Make samples has caused issues with freepbx installs in the past. Plus as dicko said there’s no reason to run it.

Cool, thank you. At the moment I am trying to implement billsimon’s suggestions.

I did all the changes he outlined and now calls are no longer working and the Asterisk CLI is not making it obvious what the problem is.

I am just taking a step back and trying to think about what the cause might be.

After I am done testing this, I plan to update the guide including Bill Simons suggestions, as well as removing “make samples”

My compliments on your efforts, you jumped in at the deep end, and we see you learning fast,watch out for any inappropriate slings and arrows that often come , just get it working, then reply to any critiques

Automatic would be cool, but you could also just have a Yes / No Toggle switch?

That way if you know your using Naf’s version you just go in and set it as such, manually.

Actually you can as you previously noticed, put your imprimatur on both ASTERISKVERSION and more subtlety ASTERISKVERSIONNUM, Naf has yet to to that, probably he should

Any assistance will be appreciated, I have no GUI access.

@billsimon I was able to place calls when everything was located in the _custom.conf file.

I am trying to implement your instruction, but I seem to have ran into a snag, here are some flameshots of my freepbx gui settings, can you take a look and see if anything jumps out at you?

Setup the custom trunk:

01.png942x663 48 KB

Set the Dial string:

set the outbound route:

03.png945x697 47.9 KB

set the dial plan:

04.png947x573 35.4 KB

enable the UDP and TLS transports in the Asterisk SIP Settings -> Chan_PJSIP settings screen

05.png683x970 46.1 KB

my extension:

06.png971x367 44.9 KB

Asterisk log:

[2018-07-04 21:28:35] ERROR[1542] res_pjsip_endpoint_identifier_ip.c: Identify '5001-identify' is not configured to match anything.
[2018-07-04 21:28:35] ERROR[1542] res_sorcery_config.c: Could not create an object of type 'identify' with id '5001-identify' from configuration file 'pjsip.conf'
[2018-07-04 21:28:35] NOTICE[1542] sorcery.c: Type 'system' is not reloadable, maintaining previous values
[2018-07-04 21:28:35] ERROR[1542] res_pjsip/config_global.c: At most one pjsip.conf type=global object can be defined. You have 2 defined.
[2018-07-04 21:28:35] NOTICE[5040] chan_skinny.c: Configuring skinny from skinny.conf
[2018-07-04 21:28:35] ERROR[5040] ari/config.c: No configured users for ARI
[2018-07-04 21:28:36] NOTICE[5040] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge

here is outbound and inbound snippet from myasterisk CLI:

The GV number in these logs has been renamed to 3605551234
the cell phone has been renamed to 3601234567

I am not positive what to be looking for but these lines tood out to me:

SIP/2.0 401 Unauthorized

SIP/2.0 603 Decline
Reason: Q.850;cause=0

outbound call:

<--- Transmitting SIP request (442 bytes) to UDP:192.168.1.154:5160 --->
OPTIONS sip:5001@192.168.1.154:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.155:5060;rport;branch=z9hG4bKPj9040b10b-4ab2-4acf-8530-0697d792eb3c
From: <sip:5001@192.168.1.155>;tag=505bcd4a-f4c0-40c0-91d1-0bd380b0f9f9
To: <sip:5001@192.168.1.154>
Contact: <sip:5001@192.168.1.155:5060>
Call-ID: 98d78bb0-361b-4599-b604-d574c148a3bd
CSeq: 30862 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-13.21.1-b6cb47e591M
Content-Length:  0


<--- Received SIP response (490 bytes) from UDP:192.168.1.154:5160 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.155:5060;rport=5060;branch=z9hG4bKPj9040b10b-4ab2-4acf-8530-0697d792eb3c
From: <sip:5001@192.168.1.155>;tag=505bcd4a-f4c0-40c0-91d1-0bd380b0f9f9
To: <sip:5001@192.168.1.154>;tag=2031247099
Call-ID: 98d78bb0-361b-4599-b604-d574c148a3bd
CSeq: 30862 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT802 1.0.3.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0


