FXO adapter


#21

Go to the asterisk command line and type sip set debug peer XXXX where XXXX is the name of the trunk.


#22

That looks very weird. Assuming that you dialed 18004377950 and your trunk name is still 99999 I would expect to see
Called SIP/99999/18004377950

Where did the + come from? Dialed manually from a device that allows such input? Added by the Outbound Route? Added by the trunk manipulation rules?

And where did the @Home come from? Is there anything called Home on your system?


(Stephan Koenig) #23

Thanks for all your input!

The xxxxxxxxxx replaced my landline number and my cell phone number. They are correct in my log, just did not want to post it here.

I took the + out. My sip providers work fine with the + added. We do a lot of international calls. Don’t know if the Grandstream would work with it, so now I took it out.

Home is the name of the trunk in the trunk settings.

Here are some parts of the log:

[2019-08-11 16:34:51] VERBOSE[2613] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.115:5062:
OPTIONS sip:5555555555@192.168.1.115:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK3ad5b10d;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.151>;tag=as4ed22f66
To: <sip:5555555555@192.168.1.115:5062>
Contact: <sip:Unknown@192.168.1.151:5060>
Call-ID: 0fe3dfd36757e3f345d98cee7e83e1c8@192.168.1.151:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.197(13.18.3)
Date: Sun, 11 Aug 2019 20:34:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


[2019-08-11 16:34:51] VERBOSE[2613] chan_sip.c:
<— SIP read from UDP:192.168.1.115:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK3ad5b10d;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.151>;tag=as4ed22f66
To: <sip:5555555555@192.168.1.115:5062>;tag=732842352
Call-ID: 0fe3dfd36757e3f345d98cee7e83e1c8@192.168.1.151:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT813 1.0.0.8
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
[2019-08-11 16:34:51] VERBOSE[2613] chan_sip.c: — (10 headers 0 lines) —
[2019-08-11 16:34:51] VERBOSE[2613] chan_sip.c: Really destroying SIP dialog ‘0fe3dfd36757e3f345d98cee7e83e1c8@192.168.1.151:5060’ Method: OPTIONS

[2019-08-11 16:35:51] VERBOSE[2613] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.115:5062:
OPTIONS sip:5555555555@192.168.1.115:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK4796ee40;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.151>;tag=as24fc58a6
To: <sip:5555555555@192.168.1.115:5062>
Contact: <sip:Unknown@192.168.1.151:5060>
Call-ID: 2ca54172457b434247e032a86e5e31e9@192.168.1.151:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.197(13.18.3)
Date: Sun, 11 Aug 2019 20:35:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


[2019-08-11 16:35:51] VERBOSE[2613] chan_sip.c:
<— SIP read from UDP:192.168.1.115:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK4796ee40;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.151>;tag=as24fc58a6
To: <sip:5555555555@192.168.1.115:5062>;tag=2086197300
Call-ID: 2ca54172457b434247e032a86e5e31e9@192.168.1.151:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT813 1.0.0.8
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
[2019-08-11 16:35:51] VERBOSE[2613] chan_sip.c: — (10 headers 0 lines) —
[2019-08-11 16:35:51] VERBOSE[2613] chan_sip.c: Really destroying SIP dialog ‘2ca54172457b434247e032a86e5e31e9@192.168.1.151:5060’ Method: OPTIONS

[2019-08-11 16:36:11] VERBOSE[19293][C-00073688] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.115:5062:
INVITE sip:14444444444%40Home@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK22126165;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as7fbf4ea0
To: <sip:14444444444%40Home@192.168.1.115>
Contact: <sip:anonymous@192.168.1.151:5060>
Call-ID: 0225eaf8633a78b54fb358de56eca1d6@192.168.1.151:5060
CSeq: 102 INVITE
User-Agent: FPBX-13.0.197(13.18.3)
Date: Sun, 11 Aug 2019 20:36:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 1743565987 1743565987 IN IP4 192.168.1.151
s=Asterisk PBX 13.18.3
c=IN IP4 192.168.1.151
t=0 0
m=audio 10146 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


