Anveo Direct with CHAN_PJSIP trunk. Asterisk 16.6.2 from SNG7-FPBX-64bit-2002-2.iso


(Martin Politick) #1

Hi there,

I’m moving from SipXecs. I have my Cisco 7960 on CHAN_PJSIP and connected that’s all good.
But I’m trying to configure Anveodirect as a PJSIP trunk and my Trunk always seem to want to go through a Sangoma SBC with IP = 104.145.12.182

I’ve followed the post 68522: community.freepbx.org anveo-direct-setup/68522 a few times already.

My public IP = 174.93.175.241
In Menu -> Settings -> Asterisk Sip Settings -> External Address = 174.93.175.241
in ssh:

grep 174.93.175.241 /etc/asterisk/*
pjsip.transports.conf:external_media_address=174.93.175.241
pjsip.transports.conf:external_signaling_address=174.93.175.241
sip_general_additional.conf:externip=174.93.175.241

But the trunk uses tells Anveo to reach me at 104.145.12.182
and it uses a 172.27.0.10 which is on my private network.
I would think that these 2 IPs should be my public address of 174.93.175.241

Do you know what I’ve forgotten to configure ?
Kind Regards,
Martin Politick Jan. 2021

Here are my SIP trace from Anveo CDR:

/*<<<|174.93.175.241:5060 @ 2021-01-22 07:19:01 */
INVITE sip:16047616555@sbc.anveo.com:5060 SIP/2.0
Via: SIP/2.0/UDP 104.145.12.182:5060;rport;branch=z9hG4bKPja4eb9fd7-9cae-4592-9c9b-6edf98a74d3e
From: sip:15816818189@**172.27.0.10**;tag=374d2ca2-4b5e-4a3e-a2d4-fccf2870fda2
To: sip:16047616555@sbc.anveo.com
Contact: sip:asterisk@:5060**104.145.12.182**
Call-ID: 5e246593-2e85-43d2-8eff-a88d95843378
CSeq: 30646 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-15.0.16.42(16.6.2)
Content-Type: application/sdp
Content-Length: 341

v=0
o=- 697861980 697861980 IN IP4 104.145.12.182
s=Asterisk
c=IN IP4 104.145.12.182
t=0 0
m=audio 13476 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

/*>>>|174.93.175.241:5060 @ 2021-01-22 07:19:01 */
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 104.145.12.182:5060;received=174.93.175.241;rport=5060;branch=z9hG4bKPja4eb9fd7-9cae-4592-9c9b-6edf98a74d3e
From: sip:15816818189@172.27.0.10;tag=374d2ca2-4b5e-4a3e-a2d4-fccf2870fda2
To: sip:16047616555@sbc.anveo.com
Call-ID: 5e246593-2e85-43d2-8eff-a88d95843378
CSeq: 30646 INVITE
Server: Anveo Callcontrol
Content-Length: 0

/*>>>|174.93.175.241:5060 @ 2021-01-22 07:19:01 */
SIP/2.0 503 Service Unavailable
Server: Anveo Callcontrol
Via: SIP/2.0/UDP 104.145.12.182:5060;received=174.93.175.241;branch=z9hG4bKPja4eb9fd7-9cae-4592-9c9b-6edf98a74d3e;rport=5060
From: sip:15816818189@172.27.0.10;tag=374d2ca2-4b5e-4a3e-a2d4-fccf2870fda2
To: sip:16047616555@sbc.anveo.com;tag=3b15a6585e026902eff9531553f11965
Call-ID: 5e246593-2e85-43d2-8eff-a88d95843378
CSeq: 30646 INVITE
Server: Anveo Callcontrol
Reason: SIP; cause=503; text=“Service Unavailable”
Content-Length: 0

/*<<<|174.93.175.241:5060 @ 2021-01-22 07:19:01 */
ACK sip:16047616555@sbc.anveo.com:5060 SIP/2.0
Via: SIP/2.0/UDP 104.145.12.182:5060;rport;branch=z9hG4bKPja4eb9fd7-9cae-4592-9c9b-6edf98a74d3e
From: sip:15816818189@172.27.0.10;tag=374d2ca2-4b5e-4a3e-a2d4-fccf2870fda2
To: sip:16047616555@sbc.anveo.com;tag=3b15a6585e026902eff9531553f11965
Call-ID: 5e246593-2e85-43d2-8eff-a88d95843378
CSeq: 30646 ACK
Max-Forwards: 70
User-Agent: FPBX-15.0.16.42(16.6.2)
Content-Length: 0


#2

You must restart (not just reload) Asterisk after changing these settings.


(Martin Politick) #3

You are right, that fixed the 104.145.12.182 Sangoma IP problem.
My mistake, I thought I rebooted the VM, but obviously I did not.

But my outboud call still fails with Anveo Direct with a:

SIP/2.0 503 Service Unavailable

I know very little in SIP signaling, but does it matter if the

From: sip:15816818189@172.27.0.10
has a private IP address in it ? Or is it like the “from” in an email and it does not need to match the actual email address?

Or do you know why or what causes a “SIP/2.0 503 Service Unavailable” ?