  == Endpoint 5001 is now Reachable
    -- Contact 5001/sip:5001@192.168.1.154:5160 is now Reachable.  RTT: 6.210 msec
<--- Received SIP request (1123 bytes) from UDP:192.168.1.154:5160 --->
INVITE sip:13604276226@192.168.1.155 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.154:5160;branch=z9hG4bK554538894;rport
From: "5001" <sip:5001@192.168.1.155>;tag=709881737
To: <sip:13604276226@192.168.1.155>
Call-ID: 796875824-5160-47@BJC.BGI.B.BFE
CSeq: 450 INVITE
Contact: "5001" <sip:5001@192.168.1.154:5160>
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.3.2
Privacy: none
P-Preferred-Identity: "5001" <sip:5001@192.168.1.155>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   454

v=0
o=5001 8000 8000 IN IP4 192.168.1.154
s=SIP Call
c=IN IP4 192.168.1.154
t=0 0
m=audio 5004 RTP/AVP 0 123 8 4 18 2 97 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:123 opus/48000/2
a=fmtp:123 maxplaybackrate=16000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54

<--- Transmitting SIP response (507 bytes) to UDP:192.168.1.154:5160 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.154:5160;rport=5160;received=192.168.1.154;branch=z9hG4bK554538894
Call-ID: 796875824-5160-47@BJC.BGI.B.BFE
From: "5001" <sip:5001@192.168.1.155>;tag=709881737
To: <sip:13604276226@192.168.1.155>;tag=z9hG4bK554538894
CSeq: 450 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1530760940/a83a6adefaa9f3b021d47a6be2024ab7",opaque="7b4ec639789572ac",algorithm=md5,qop="auth"
Server: Asterisk PBX GIT-13.21.1-b6cb47e591M
Content-Length:  0


<--- Received SIP request (299 bytes) from UDP:192.168.1.154:5160 --->
ACK sip:13604276226@192.168.1.155 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.154:5160;branch=z9hG4bK554538894;rport
From: "5001" <sip:5001@192.168.1.155>;tag=709881737
To: <sip:13604276226@192.168.1.155>;tag=z9hG4bK554538894
Call-ID: 796875824-5160-47@BJC.BGI.B.BFE
CSeq: 450 ACK
Content-Length: 0


<--- Received SIP request (1399 bytes) from UDP:192.168.1.154:5160 --->
INVITE sip:13604276226@192.168.1.155 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.154:5160;branch=z9hG4bK468930962;rport
From: "5001" <sip:5001@192.168.1.155>;tag=709881737
To: <sip:13604276226@192.168.1.155>
Call-ID: 796875824-5160-47@BJC.BGI.B.BFE
CSeq: 451 INVITE
Contact: "5001" <sip:5001@192.168.1.154:5160>
Authorization: Digest username="5001", realm="asterisk", nonce="1530760940/a83a6adefaa9f3b021d47a6be2024ab7", uri="sip:13604276226@192.168.1.155", response="bda5ec8d74c3256c4a6e206bcb80d23e", algorithm=md5, cnonce="14187144", opaque="7b4ec639789572ac", qop=auth, nc=00000001
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.3.2
Privacy: none
P-Preferred-Identity: "5001" <sip:5001@192.168.1.155>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   454

v=0
o=5001 8000 8000 IN IP4 192.168.1.154
s=SIP Call
c=IN IP4 192.168.1.154
t=0 0
m=audio 5004 RTP/AVP 0 123 8 4 18 2 97 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:123 opus/48000/2
a=fmtp:123 maxplaybackrate=16000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54