[2019-08-11 16:36:11] VERBOSE[19293][C-00073688] app_dial.c: Called SIP/5555555555/14444444444@Home
[2019-08-11 16:36:11] VERBOSE[2613] chan_sip.c:
<— SIP read from UDP:192.168.1.115:5062 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK22126165;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as7fbf4ea0
To: <sip:14444444444%40Home@192.168.1.115>
Call-ID: 0225eaf8633a78b54fb358de56eca1d6@192.168.1.151:5060
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT813 1.0.0.8
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
[2019-08-11 16:36:11] VERBOSE[2613] chan_sip.c: — (10 headers 0 lines) —
[2019-08-11 16:36:11] VERBOSE[2613] chan_sip.c:
<— SIP read from UDP:192.168.1.115:5062 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK22126165;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as7fbf4ea0
To: <sip:14444444444%40Home@192.168.1.115>;tag=565558085
Call-ID: 0225eaf8633a78b54fb358de56eca1d6@192.168.1.151:5060
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT813 1.0.0.8
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
[2019-08-11 16:36:11] VERBOSE[2613] chan_sip.c: — (10 headers 0 lines) —
[2019-08-11 16:36:11] VERBOSE[2613][C-00073688] chan_sip.c: Transmitting (NAT) to 192.168.1.115:5062:
ACK sip:14444444444%40Home@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK22126165;rport
Max-Forwards: 70
From: “Anonymous” <sip:anonymous@anonymous.invalid>;tag=as7fbf4ea0
To: <sip:14444444444%40Home@192.168.1.115>;tag=565558085
Contact: <sip:anonymous@192.168.1.151:5060>
Call-ID: 0225eaf8633a78b54fb358de56eca1d6@192.168.1.151:5060
CSeq: 102 ACK
User-Agent: FPBX-13.0.197(13.18.3)
Content-Length: 0
[2019-08-11 16:36:11] VERBOSE[19293][C-00073688] chan_sip.c: Scheduling destruction of SIP dialog ‘0225eaf8633a78b54fb358de56eca1d6@192.168.1.151:5060’ in 6400 ms (Method: INVITE)
[2019-08-11 16:36:11] VERBOSE[19293][C-00073688] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2019-08-11 16:36:11] VERBOSE[19293][C-00073688] pbx.c: Executing [s@macro-dialout-trunk:25] NoOp(“SIP/1102-0000135d”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21”) in new stack


#24

That is not correct, though I have no idea what may be causing it.

You should be seeing INVITE sip:14444444444@192.168.1.115 SIP/2.0

Default FreePBX configuration files do not contain the word ‘Home’ so that is probably something you defined. And, you likely have some unusual setting that is causing @Home (escaped to %40Home) to be appended to the number. Do you know what that may be?

Do you have other trunks e.g. to VoIP providers that are working properly? If so, can you confirm that @Home is not being added to numbers sent on those trunks?

What kind of client(s) are you using (IP phones, analog phones with ATA, softphones, mobile SIP apps, etc.)?


#25

Can you post the screenshots of the trunk and the ht503 configuration?


(Stephan Koenig) #26

I have other VoIP providers that work totally fine. No “home” added there.

We have all kinds of IP phones, but also some ATA adapters.


(Stephan Koenig) #27


#28

You should set outgoing call without authentication to no


(Stephan Koenig) #29

Done, but no change.


(Stephan Koenig) #30

I tried from a different extension:

[2019-08-11 23:10:08] VERBOSE[14994][C-0007368e] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.115:5062:
INVITE sip:15555555555%40Home@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK6305d382;rport
Max-Forwards: 70
From: <sip:14444444444@192.168.1.151>;tag=as73cbbd82
To: <sip:15555555555%40Home@192.168.1.115>
Contact: <sip:14444444444@192.168.1.151:5060>
Call-ID: 3bd46d7078afbb6d4139720d0d1ed6ef@192.168.1.151:5060
CSeq: 102 INVITE
User-Agent: FPBX-13.0.197(13.18.3)
Date: Mon, 12 Aug 2019 03:10:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 613272460 613272460 IN IP4 192.168.1.151
s=Asterisk PBX 13.18.3
c=IN IP4 192.168.1.151
t=0 0
m=audio 13748 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