Kind Regards,
Martin Politick.

FULL ANVEO TRACE

/*<<<|174.93.175.241:5060 @ 2021-01-23 05:30:47 */
INVITE sip:16047616555@sbc.anveo.com:5060 SIP/2.0
Via: SIP/2.0/UDP 174.93.175.241:5060;rport;branch=z9hG4bKPj2fde4a1e-fb59-49a9-a886-f88a56f00920
From: sip:15816818189@172.27.0.10;tag=8caf6b8b-480d-46e6-8462-1d37b7311b04
To: sip:16047616555@sbc.anveo.com
Contact: sip:asterisk@174.93.175.241:5060
Call-ID: 223e8dfe-f2a8-4826-a308-ccc832fafd36
CSeq: 16329 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-15.0.16.42(16.6.2)
Content-Type: application/sdp
Content-Length: 343

v=0
o=- 1171049796 1171049796 IN IP4 174.93.175.241
s=Asterisk
c=IN IP4 174.93.175.241
t=0 0
m=audio 18766 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

/*>>>|174.93.175.241:5060 @ 2021-01-23 05:30:47 */
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 174.93.175.241:5060;rport=5060;branch=z9hG4bKPj2fde4a1e-fb59-49a9-a886-f88a56f00920
From: sip:15816818189@172.27.0.10;tag=8caf6b8b-480d-46e6-8462-1d37b7311b04
To: sip:16047616555@sbc.anveo.com
Call-ID: 223e8dfe-f2a8-4826-a308-ccc832fafd36
CSeq: 16329 INVITE
Server: Anveo Callcontrol
Content-Length: 0

/*>>>|174.93.175.241:5060 @ 2021-01-23 05:30:47 */
SIP/2.0 503 Service Unavailable
Server: Anveo Callcontrol
Via: SIP/2.0/UDP 174.93.175.241:5060;branch=z9hG4bKPj2fde4a1e-fb59-49a9-a886-f88a56f00920;rport=5060
From: sip:15816818189@172.27.0.10;tag=8caf6b8b-480d-46e6-8462-1d37b7311b04
To: sip:16047616555@sbc.anveo.com;tag=315f68fe8becb9b8fa872e3305232146
Call-ID: 223e8dfe-f2a8-4826-a308-ccc832fafd36
CSeq: 16329 INVITE
Server: Anveo Callcontrol
Reason: SIP; cause=503; text=“Service Unavailable”
Content-Length: 0

/*<<<|174.93.175.241:5060 @ 2021-01-23 05:30:47 */
ACK sip:16047616555@sbc.anveo.com:5060 SIP/2.0
Via: SIP/2.0/UDP 174.93.175.241:5060;rport;branch=z9hG4bKPj2fde4a1e-fb59-49a9-a886-f88a56f00920
From: sip:15816818189@172.27.0.10;tag=8caf6b8b-480d-46e6-8462-1d37b7311b04
To: sip:16047616555@sbc.anveo.com;tag=315f68fe8becb9b8fa872e3305232146
Call-ID: 223e8dfe-f2a8-4826-a308-ccc832fafd36
CSeq: 16329 ACK
Max-Forwards: 70
User-Agent: FPBX-15.0.16.42(16.6.2)
Content-Length: 0

Full ASTERISK trace:

<— Received SIP request (955 bytes) from UDP:192.168.10.94:51408 —>
INVITE sip:16047616555@172.27.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.94:5060;branch=z9hG4bK5976d8ce
From: “Martin” sip:222@172.27.0.10;tag=000f24c6a21a0a5c5de0c4e9-1ef5654c
To: sip:16047616555@172.27.0.10
Call-ID: 000f24c6-a21a000b-3aa41f0e-1c641f4d@192.168.10.94
Max-Forwards: 70
Date: Sat, 23 Jan 2021 04:19:55 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: sip:222@192.168.10.94:5060;transport=udp
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 278
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 19268 0 IN IP4 192.168.10.94
s=SIP Call
t=0 0
m=audio 17286 RTP/AVP 0 8 18 101
c=IN IP4 192.168.10.94
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<— Transmitting SIP response (519 bytes) to UDP:192.168.10.94:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.94:5060;received=192.168.10.94;branch=z9hG4bK5976d8ce
Call-ID: 000f24c6-a21a000b-3aa41f0e-1c641f4d@192.168.10.94
From: “Martin” sip:222@172.27.0.10;tag=000f24c6a21a0a5c5de0c4e9-1ef5654c
To: sip:16047616555@172.27.0.10;tag=z9hG4bK5976d8ce
CSeq: 101 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1611375591/545476cb75062cda728b5bdabea28fb6”,opaque=“627ea8784af17a39”,algorithm=md5,qop=“auth”
Server: FPBX-15.0.16.42(16.6.2)
Content-Length: 0