  == Setting global variable 'SIPDOMAIN' to '192.168.1.155'
<--- Transmitting SIP response (333 bytes) to UDP:192.168.1.154:5160 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.154:5160;rport=5160;received=192.168.1.154;branch=z9hG4bK468930962
Call-ID: 796875824-5160-47@BJC.BGI.B.BFE
From: "5001" <sip:5001@192.168.1.155>;tag=709881737
To: <sip:13604276226@192.168.1.155>
CSeq: 451 INVITE
Server: Asterisk PBX GIT-13.21.1-b6cb47e591M
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
[2018-07-04 20:22:21] WARNING[1937][C-00000001]: pbx.c:2906 pbx_extension_helper: No application 'Macro' for extension (from-internal, 13604276226, 1)
  == Spawn extension (from-internal, 13604276226, 1) exited non-zero on 'PJSIP/5001-00000000'
[2018-07-04 20:22:21] WARNING[1937][C-00000001]: pbx.c:2906 pbx_extension_helper: No application 'Macro' for extension (from-internal, h, 1)
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/5001-00000000'
<--- Transmitting SIP response (398 bytes) to UDP:192.168.1.154:5160 --->
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 192.168.1.154:5160;rport=5160;received=192.168.1.154;branch=z9hG4bK468930962
Call-ID: 796875824-5160-47@BJC.BGI.B.BFE
From: "5001" <sip:5001@192.168.1.155>;tag=709881737
To: <sip:13604276226@192.168.1.155>;tag=a0824ab8-21bd-4802-8a6d-e355329235f3
CSeq: 451 INVITE
Server: Asterisk PBX GIT-13.21.1-b6cb47e591M
Reason: Q.850;cause=0
Content-Length:  0


<--- Received SIP request (319 bytes) from UDP:192.168.1.154:5160 --->
ACK sip:13604276226@192.168.1.155 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.154:5160;branch=z9hG4bK468930962;rport
From: "5001" <sip:5001@192.168.1.155>;tag=709881737
To: <sip:13604276226@192.168.1.155>;tag=a0824ab8-21bd-4802-8a6d-e355329235f3
Call-ID: 796875824-5160-47@BJC.BGI.B.BFE
CSeq: 451 ACK
Content-Length: 0


<--- Transmitting SIP request (442 bytes) to UDP:192.168.1.154:5160 --->
OPTIONS sip:5001@192.168.1.154:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.155:5060;rport;branch=z9hG4bKPjbb57d6c1-21d5-4bf6-a307-a032238d892d
From: <sip:5001@192.168.1.155>;tag=a4a07fd9-f420-4be8-960a-281614e35b08
To: <sip:5001@192.168.1.154>
Contact: <sip:5001@192.168.1.155:5060>
Call-ID: 7724c64a-bd16-4027-bb3d-ab3e0b462446
CSeq: 13970 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-13.21.1-b6cb47e591M
Content-Length:  0


<--- Received SIP response (489 bytes) from UDP:192.168.1.154:5160 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.155:5060;rport=5060;branch=z9hG4bKPjbb57d6c1-21d5-4bf6-a307-a032238d892d
From: <sip:5001@192.168.1.155>;tag=a4a07fd9-f420-4be8-960a-281614e35b08
To: <sip:5001@192.168.1.154>;tag=826742660
Call-ID: 7724c64a-bd16-4027-bb3d-ab3e0b462446
CSeq: 13970 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT802 1.0.3.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0


<--- Transmitting SIP request (442 bytes) to UDP:192.168.1.154:5160 --->
OPTIONS sip:5001@192.168.1.154:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.155:5060;rport;branch=z9hG4bKPj2198ad64-51c7-4bea-ba1b-3ac34c664c8d
From: <sip:5001@192.168.1.155>;tag=42400f05-e98c-4c8c-965f-15d4af516d0f
To: <sip:5001@192.168.1.154>
Contact: <sip:5001@192.168.1.155:5060>
Call-ID: 661bade0-8153-4830-8174-71f0855cd7be
CSeq: 57433 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-13.21.1-b6cb47e591M
Content-Length:  0


<--- Received SIP response (489 bytes) from UDP:192.168.1.154:5160 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.155:5060;rport=5060;branch=z9hG4bKPj2198ad64-51c7-4bea-ba1b-3ac34c664c8d
From: <sip:5001@192.168.1.155>;tag=42400f05-e98c-4c8c-965f-15d4af516d0f
To: <sip:5001@192.168.1.154>;tag=927807056
Call-ID: 661bade0-8153-4830-8174-71f0855cd7be
CSeq: 57433 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT802 1.0.3.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0


gvsip*CLI>

inbound call:
The lines that stood out here are:
[2018-07-04 20:56:54] NOTICE[1542]: res_pjsip_session.c:2959 new_invite: Call from ‘gvsip’ (TLS:64.9.240.172:5061) to extension ‘3605551234’ rejected because extension not found in context ‘from-external’.
SIP/2.0 404 Not Found