[2019-08-11 23:10:08] VERBOSE[14994][C-0007368e] app_dial.c: Called SIP/4444444444/15555555555@Home
[2019-08-11 23:10:08] VERBOSE[2613] chan_sip.c:
<— SIP read from UDP:192.168.1.115:5062 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK6305d382;rport=5060
From: <sip:14444444444@192.168.1.151>;tag=as73cbbd82
To: <sip:15555555555%40Home@192.168.1.115>
Call-ID: 3bd46d7078afbb6d4139720d0d1ed6ef@192.168.1.151:5060
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT813 1.0.0.8
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
[2019-08-11 23:10:08] VERBOSE[2613] chan_sip.c: — (10 headers 0 lines) —
[2019-08-11 23:10:08] VERBOSE[2613] chan_sip.c:
<— SIP read from UDP:192.168.1.115:5062 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK6305d382;rport=5060
From: <sip:14444444444@192.168.1.151>;tag=as73cbbd82
To: <sip:15555555555%40Home@192.168.1.115>;tag=485746598
Call-ID: 3bd46d7078afbb6d4139720d0d1ed6ef@192.168.1.151:5060
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT813 1.0.0.8
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
[2019-08-11 23:10:08] VERBOSE[2613] chan_sip.c: — (10 headers 0 lines) —
[2019-08-11 23:10:08] VERBOSE[2613][C-0007368e] chan_sip.c: Transmitting (NAT) to 192.168.1.115:5062:
ACK sip:15555555555%40Home@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK6305d382;rport
Max-Forwards: 70
From: <sip:14444444444@192.168.1.151>;tag=as73cbbd82
To: <sip:15555555555%40Home@192.168.1.115>;tag=485746598
Contact: <sip:14444444444@192.168.1.151:5060>
Call-ID: 3bd46d7078afbb6d4139720d0d1ed6ef@192.168.1.151:5060
CSeq: 102 ACK
User-Agent: FPBX-13.0.197(13.18.3)
Content-Length: 0


[2019-08-11 23:10:08] WARNING[2613][C-0007368e] chan_sip.c: Received response: “Forbidden” from ‘<sip:14444444444@192.168.1.151>;tag=as73cbbd82’
[2019-08-11 23:10:08] VERBOSE[14994][C-0007368e] chan_sip.c: Scheduling destruction of SIP dialog ‘3bd46d7078afbb6d4139720d0d1ed6ef@192.168.1.151:5060’ in 6400 ms (Method: INVITE)
[2019-08-11 23:10:08] VERBOSE[14994][C-0007368e] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)


#31

I’m puzzled. Perhaps a FreePBX bug left something around from a previous attempt, even though it’s not now visible in the GUI.

From a shell prompt, do
grep -i home /etc/asterisk/*conf

If anything appears, examine the file found and post the context containing ‘home’.

Also post FreePBX version, Asterisk version. Are your modules up to date?

Describe what number manipulation, if any, is done by the relevant Outbound Route and the trunk.

Make another call attempt, paste the relevant section of the Asterisk log at pastebin.freepbx.org and post the link here.


(Lorne Gaetz) #32

Restart Asterisk, it will sometimes fix malformed.dial strings:

fwconsole restart

(Stephan Koenig) #33

Thanks! That was a step forward! Not done yet, but the “home” disappeared!

https://pastebin.freepbx.org/view/bb6d1a48


#34

Please paste also the basic and advance settings screenshot from the ht503


(Stephan Koenig) #35


(Stephan Koenig) #36


#37

OK, the HT is still sending 403 Forbidden, but without any info as to what it doesn’t like.

Confirm that the validation check options at the bottom of page 55 of


are set to No (the default).

Confirm that Dial Plan is set to the default of {x+} or (if you had to change it) permits the digit string you are dialing.

Confirm that Stage Method is set to 1.

If the above doesn’t help, set up Syslog on the HT (at Extra Debug level), attempt an outgoing call, and with luck the log will tell why the call was rejected.


(Stephan Koenig) #38

Stage Method was set to 2.

Changed it to 1 and all works perfectly!

YOU ARE ALL AMAZING!! THANK YOU SO MUCH!!


(system) closed #39

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