<— Received SIP request (365 bytes) from UDP:192.168.10.94:51409 —>
ACK sip:16047616555@172.27.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.94:5060;branch=z9hG4bK5976d8ce
From: “Martin” sip:222@172.27.0.10;tag=000f24c6a21a0a5c5de0c4e9-1ef5654c
To: sip:16047616555@172.27.0.10;tag=z9hG4bK5976d8ce
Call-ID: 000f24c6-a21a000b-3aa41f0e-1c641f4d@192.168.10.94
Date: Sat, 23 Jan 2021 04:19:55 GMT
CSeq: 101 ACK
Content-Length: 0

<— Received SIP request (1219 bytes) from UDP:192.168.10.94:51410 —>
INVITE sip:16047616555@172.27.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.94:5060;branch=z9hG4bK59fddba9
From: “Martin” sip:222@172.27.0.10;tag=000f24c6a21a0a5c5de0c4e9-1ef5654c
To: sip:16047616555@172.27.0.10
Call-ID: 000f24c6-a21a000b-3aa41f0e-1c641f4d@192.168.10.94
Max-Forwards: 70
Date: Sat, 23 Jan 2021 04:19:55 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: sip:222@192.168.10.94:5060;transport=udp
Authorization: Digest username=“222”,realm=“asterisk”,uri="sip:16047616555@172.27.0.10",response=“ff77d55393ba626694fc75d6cb190eed”,nonce=“1611375591/545476cb75062cda728b5bdabea28fb6”,opaque=“627ea8784af17a39”,cnonce=“1d2250af”,qop=auth,nc=00000001,algorithm=md5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 278
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 19268 0 IN IP4 192.168.10.94
s=SIP Call
t=0 0
m=audio 17286 RTP/AVP 0 8 18 101
c=IN IP4 192.168.10.94
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

== Setting global variable ‘SIPDOMAIN’ to ‘172.27.0.10’
<— Transmitting SIP response (347 bytes) to UDP:192.168.10.94:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.94:5060;received=192.168.10.94;branch=z9hG4bK59fddba9
Call-ID: 000f24c6-a21a000b-3aa41f0e-1c641f4d@192.168.10.94
From: “Martin” sip:222@172.27.0.10;tag=000f24c6a21a0a5c5de0c4e9-1ef5654c
To: sip:16047616555@172.27.0.10
CSeq: 102 INVITE
Server: FPBX-15.0.16.42(16.6.2)
Content-Length: 0