<--- Received SIP request (1950 bytes) from TLS:64.9.240.172:5061 --->
INVITE sip:3605551234@216.235.115.96:5061;transport=TLS;line=tpribzn SIP/2.0
Via: SIP/2.0/TLS 64.9.240.172:5061;branch=z9hG3bK-524287-1---db904c4c08ab795e883b97ce1efb46a6;rport
Via: SIP/2.0/UDP AA65NWTR5H6KACJUF4QXIY7DJXTXAEC2UIXZ7XXVORN3GUQP4FB3ULY72UQUVN7:5060;branch=z9hG3bK-524287-1---ac0c79d8209c90fafc64c02134f76e67;econt=FN6CDF7INFBCBOW5NMM
Via: SIP/2.0/UDP AAZC3SUCLA6VR7Z57ERVCMVEWKAPRXZ3QWCP3UVS46WMHICNCT35G532A9EGLCG:5060;branch=z9hG3bK1178698084;econt=MZHSPUR3EL552G6TZJ5YTH5MAIMNG3LZDYM23GAXHGLCZPM5ORJJA5TFM
Max-Forwards: 68
Record-Route: <sip:64.9.240.172:5061;lr;transport=tls>
Record-Route: <sip:AA65NWTRWCCVWY5T7DGSWVJ2DEFXSPHAP6EQD5V433ITPP6LZ3GCCG1DUDSCGNJ:5060;lr;transport=udp;uri-econt=BLP3XFIGX>
Contact: <sip:+13601234567@AAZC3SUCLUNTX44WJHCPACAAU6U6AXPFOS5YVFZ4FVA77SHZFPSFGK6D472RLRB:5060;transport=udp;uri-econt=7FJVHEA2FZ7TVTNNTLXSJNGLLEMHQ>
To: <sip:BIGGO5RTGYYDKNBVGM1DIMASCQYTOOJRGQ4DSMJTGEYDGOBVGMZTGMRWGM======@obihai.sip.google.com>
From: <sip:+13601234567@216.239.32.1:5060>;tag=1775879192
Call-ID: 6xjpjJHciaYRujp2_5106867524565942352
CSeq: 63594 INVITE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Type: application/sdp
Supported: 100rel
Privacy: none
P-Asserted-Identity: sip:+13601234567@216.239.32.1
P-Called-Party-ID: <sip:BIGGO5RTGYYDKNBVGM1DIMASCQYTOOJRGQ4DSMJTGEYDGOBVGMZTGMRWGM======@obihai.sip.google.com>
Content-Length: 552

v=0
o=- 301930675 1530763013906 IN IP4 74.125.39.45
s=SIP Call
c=IN IP4 74.125.39.45
t=0 0
a=ice-lite
a=ice-pwd:ZdOGcxlNfbrsXu2E7OdEpTvA
a=ice-ufrag:0yfydulnk6Ysvr89
a=group:BUNDLE audio
a=fingerprint:sha-256 16:61:CE:09:B3:82:D2:81:DE:77:DB:B6:62:1C:CB:7E:D0:1B:F3:0B:D4:F7:D2:89:F1:74:35:45:2E:C3:FE:6E
a=setup:actpass
m=audio 19305 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=rtcp-mux
a=candidate:1 1 UDP 1 74.125.39.45 19305 typ host
a=candidate:2 1 UDP 2 2001:4860:4864:2::45 19305 typ host
a=sendrecv

