1?notblind") in new stack
    -- Goto (from-internal,159827222070,7)
    -- Executing [[email protected]:7] GotoIf("PJSIP/999-0000000b", "1?restrictedroute-cfcd208495d565ef66e7dff9f98764da,159827222070,2:outbound-allroutes,159827222070,2") in new stack
    -- Goto (restrictedroute-cfcd208495d565ef66e7dff9f98764da,159827222070,2)
    -- Channel 'PJSIP/999-0000000b' sent to invalid extension: context,exten,priority=restrictedroute-cfcd208495d565ef66e7dff9f98764da,159827222070,2
    -- Executing [[email protected]:1] Goto("PJSIP/999-0000000b", "bad-number,s,1") in new stack
    -- Goto (bad-number,s,1)
    -- Executing [[email protected]:1] Goto("PJSIP/999-0000000b", "11,1") in new stack
    -- Goto (bad-number,11,1)
    -- Executing [[email protected]:1] ResetCDR("PJSIP/999-0000000b", "") in new stack
    -- Executing [[email protected]:2] NoCDR("PJSIP/999-0000000b", "") in new stack
    -- Executing [[email protected]:3] Progress("PJSIP/999-0000000b", "") in new stack
    -- Executing [[email protected]:4] Wait("PJSIP/999-0000000b", "1") in new stack
    -- Executing [[email protected]:5] Playback("PJSIP/999-0000000b", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
    -- <PJSIP/999-0000000b> Playing 'silence/1.ulaw' (language 'en')
    -- <PJSIP/999-0000000b> Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
    -- <PJSIP/999-0000000b> Playing 'check-number-dial-again.ulaw' (language 'en')

puneet1984 · March 5, 2023, 6:04am

hello

got it working again…
problem was blocked list in the outbound routes…

earlier there was no menu in the outbound routes — route A — additional settings — blocked extensions

after i installed extension routing module this additional sub section was added… and by default it blocked all extensions from outbound routes.

system · March 12, 2023, 6:05am

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