== 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 [16047616555@from-internal:1] Macro(“PJSIP/222-00000002”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“PJSIP/222-00000002”, “TOUCH_MONITOR=1611375591.2”) in new stack
– Executing [s@macro-user-callerid:2] Set(“PJSIP/222-00000002”, “AMPUSER=222”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“PJSIP/222-00000002”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“PJSIP/222-00000002”, “1?Set(REALCALLERIDNUM=222)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“PJSIP/222-00000002”, “AMPUSER=222”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“PJSIP/222-00000002”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“PJSIP/222-00000002”, “AMPUSERCIDNAME=Martin Politick”) in new stack
– Executing [s@macro-user-callerid:8] ExecIf(“PJSIP/222-00000002”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“PJSIP/222-00000002”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“PJSIP/222-00000002”, “AMPUSERCID=222”) in new stack
– Executing [s@macro-user-callerid:11] Set(“PJSIP/222-00000002”, “__DIAL_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-user-callerid:12] Set(“PJSIP/222-00000002”, “CALLERID(all)=“Martin Politick” <222>”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“PJSIP/222-00000002”, “0?Set(CALLERID(all)=EXTERNAL)”) in new stack
– Executing [s@macro-user-callerid:14] GotoIf(“PJSIP/222-00000002”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:15] ExecIf(“PJSIP/222-00000002”, “1?Set(GROUP(concurrency_limit)=222)”) in new stack
– Executing [s@macro-user-callerid:16] ExecIf(“PJSIP/222-00000002”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:17] NoOp(“PJSIP/222-00000002”, “Macro Depth is 1”) in new stack
– Executing [s@macro-user-callerid:18] GotoIf(“PJSIP/222-00000002”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] GotoIf(“PJSIP/222-00000002”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,38)
– Executing [s@macro-user-callerid:38] Set(“PJSIP/222-00000002”, “CALLERID(number)=222”) in new stack
– Executing [s@macro-user-callerid:39] Set(“PJSIP/222-00000002”, “CALLERID(name)=Martin Politick”) in new stack
– Executing [s@macro-user-callerid:40] GotoIf(“PJSIP/222-00000002”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:41] Set(“PJSIP/222-00000002”, “CDR(cnam)=Martin Politick”) in new stack
– Executing [s@macro-user-callerid:42] Set(“PJSIP/222-00000002”, “CDR(cnum)=222”) in new stack
– Executing [s@macro-user-callerid:43] Set(“PJSIP/222-00000002”, “CHANNEL(language)=en”) in new stack
– Executing [16047616555@from-internal:2] Gosub(“PJSIP/222-00000002”, “sub-record-check,s,1(out,16047616555,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“PJSIP/222-00000002”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“PJSIP/222-00000002”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“PJSIP/222-00000002”, “NOW=1611375591”) in new stack
– Executing [s@sub-record-check:4] Set(“PJSIP/222-00000002”, “__DAY=22”) in new stack
– Executing [s@sub-record-check:5] Set(“PJSIP/222-00000002”, “__MONTH=01”) in new stack
– Executing [s@sub-record-check:6] Set(“PJSIP/222-00000002”, “__YEAR=2021”) in new stack
– Executing [s@sub-record-check:7] Set(“PJSIP/222-00000002”, “__TIMESTR=20210122-201951”) in new stack
– Executing [s@sub-record-check:8] Set(“PJSIP/222-00000002”, “__FROMEXTEN=222”) in new stack
– Executing [s@sub-record-check:9] Set(“PJSIP/222-00000002”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“PJSIP/222-00000002”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“PJSIP/222-00000002”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“PJSIP/222-00000002”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“PJSIP/222-00000002”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“PJSIP/222-00000002”, “3?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“PJSIP/222-00000002”, “1?sub-record-check,out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] NoOp(“PJSIP/222-00000002”, “Outbound Recording Check from 222 to 16047616555”) in new stack
– Executing [out@sub-record-check:2] Set(“PJSIP/222-00000002”, “RECMODE=dontcare”) in new stack
– Executing [out@sub-record-check:3] ExecIf(“PJSIP/222-00000002”, “1?Goto(routewins)”) in new stack
– Goto (sub-record-check,out,7)
– Executing [out@sub-record-check:7] Gosub(“PJSIP/222-00000002”, “recordcheck,1(dontcare,out,16047616555)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/222-00000002”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/222-00000002”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“PJSIP/222-00000002”, “”) in new stack
– Executing [out@sub-record-check:8] Return(“PJSIP/222-00000002”, “”) in new stack
– Executing [16047616555@from-internal:3] ExecIf(“PJSIP/222-00000002”, “0 ?Set(CDR(accountcode)=)”) in new stack
– Executing [16047616555@from-internal:4] Set(“PJSIP/222-00000002”, “MOHCLASS=default”) in new stack
– Executing [16047616555@from-internal:5] Set(“PJSIP/222-00000002”, “_NODEST=”) in new stack
– Executing [16047616555@from-internal:6] Macro(“PJSIP/222-00000002”, “dialout-trunk,1,16047616555,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“PJSIP/222-00000002”, “DIAL_TRUNK=1”) in new stack
– Executing [s@macro-dialout-trunk:2] ExecIf(“PJSIP/222-00000002”, “0?Set(DIAL_OPTIONS=Hhtr)”) in new stack
– Executing [s@macro-dialout-trunk:3] GosubIf(“PJSIP/222-00000002”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:4] ExecIf(“PJSIP/222-00000002”, “0?Set(CALLERID(num)=222)”) in new stack
– Executing [s@macro-dialout-trunk:5] GotoIf(“PJSIP/222-00000002”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“PJSIP/222-00000002”, “DIAL_NUMBER=16047616555”) in new stack
– Executing [s@macro-dialout-trunk:7] Set(“PJSIP/222-00000002”, “DIAL_TRUNK_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dialout-trunk:8] Set(“PJSIP/222-00000002”, “OUTBOUND_GROUP=OUT_1”) in new stack
– Executing [s@macro-dialout-trunk:9] Set(“PJSIP/222-00000002”, “DIAL_TRUNK_OPTIONS=T”) in new stack
– Executing [s@macro-dialout-trunk:10] GotoIf(“PJSIP/222-00000002”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,12)