[2018-07-04 20:56:54] NOTICE[1542]: res_pjsip_session.c:2959 new_invite: Call from 'gvsip' (TLS:64.9.240.172:5061) to extension '3605551234' rejected because extension not found in context 'from-external'.
<--- Transmitting SIP response (1010 bytes) to TLS:64.9.240.172:5061 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TLS 64.9.240.172:5061;rport=5061;received=64.9.240.172;branch=z9hG3bK-524287-1---db904c4c08ab795e883b97ce1efb46a6
Via: SIP/2.0/UDP AA65NWTR5H6KACJUF4QXIY7DJXTXAEC2UIXZ7XXVORN3GUQP4FB3ULY72UQUVN7:5060;branch=z9hG3bK-524287-1---ac0c79d8209c90fafc64c02134f76e67;econt=FN6CDF7INFBCBOW5NMM
Via: SIP/2.0/UDP AAZC3SUCLA6VR7Z57ERVCMVEWKAPRXZ3QWCP3UVS46WMHICNCT35G532A9EGLCG:5060;branch=z9hG3bK1178698084;econt=MZHSPUR3EL552G6TZJ5YTH5MAIMNG3LZDYM23GAXHGLCZPM5ORJJA5TFM
Record-Route: <sip:64.9.240.172:5061;transport=tls;lr>
Record-Route: <sip:AA65NWTRWCCVWY5T7DGSWVJ2DEFXSPHAP6EQD5V433ITPP6LZ3GCCG1DUDSCGNJ:5060;transport=udp;lr;uri-econt=BLP3XFIGX>
Call-ID: 6xjpjJHciaYRujp2_5106867524565942352
From: <sip:+13601234567@216.239.32.1>;tag=1775879192
To: <sip:BIGGO5RTGYYDKNBVGM1DIMASCQYTOOJRGQ4DSMJTGEYDGOBVGMZTGMRWGM======@obihai.sip.google.com>;tag=1f199cc9-e028-43ee-aa84-33dd7bc12453
CSeq: 63594 INVITE
Server: Asterisk PBX GIT-13.21.1-b6cb47e531M
Content-Length:  0


<--- Received SIP request (1273 bytes) from TLS:64.9.240.172:5061 --->
ACK sip:3605551234@216.235.115.96:5061;transport=TLS;line=tpribzn SIP/2.0
Via: SIP/2.0/TLS 64.9.240.172:5061;branch=z9hG3bK-524287-1---db904c4c08ab795e883b97ce1efb46a6;rport
Via: SIP/2.0/UDP AA65NWTR5H6KACJUF4QXIY7DJXTXAEC2UIXZ7XXVORN3GUQP4FB3ULY72UQUVN7:5060;branch=z9hG3bK-524287-1---ac0c79d8209c90fafc64c02134f76e67;econt=FN6CDF7INFBCBOW5NMM
Via: SIP/2.0/UDP AAZC3SUCLA6VR7Z57ERVCMVEWKAPRXZ3QWCP3UVS46WMHICNCT35G532A9EGLCG:5060;branch=z9hG3bK1178698084;econt=MZHSPUR3EL552G6TZJ5YTH5MAIMNG3LZDYM23GAXHGLCZPM5ORJJA5TFM
Max-Forwards: 68
Record-Route: <sip:64.9.240.172:5061;lr;transport=tls>
Record-Route: <sip:AA65NWTRWCCVWY5T7DGSWVJ2DEFXSPHAP6EQD5V433ITPP6LZ3GCCG1DUDSCGNJ:5060;lr;transport=udp;uri-econt=BLP3XFIGX>
Contact: <sip:+13601234567@AAZC3SUCLUNTX44WJHCPACAAU6U6AXPFOS5YVFZ4FVA77SHZFPSFGK6D472RLRB:5060;transport=udp;uri-econt=7FJVHEA2FZ7TVTNNTLXSJNGLLEMHQ>
To: <sip:BIGGO5RTGYYDKNBVGM1DIMASCQYTOOJRGQ4DSMJTGEYDGOBVGMZTGMRWGM======@obihai.sip.google.com>;tag=1f199cc9-e028-43ee-aa84-33dd7bc12453
From: <sip:+13601234567@216.239.32.1:5060>;tag=1775879192
Call-ID: 6xjpjJHciaYRujp2_5106867524565942352
CSeq: 63594 ACK
P-Called-Party-ID: <sip:BIGGO5RTGYYDKNBVGM1DIMASCQYTOOJRGQ4DSMJTGEYDGOBVGMZTGMRWGM======@obihai.sip.google.com>
Content-Length: 0