– Executing [s@macro-dialout-trunk:12] GotoIf(“PJSIP/222-00000002”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:13] Macro(“PJSIP/222-00000002”, “outbound-callerid,1”) in new stack
– Executing [s@macro-outbound-callerid:1] NoOp(“PJSIP/222-00000002”, “222”) in new stack
– Executing [s@macro-outbound-callerid:2] NoOp(“PJSIP/222-00000002”, “”) in new stack
– Executing [s@macro-outbound-callerid:3] NoOp(“PJSIP/222-00000002”, “off”) in new stack
– Executing [s@macro-outbound-callerid:4] ExecIf(“PJSIP/222-00000002”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:5] ExecIf(“PJSIP/222-00000002”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:6] ExecIf(“PJSIP/222-00000002”, “0?Set(REALCALLERIDNUM=222)”) in new stack
– Executing [s@macro-outbound-callerid:7] ExecIf(“PJSIP/222-00000002”, “0?Set(AMPUSER=222)”) in new stack
– Executing [s@macro-outbound-callerid:8] GotoIf(“PJSIP/222-00000002”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] Set(“PJSIP/222-00000002”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:13] Set(“PJSIP/222-00000002”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:14] Set(“PJSIP/222-00000002”, “TRUNKOUTCID=15816818189”) in new stack
– Executing [s@macro-outbound-callerid:15] GotoIf(“PJSIP/222-00000002”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,21)
– Executing [s@macro-outbound-callerid:21] ExecIf(“PJSIP/222-00000002”, “1?Set(CALLERID(all)=15816818189)”) in new stack
– Executing [s@macro-outbound-callerid:22] ExecIf(“PJSIP/222-00000002”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:23] ExecIf(“PJSIP/222-00000002”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:24] ExecIf(“PJSIP/222-00000002”, “0?Set(CALLERID(all)=222)”) in new stack
– Executing [s@macro-outbound-callerid:25] Set(“PJSIP/222-00000002”, “TIOHIDE=no”) in new stack
– Executing [s@macro-outbound-callerid:26] ExecIf(“PJSIP/222-00000002”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:27] ExecIf(“PJSIP/222-00000002”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:28] ExecIf(“PJSIP/222-00000002”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:29] ExecIf(“PJSIP/222-00000002”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:30] Set(“PJSIP/222-00000002”, “CDR(outbound_cnum)=15816818189”) in new stack
– Executing [s@macro-outbound-callerid:31] Set(“PJSIP/222-00000002”, “CDR(outbound_cnam)=”) in new stack
– Executing [s@macro-dialout-trunk:14] GosubIf(“PJSIP/222-00000002”, “0?sub-flp-1,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:15] Set(“PJSIP/222-00000002”, “OUTNUM=16047616555”) in new stack
– Executing [s@macro-dialout-trunk:16] Set(“PJSIP/222-00000002”, “custom=PJSIP”) in new stack
– Executing [s@macro-dialout-trunk:17] ExecIf(“PJSIP/222-00000002”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
– Executing [s@macro-dialout-trunk:18] ExecIf(“PJSIP/222-00000002”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:19] Macro(“PJSIP/222-00000002”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“PJSIP/222-00000002”, “”) in new stack
– Executing [s@macro-dialout-trunk:20] GotoIf(“PJSIP/222-00000002”, “0?skipcrm”) in new stack
– Executing [s@macro-dialout-trunk:21] Set(“PJSIP/222-00000002”, “__CRM_DIRECTION=OUTBOUND”) in new stack
– Executing [s@macro-dialout-trunk:22] Set(“PJSIP/222-00000002”, “__CRM_DESTINATION=16047616555”) in new stack
– Executing [s@macro-dialout-trunk:23] Set(“PJSIP/222-00000002”, “__CRM_SOURCE=222”) in new stack
– Executing [s@macro-dialout-trunk:24] AGI(“PJSIP/222-00000002”, “agi://127.0.0.1/sangomacrm.agi”) in new stack
– <PJSIP/222-00000002>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
– Executing [s@macro-dialout-trunk:25] Set(“PJSIP/222-00000002”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
– Executing [s@macro-dialout-trunk:26] NoOp(“PJSIP/222-00000002”, “CRM Finished”) in new stack
– Executing [s@macro-dialout-trunk:27] GotoIf(“PJSIP/222-00000002”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:28] ExecIf(“PJSIP/222-00000002”, “1?Set(CONNECTEDLINE(num,i)=16047616555)”) in new stack
– Executing [s@macro-dialout-trunk:29] ExecIf(“PJSIP/222-00000002”, “1?Set(CONNECTEDLINE(name,i)=CID:15816818189)”) in new stack
– Executing [s@macro-dialout-trunk:30] ExecIf(“PJSIP/222-00000002”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)15816818189)”) in new stack
– Executing [s@macro-dialout-trunk:31] GotoIf(“PJSIP/222-00000002”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:32] ExecIf(“PJSIP/222-00000002”, “0?Set(DIAL_TRUNK_OPTIONS=)”) in new stack
– Executing [s@macro-dialout-trunk:33] Dial(“PJSIP/222-00000002”, “PJSIP/16047616555@Anveo_Direct,300,Tb(func-apply-sipheaders^s^1,(1))”) in new stack
– PJSIP/Anveo_Direct-00000003 Internal Gosub(func-apply-sipheaders,s,1(1)) start
– Executing [s@func-apply-sipheaders:1] ExecIf(“PJSIP/Anveo_Direct-00000003”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
– Executing [s@func-apply-sipheaders:2] NoOp(“PJSIP/Anveo_Direct-00000003”, “Applying SIP Headers to channel PJSIP/Anveo_Direct-00000003”) in new stack
– Executing [s@func-apply-sipheaders:3] Set(“PJSIP/Anveo_Direct-00000003”, “TECH=PJSIP”) in new stack
– Executing [s@func-apply-sipheaders:4] Set(“PJSIP/Anveo_Direct-00000003”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:5] While(“PJSIP/Anveo_Direct-00000003”, “0”) in new stack
– Jumping to priority 13
– Executing [s@func-apply-sipheaders:14] Return(“PJSIP/Anveo_Direct-00000003”, “”) in new stack
== Spawn extension (from-pstn, 16047616555, 1) exited non-zero on ‘PJSIP/Anveo_Direct-00000003’
– PJSIP/Anveo_Direct-00000003 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
– Called PJSIP/16047616555@Anveo_Direct
<— Transmitting SIP request (1023 bytes) to UDP:169.48.232.158:5060 —>
INVITE sip:16047616555@sbc.anveo.com:5060 SIP/2.0
Via: SIP/2.0/UDP 174.93.175.241:5060;rport;branch=z9hG4bKPj4adf83af-e7f9-4da0-9157-2caa639b5aa7
From: sip:15816818189@172.27.0.10;tag=83b7c6a5-f946-4317-8178-43bc0d8e1501
To: sip:16047616555@sbc.anveo.com
Contact: sip:asterisk@174.93.175.241:5060
Call-ID: 22cd9017-ee8e-4a79-ba06-c3ce564f4ae5
CSeq: 25031 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-15.0.16.42(16.6.2)
Content-Type: application/sdp
Content-Length: 343

v=0
o=- 1755389975 1755389975 IN IP4 174.93.175.241
s=Asterisk
c=IN IP4 174.93.175.241
t=0 0
m=audio 16172 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Received SIP response (352 bytes) from UDP:169.48.232.158:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 174.93.175.241:5060;rport=5060;branch=z9hG4bKPj4adf83af-e7f9-4da0-9157-2caa639b5aa7
From: sip:15816818189@172.27.0.10;tag=83b7c6a5-f946-4317-8178-43bc0d8e1501
To: sip:16047616555@sbc.anveo.com
Call-ID: 22cd9017-ee8e-4a79-ba06-c3ce564f4ae5
CSeq: 25031 INVITE
Server: Anveo Callcontrol
Content-Length: 0

<— Received SIP response (481 bytes) from UDP:169.48.232.158:5060 —>
SIP/2.0 503 Service Unavailable
Server: Anveo Callcontrol
Via: SIP/2.0/UDP 174.93.175.241:5060;branch=z9hG4bKPj4adf83af-e7f9-4da0-9157-2caa639b5aa7;rport=5060
From: sip:15816818189@172.27.0.10;tag=83b7c6a5-f946-4317-8178-43bc0d8e1501
To: sip:16047616555@sbc.anveo.com;tag=f888121cc310cdc49ab576d89c23494d
Call-ID: 22cd9017-ee8e-4a79-ba06-c3ce564f4ae5
CSeq: 25031 INVITE
Server: Anveo Callcontrol
Reason: SIP; cause=503; text=“Service Unavailable”
Content-Length: 0

<— Transmitting SIP request (438 bytes) to UDP:169.48.232.158:5060 —>
ACK sip:16047616555@sbc.anveo.com:5060 SIP/2.0
Via: SIP/2.0/UDP 174.93.175.241:5060;rport;branch=z9hG4bKPj4adf83af-e7f9-4da0-9157-2caa639b5aa7
From: sip:15816818189@172.27.0.10;tag=83b7c6a5-f946-4317-8178-43bc0d8e1501
To: sip:16047616555@sbc.anveo.com;tag=f888121cc310cdc49ab576d89c23494d
Call-ID: 22cd9017-ee8e-4a79-ba06-c3ce564f4ae5
CSeq: 25031 ACK
Max-Forwards: 70
User-Agent: FPBX-15.0.16.42(16.6.2)
Content-Length: 0

== Everyone is busy/congested at this time (1:0/1/0)
– Executing [s@macro-dialout-trunk:34] NoOp(“PJSIP/222-00000002”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34”) in new stack
– Executing [s@macro-dialout-trunk:35] GotoIf(“PJSIP/222-00000002”, “0?continue,1:s-CONGESTION,1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing [s-CONGESTION@macro-dialout-trunk:1] Set(“PJSIP/222-00000002”, “RC=34”) in new stack
– Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(“PJSIP/222-00000002”, “34,1”) in new stack
– Goto (macro-dialout-trunk,34,1)
– Executing [34@macro-dialout-trunk:1] Goto(“PJSIP/222-00000002”, “continue,1”) in new stack
– Goto (macro-dialout-trunk,continue,1)
– Executing [continue@macro-dialout-trunk:1] NoOp(“PJSIP/222-00000002”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:2] ExecIf(“PJSIP/222-00000002”, “1?Set(CALLERID(number)=222)”) in new stack
– Executing [16047616555@from-internal:7] Macro(“PJSIP/222-00000002”, “dialout-trunk,2,16047616555,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“PJSIP/222-00000002”, “DIAL_TRUNK=2”) in new stack
– Executing [s@macro-dialout-trunk:2] ExecIf(“PJSIP/222-00000002”, “0?Set(DIAL_OPTIONS=Hhtr)”) in new stack
– Executing [s@macro-dialout-trunk:3] GosubIf(“PJSIP/222-00000002”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:4] ExecIf(“PJSIP/222-00000002”, “0?Set(CALLERID(num)=222)”) in new stack
– Executing [s@macro-dialout-trunk:5] GotoIf(“PJSIP/222-00000002”, “1?disabletrunk,1”) in new stack
– Goto (macro-dialout-trunk,disabletrunk,1)
– Executing [disabletrunk@macro-dialout-trunk:1] NoOp(“PJSIP/222-00000002”, “TRUNK: SIP/Anveo_Out_CHAN-SIP DISABLED - falling through to next trunk”) in new stack
– Executing [16047616555@from-internal:8] Macro(“PJSIP/222-00000002”, “outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“PJSIP/222-00000002”, “”) in new stack
– Executing [s@macro-outisbusy:2] GotoIf(“PJSIP/222-00000002”, “0?emergency,1”) in new stack
– Executing [s@macro-outisbusy:3] GotoIf(“PJSIP/222-00000002”, “0?intracompany,1”) in new stack
– Executing [s@macro-outisbusy:4] Playback(“PJSIP/222-00000002”, “all-circuits-busy-now&please-try-call-later, noanswer”) in new stack
> 0x7f6584326470 – Strict RTP learning after remote address set to: 192.168.10.94:17286
<— Transmitting SIP response (891 bytes) to UDP:192.168.10.94:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.94:5060;received=192.168.10.94;branch=z9hG4bK59fddba9
Call-ID: 000f24c6-a21a000b-3aa41f0e-1c641f4d@192.168.10.94
From: “Martin” sip:222@172.27.0.10;tag=000f24c6a21a0a5c5de0c4e9-1ef5654c
To: sip:16047616555@172.27.0.10;tag=8270fdba-e9fb-4410-8d68-fc9b953206ab
CSeq: 102 INVITE
Server: FPBX-15.0.16.42(16.6.2)
Contact: sip:172.27.0.10:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
P-Asserted-Identity: “CID:15816818189” sip:16047616555@172.27.0.10
Content-Type: application/sdp
Content-Length: 245