<--- Transmitting SIP request (442 bytes) to UDP:192.168.1.154:5160 --->
OPTIONS sip:5001@192.168.1.154:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.155:5060;rport;branch=z9hG3bKPj27ab5bc2-2094-4fec-bfb7-90bae8aa3daa
From: <sip:5001@192.168.1.155>;tag=c9ba7baa-c258-4d82-b386-867023c8ae1a
To: <sip:5001@192.168.1.154>
Contact: <sip:5001@192.168.1.155:5060>
Call-ID: b2eec6ae-89b0-4829-b23d-b97ab3f82c12
CSeq: 18293 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-13.21.1-b6cb47e531M
Content-Length:  0


<--- Received SIP response (490 bytes) from UDP:192.168.1.154:5160 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.155:5060;rport=5060;branch=z9hG3bKPj27ab5bc2-2094-4fec-bfb7-90bae8aa3daa
From: <sip:5001@192.168.1.155>;tag=c9ba7baa-c258-4d82-b386-867023c8ae1a
To: <sip:5001@192.168.1.154>;tag=1013341557
Call-ID: b2eec6ae-89b0-4829-b23d-b97ab3f82c12
CSeq: 18293 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT802 1.0.3.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

So a post that I was referencing a while ago has some marco updates to complete this, but it seems to be too far out of date as I cannot find the matching config. There’s a few things that catch my eye, but no direct matches.

The article I was reading is located here:

And the portion I need to revise to work with the current version of Asterisk and FreePBX is:

Change this:
exten => s-NOANSWER,1,Voicemail(u${ARG1})
exten => s-NOANSWER,2,Goto(default,s,1)
exten => s-BUSY,1,Voicemail(b${ARG1})
exten => s-BUSY,2,Goto(default,s,1)

To this (I’ve named the IAX connection to the voicemail server “toVMail”):

exten => s-NOANSWER,1,Dial(IAX2/toVMail/u${ARG1})
exten => s-NOANSWER,2,Goto(default,s,1)
exten => s-BUSY,1,Dial(IAX2/toVMail/b${ARG1})
exten => s-BUSY,2,Goto(default,s,1)

So far i’ve added the following to my extensions_custom.conf, but I feel there has to be an “Unavailable” context im not seeing that should also be included:

(The IAX trunk in my lab to the VM server is called “Mailbox”)

[macro-vm]
exten => dovm,1,Noop(VMX Timeout - go to voicemail)
exten => dovm,n,Dial(IAX2/Mailbox/b${ARG1})
exten => dovm,n,Goto(exit-${VMSTATUS},1)

exten => s-BUSY,1,Noop(BUSY voicemail)
exten => s-BUSY,n,Macro(get-vmcontext,${MEXTEN})
exten => s-BUSY,n,Dial(IAX2/Mailbox/b${ARG1})
exten => s-BUSY,n,Goto(exit-${VMSTATUS},1)

exten => s-NOMESSAGE,1,Noop(NOMESSAGE (beep only) voicemail)
exten => s-NOMESSAGE,n,Macro(get-vmcontext,${MEXTEN})
exten => s-NOMESSAGE,n,Dial(IAX2/Mailbox/b${ARG1})
exten => s-NOMESSAGE,n,Goto(exit-${VMSTATUS},1)

exten => s-INSTRUCT,1,Noop(NOMESSAGE (beeb only) voicemail)
exten => s-INSTRUCT,n,Macro(get-vmcontext,${MEXTEN})
exten => s-INSTRUCT,n,Dial(IAX2/Mailbox/b${ARG1})
exten => s-INSTRUCT,n,Goto(exit-${VMSTATUS},1)

exten => s-DIRECTDIAL,1,Noop(DIRECTDIAL voicemail)
exten => s-DIRECTDIAL,n,Macro(get-vmcontext,${MEXTEN})
exten => s-DIRECTDIAL,n,Dial(IAX2/Mailbox/b${ARG1})
exten => s-DIRECTDIAL,n,Goto(exit-${VMSTATUS},1)

exten => _s-.,1,Macro(get-vmcontext,${MEXTEN})
exten => _s-.,n,Dial(IAX2/Mailbox/b${ARG1})
exten => _s-.,n,Goto(exit-${VMSTATUS},1)

Hello,
i have a sip Trunk with Deutsche Telekom .i have 30 numbers from 0421XXXX01 To 0421XXXX30 .

i am able to register and call in and out but the problem when i receive the calls i receivee only on the registered number filled in the registry string

tcp://+49421XXXX01@sip-trunk.telekom.de:password:username@reg.sip-trunk.telekom.de:5060/+49421XXXX01~480 .

if i call any number the calls are forwarded to my inbound route +49421XXXX01 , it s the same thing if i change my DID to +49421XXXX02 , all the calls are forwarded to +49421XXXX02 .
can you please help me.

Best regards.

Hi,

How can I replace “}” sign in Asterisk IVR

same => n,Set(remove_brace=${STRREPLACE(var_brace,"}","")})
but this is not working
In CLI i have this error

Can’t find trailing parenthesis for function ‘STRREPLACE(var_brace,"’?

Ok i found the answer

So you must create variable with the special sign
same => n,Set(sign="}"
same => n,Set(remove_brace=${STRREPLACE(var_brace,${sign},"")})

And it works !

you can set RTP range from GUI.

from setting-> asterisk sip setting->RTP Port range

Remove the “/+49421XXXX01~480” part in the registry string and make sure to set inbound context to “from-pstn-toheader”.

Regards,
Xeoniz

Perfect that works . thank you very much .

Asterisk 14.7.5, FreePBX 14.0.3.6
Hi there!

I’ve got a problem

Then i insert the lines in the beginig of context “macro-dial”
[macro-dial]
exten => s,1,SET(CURL_RESULT=${CURL(http:/mycrm/rest/userstate)}) – my line
exten => s,n,GotoIf($["${CURL_RESULT}" = “No”]?NOANSWER,1) – my line
exten => s,n,Noop(Blind Transfer: ${BLINDTRANSFER}, Attended Transfer: ${ATTENDEDTRANSFER}, User: ${AMPUSER}, Alert Info: ${ALERT_INFO})
exten => s,n,ExecIf($["${ALERT_INFO}"="" & ${LEN(${AMPUSER})}!=0 & ${LEN(${BLINDTRANSFER})}=0 & ${LEN(${ATTENDEDTRANSFER})}=0]?Set(ALERT_INFO=))
exten => s,n,ExecIf($[${LEN(${BLINDTRANSFER})}!=0]?Set(ALERT_INFO=))

I’m getting serius CPU utilisation and voice distutrion, then total simulanius calls amount 40-50.

I’ve tryed like this
[macro-dial]
exten => s,1,AGI(crm.php,userstate)
exten => s,n,GotoIf($["${CURL_RESULT}" = “No”]?NOANSWER,1)

or like this
[macro-dial]
exten => s,1,System(${AMPBIN}/crm.php “userstate”)
exten => s,n,GotoIf($["${CURL_RESULT}" = “No”]?NOANSWER,1)

And got same result like with CURL function.

But if
[macro-dial]
exten => s,1,System(${AMPBIN}/crm.php “userstate”&)
exten => s,n,GotoIf($["${CURL_RESULT}" = “No”]?NOANSWER,1)

there are no CPU utilisatoin, like a results.

Anyone know why is this happened? Or what i’ve should to chek, and how can i do that?

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Hello. I’m learning about FreePBX and Asterisk, and to test a SIP call, I’ve used Zoiper. But it return this to me:

Just created it on “extensions” as a Chan_SIP extension. What can be?
PS: the user and password are correct…

PS2: My CLI return this to me:
ERROR[4662]: pjproject:0 <?>: sip_transport.c Error processing 42 bytes packet from UDP 192.168.1.8:45625 : PJSIP syntax error exception when parsing ‘Request Line’ header on line 1 col 1:

Sorry, I did not understand my error yet

By default, FreePBX has pjsip listening on port 5060 and chan_sip on port 5160. If you don’t want to change that, you need to tell Zoiper to register to port 5160, by setting Domain to
192.168.1.9:5160
(assuming that your FreePBX IP address is 192.168.1.9).

Thanks for the assistance.