v=0
o=- 19268 2 IN IP4 172.27.0.10
s=Asterisk
c=IN IP4 172.27.0.10
t=0 0
m=audio 19074 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

-- <PJSIP/222-00000002> Playing 'all-circuits-busy-now.ulaw' (language 'en')
   > 0x7f6584326470 -- Strict RTP switching to RTP target address 192.168.10.94:17286 as source
-- <PJSIP/222-00000002> Playing 'please-try-call-later.ulaw' (language 'en')

<— Received SIP request (664 bytes) from UDP:192.168.10.94:51411 —>
CANCEL sip:16047616555@172.27.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.94:5060;branch=z9hG4bK59fddba9
From: “Martin” sip:222@172.27.0.10;tag=000f24c6a21a0a5c5de0c4e9-1ef5654c
To: sip:16047616555@172.27.0.10
Call-ID: 000f24c6-a21a000b-3aa41f0e-1c641f4d@192.168.10.94
Max-Forwards: 70
Date: Sat, 23 Jan 2021 04:19:58 GMT
CSeq: 102 CANCEL
User-Agent: Cisco-CP7960G/8.0
Content-Length: 0
Authorization: Digest username=“222”,realm=“asterisk”,uri="sip:16047616555@172.27.0.10",response=“d7fe06fa999d23e152ba6f660eb607fe”,nonce=“1611375591/545476cb75062cda728b5bdabea28fb6”,opaque=“627ea8784af17a39”,cnonce=“3987bd13”,qop=auth,nc=00000002,algorithm=md5

<— Transmitting SIP response (384 bytes) to UDP:192.168.10.94:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.94:5060;received=192.168.10.94;branch=z9hG4bK59fddba9
Call-ID: 000f24c6-a21a000b-3aa41f0e-1c641f4d@192.168.10.94
From: “Martin” sip:222@172.27.0.10;tag=000f24c6a21a0a5c5de0c4e9-1ef5654c
To: sip:16047616555@172.27.0.10;tag=8270fdba-e9fb-4410-8d68-fc9b953206ab
CSeq: 102 CANCEL
Server: FPBX-15.0.16.42(16.6.2)
Content-Length: 0

<— Transmitting SIP response (581 bytes) to UDP:192.168.10.94:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.10.94:5060;received=192.168.10.94;branch=z9hG4bK59fddba9
Call-ID: 000f24c6-a21a000b-3aa41f0e-1c641f4d@192.168.10.94
From: “Martin” sip:222@172.27.0.10;tag=000f24c6a21a0a5c5de0c4e9-1ef5654c
To: sip:16047616555@172.27.0.10;tag=8270fdba-e9fb-4410-8d68-fc9b953206ab
CSeq: 102 INVITE
Server: FPBX-15.0.16.42(16.6.2)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
P-Asserted-Identity: “CID:15816818189” sip:16047616555@172.27.0.10
Content-Length: 0

-- Executing [h@from-internal:1] Macro("PJSIP/222-00000002", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/222-00000002", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/222-00000002", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] NoOp("PJSIP/222-00000002", " montior file= ") in new stack
-- Executing [s@macro-hangupcall:5] GotoIf("PJSIP/222-00000002", "1?skipagi") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] Hangup("PJSIP/222-00000002", "") in new stack

== Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/222-00000002’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/222-00000002’
– PJSIP/222-00000002 Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“PJSIP/222-00000002”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“PJSIP/222-00000002”, “HANGUP CAUSE: 34”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“PJSIP/222-00000002”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“PJSIP/222-00000002”, “MASTER CHANNEL: 1611375591.2 = 1611375591.2”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“PJSIP/222-00000002”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“PJSIP/222-00000002”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“PJSIP/222-00000002”, “agi://127.0.0.1/sangomacrm.agi”) in new stack
– <PJSIP/222-00000002>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“PJSIP/222-00000002”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/222-00000002’
– PJSIP/222-00000002 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
<— Received SIP request (386 bytes) from UDP:192.168.10.94:51412 —>
ACK sip:16047616555@172.27.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.94:5060;branch=z9hG4bK59fddba9
From: “Martin” sip:222@172.27.0.10;tag=000f24c6a21a0a5c5de0c4e9-1ef5654c
To: sip:16047616555@172.27.0.10;tag=8270fdba-e9fb-4410-8d68-fc9b953206ab
Call-ID: 000f24c6-a21a000b-3aa41f0e-1c641f4d@192.168.10.94
Date: Sat, 23 Jan 2021 04:19:58 GMT
CSeq: 102 ACK
Content-Length: 0

<— Transmitting SIP request (435 bytes) to UDP:169.48.232.158:5060 —>
OPTIONS sip:sbc.anveo.com:5060 SIP/2.0
Via: SIP/2.0/UDP 174.93.175.241:5060;rport;branch=z9hGa4
From: sip:Anveo_Direct@172.27.0.10;tag=f58a5a
To: sip:sbc.anveo.com
Contact: sip:Anveo_Direct@174.93.175.241:5060
Call-ID: 7063f918-257b-4b73-8ab2-4854c571193b
CSeq: 28091 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.16.42(16.6.2)
Content-Length: 0

<— Received SIP response (375 bytes) from UDP:169.48.232.158:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.93.175.241:5060;rport=5060;branch=z9hG4bKPj08e919ea-49b2-4568-b11a-36231b2d7ba4
From: sip:Anveo_Direct@172.27.0.10;tag=f58a
To: sip:sbc.anveo.com;tag=17865
Call-ID: 7063f918-257
CSeq: 28091 OPTIONS
Server: Anveo Callcontrol
Content-Length: 0


#4

The SIP sent to Anveo looks fine, unless perhaps you configured a Dialing Prefix for the trunk at Anveo, in which case you must set a matching Outbound Dial Prefix in the FreePBX Trunk settings.

Otherwise, the problem is likely with your Anveo settings. Confirm that your public IP address is in the Authorized IP Addresses list. Use their Call Simulator to confirm that you don’t have a problem with the carrier list, rate cap, etc. Check the Account Limits section and confirm that you have a positive balance.


(Martin Politick) #5

Thank you @Stewart1, I’ve verified all of that. I think I’ll have to use of the 4 free tech. tickets.
I’ve completely removed the Dial prefix (to remove one thing out of the equation. Yes my IP is in the authorized list. From my tests, when it’s not, I don’t even get a Call Record and no SIP trace (which is logical).

I’ve used the simulator and no, I had removed the cap and I didn’t exclude any carrier (Is that what you meant by the carrier list ?). I have more than 50$ in the account.
AnveoSettings

Kind Regards,
Martin Politick, Jan 2021


#6

Try setting your failover depth to 8 and retest. I don’t have much faith in some of the international carriers, even those Anveo calls Prime.

2021-01-24

1015 is Airtel, 1001 is Lanck, 24 is Alcazar; I’m not snowed by any of them (for Canada).
25 is Bandwidth (good) and 27 is Iristel (they’re a Canadian carrier and that’s their business.)

If you still have trouble, post another SIP trace from the Anveo end.


(Martin Politick) #7

YES!!!
Thx @Stewart1.

That was it, a strange behavior for Anveo to allow no fail-over then …
AnveoSettingsRouting
Thank you, now I will hardened the outbound and add the prefix & cap.

Thank you again @Stewart1, for fixing a Anveo setting in the end.

Kind Regards,
Martin Politick. Jan 2021


#8

That’s there so you can do the failover by specifying additional trunks in your Outbound Route, which could be a more reliable (and more expensive) provider, or another Anveo trunk with more conservative carrier selection.

Unfortunately, Anveo does not distinguish failover due to timeout from failover where the carrier rejected the call. FreePBX also lacks a built-in way to make this distinction. I’ve learned from experience that if there are two or more timeouts, the end user will likely assume the line is dead and hang up, even though the call would eventually succeed.

So, my system gives Anveo only one shot (from a conservatively selected list of carriers) and fails over to a high end provider. I typically see about 2% of calls failing at Anveo. For example, if I’m paying Anveo $0.002 and Flowroute $0.01, that’s an average cost of $0.00216. In my application that’s enough of a saving to justify the administrative hassle of an additional vendor.

There are also a few calls, perhaps 0.05%, where Anveo doesn’t fail over but the result is problematic (poor quality, dropped call, DTMF doesn’t pass, etc.), where Flowroute might have had a better result. However, this is virtually impossible to troubleshoot or even measure, because phones are generally unreliable these days. Even before Covid, we had businesses running VoIP on overloaded networks, users on crappy car Bluetooth in areas with poor cellular coverage, etc. Now, we have millions of people working from home using softphones with laptop mic and speaker, connected over flaky Wi-Fi, while their internet connections are saturated by the kids taking classes by Zoom. Even if your system were perfect, a few percent of calls will be bad. If you improve the result by 0.05%, there is no way to even tell.