FusionPbx and Grandstream HT-503

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Daniele

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Feb 1, 2017
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Hi,
I'm trying to move on with FusionPbx and make a test scenario where I've a PSTN gateway: Grandstream HT-503.
The gateway so converts PSTN calls to voip and send that to FusionPbx, and FusionPbx can make calls via HT-503 using PSTN access.

I know well how configure that in Asterisk, but seems that in FreeSwitch and FusionPbx the configuration is different.
I read some advices on this old forum: https://freeswitch.org/confluence/display/FREESWITCH/Grandstream#Grandstream-GrandstreamHandyTone503

  1. First of all, I created a gateway called "RTG" with proxy and real the ip of the HT-503 and I set "register" to false.
  2. I set the "General" section of my HT-503 an unconditional forward of all calls from PSTN to voip using rtg@mydomainname on port 5060.
  3. I created the user "rtg" into FusionPbx
  4. In the "FXO" section of HT-503 I set proxy to the ip of my fusionPbx, user and password of the user "rtg"
So far doesn't work neither outgoing calls neither incoming calls.
But for incoming calls seems my HT-503 is sending correctly to FusionPbx that refuse those with these messages:

Code:
0932e059-4ba2-4856-a334-3dac768b48f4 2017-02-14 16:40:47.167729 [NOTICE] sofia.c:2331 Hangup sofia/internal/02123456@mydomain.com [CS_NEW] [CALL_REJECTED]
and you must configure your device to use the proper domain in it's authentication credentials.
You must define a domain called 'mydomain.com' in your directory and add a user with the id="rtg" attribute
2017-02-14 16:40:47.167729 [WARNING] sofia_reg.c:2906 Can't find user [rtg@mydomain.com] from 77.15.21.142
2017-02-14 16:40:47.147727 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f046c07c040 released.
2017-02-14 16:40:47.147727 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f046c07c040 Connected.
2017-02-14 16:40:47.127737 [DEBUG] sofia.c:9982 IP 77.15.21.142 Rejected by acl "domains". Falling back to Digest auth.
0932e059-4ba2-4856-a334-3dac768b48f4 2017-02-14 16:40:47.127737 [DEBUG] sofia.c:9815 sofia/internal/020000001@mydomain.com receiving invite from 77.15.21.142:49898 version: 1.6.14 -23-e460bf8 64bit
2017-02-14 16:40:47.127737 [DEBUG] sofia.c:2441 Re-attaching to session 0932e059-4ba2-4856-a334-3dac768b48f4
0932e059-4ba2-4856-a334-3dac768b48f4 2017-02-14 16:40:47.067741 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/020000001@mydomain.com) State NEW
2017-02-14 16:40:47.067741 [WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile 'internal' for [rtg@mydomain.com] from ip 77.15.21.142
2017-02-14 16:40:47.067741 [DEBUG] sofia.c:2333 detaching session 0932e059-4ba2-4856-a334-3dac768b48f4
2017-02-14 16:40:47.067741 [DEBUG] sofia.c:9982 IP 77.15.21.142 Rejected by acl "domains". Falling back to Digest auth.
0932e059-4ba2-4856-a334-3dac768b48f4 2017-02-14 16:40:47.067741 [DEBUG] sofia.c:9815 sofia/internal/020000001@mydomain.com receiving invite from 77.15.21.142:49898 version: 1.6.14 -23-e460bf8 64bit

I also tried to add ip 77.15.21.142 to my Access control for "mydomain.com" but seems it is ignored.

I hope someone is able to give me some advice.

Thanks
 

Daniele

Member
Feb 1, 2017
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Oh, you are right. The external profile is bound on the 5080. Anyway also changing that the calls is rejected from fusionPbx.

For this reason apparently:

2017-02-14 17:33:27.627742 [WARNING] sofia_reg.c:2906 Can't find user [rtg@mydomain.com]

Should I make a particular configuration of the user in fusionPbx?

Thanks
 

DigitalDaz

Administrator
Staff member
Sep 29, 2016
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Just create a destination for the real number then forward it to the
realnumber@
 
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Daniele

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Feb 1, 2017
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I created destinations in fusionpbx for the real number of the RTG that is 02123456. Despite this, I continue to see the call is rejected as before.

Thanks
 

Daniele

Member
Feb 1, 2017
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Sure,
logs are the same I posted before but maybe I am missing something:

Code:
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/02998877@mydomain.com) State DESTROY going to sleep
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:181 sofia/internal/02998877@mydomain.com Standard DESTROY
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] mod_sofia.c:343 sofia/internal/02998877@mydomain.com SOFIA DESTROY
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/02998877@mydomain.com) State DESTROY
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/02998877@mydomain.com) Running State Change CS_DESTROY (Cur 2 Tot 386)
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [NOTICE] switch_core_session.c:1669 Close Channel sofia/internal/02998877@mydomain.com [CS_DESTROY]
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [NOTICE] switch_core_session.c:1665 Session 386 (sofia/internal/02998877@mydomain.com) Ended
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_session.c:1647 Session 386 (sofia/internal/02998877@mydomain.com) Locked, Waiting on external entities
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/02998877@mydomain.com) State Change CS_REPORTING -> CS_DESTROY
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/02998877@mydomain.com) State REPORTING going to sleep
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:174 sofia/internal/02998877@mydomain.com Standard REPORTING, cause: CALL_REJECTED
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/02998877@mydomain.com) State REPORTING
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/02998877@mydomain.com) Running State Change CS_REPORTING (Cur 3 Tot 386)
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/02998877@mydomain.com) State Change CS_HANGUP -> CS_REPORTING
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/02998877@mydomain.com) State HANGUP going to sleep
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:60 sofia/internal/02998877@mydomain.com Standard HANGUP, cause: CALL_REJECTED
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] mod_sofia.c:438 Channel sofia/internal/02998877@mydomain.com hanging up, cause: CALL_REJECTED
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/02998877@mydomain.com) State HANGUP
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/02998877@mydomain.com) Callstate Change DOWN -> HANGUP
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/02998877@mydomain.com) Running State Change CS_HANGUP (Cur 3 Tot 386)
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] sofia.c:1453 Channel is already hungup.
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [DEBUG] sofia.c:1453 Channel is already hungup.
2017-02-15 16:45:36.267736 [WARNING] sofia_reg.c:1737 SIP auth failure (INVITE) on sofia profile 'internal' for [rtg@mydomain.com] from ip 78.32.25.45
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.267736 [NOTICE] sofia.c:2331 Hangup sofia/internal/02998877@mydomain.com [CS_NEW] [CALL_REJECTED]
and you must configure your device to use the proper domain in it's authentication credentials.
You must define a domain called 'mydomain.com' in your directory and add a user with the id="rtg" attribute
2017-02-15 16:45:36.267736 [WARNING] sofia_reg.c:2906 Can't find user [rtg@mydomain.com] from 78.32.25.45
2017-02-15 16:45:36.267736 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f047c28a450 released.
2017-02-15 16:45:36.247734 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f047c28a450 Connected.
2017-02-15 16:45:36.247734 [DEBUG] sofia.c:9982 IP 78.32.25.45 Rejected by acl "domains". Falling back to Digest auth.
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.247734 [DEBUG] sofia.c:9815 sofia/internal/02998877@mydomain.com receiving invite from 78.32.25.45:63193 version: 1.6.14 -23-e460bf8 64bit
2017-02-15 16:45:36.247734 [DEBUG] sofia.c:2441 Re-attaching to session a7b608c0-0f5c-47a9-b61f-08af42f8eb9d
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.187753 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/02998877@mydomain.com) State NEW
2017-02-15 16:45:36.187753 [WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile 'internal' for [rtg@mydomain.com] from ip 78.32.25.45
2017-02-15 16:45:36.187753 [DEBUG] sofia.c:2333 detaching session a7b608c0-0f5c-47a9-b61f-08af42f8eb9d
2017-02-15 16:45:36.187753 [DEBUG] sofia.c:9982 IP 78.32.25.45 Rejected by acl "domains". Falling back to Digest auth.
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.187753 [DEBUG] sofia.c:9815 sofia/internal/02998877@mydomain.com receiving invite from 78.32.25.45:63193 version: 1.6.14 -23-e460bf8 64bit
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.187753 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/02998877@mydomain.com) Running State Change CS_NEW (Cur 3 Tot 386)
a7b608c0-0f5c-47a9-b61f-08af42f8eb9d 2017-02-15 16:45:36.187753 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/02998877@mydomain.com [a7b608c0-0f5c-47a9-b61f-08af42f8eb9d]
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:34.447740 [DEBUG] switch_rtp.c:6994 Correct audio ip/port confirmed.
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.527741 [DEBUG] switch_rtp.c:6994 Correct audio ip/port confirmed.
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.367744 [DEBUG] mod_sofia.c:631 SOFIA EXCHANGE_MEDIA
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.367744 [DEBUG] switch_core_state_machine.c:653 (sofia/external/02123456) State EXCHANGE_MEDIA
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.367744 [DEBUG] switch_core_state_machine.c:584 (sofia/external/02123456) Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 385)
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.367744 [DEBUG] switch_ivr_bridge.c:1566 (sofia/external/02123456) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:33.367744 [DEBUG] switch_ivr_originate.c:3686 Originate Resulted in Success: [sofia/external/02123456]
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:33.367744 [DEBUG] sofia.c:7041 Channel sofia/internal/911@mydomain.com entering state [early][183]
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:33.367744 [DEBUG] switch_channel.c:3473 (sofia/internal/911@mydomain.com) Callstate Change RINGING -> EARLY
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:33.367744 [NOTICE] mod_sofia.c:2356 Pre-Answer sofia/internal/911@mydomain.com!
bb85c7fc-b9f0-4352-b152-517bb4fc3cad
bb85c7fc-b9f0-4352-b152-517bb4fc3cad a=sendrecv
bb85c7fc-b9f0-4352-b152-517bb4fc3cad a=ptime:20
bb85c7fc-b9f0-4352-b152-517bb4fc3cad a=fmtp:101 0-16
bb85c7fc-b9f0-4352-b152-517bb4fc3cad a=rtpmap:101 telephone-event/8000
bb85c7fc-b9f0-4352-b152-517bb4fc3cad a=rtpmap:3 GSM/8000
bb85c7fc-b9f0-4352-b152-517bb4fc3cad m=audio 30004 RTP/AVP 3 101
bb85c7fc-b9f0-4352-b152-517bb4fc3cad t=0 0
bb85c7fc-b9f0-4352-b152-517bb4fc3cad c=IN IP4 1.2.3.4
bb85c7fc-b9f0-4352-b152-517bb4fc3cad s=FreeSWITCH
bb85c7fc-b9f0-4352-b152-517bb4fc3cad o=FreeSWITCH 1487143529 1487143530 IN IP4 1.2.3.4
bb85c7fc-b9f0-4352-b152-517bb4fc3cad v=0
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:33.367744 [DEBUG] mod_sofia.c:2353 Ring SDP:
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:33.367744 [DEBUG] switch_core_media.c:7039 sofia/internal/911@mydomain.com Set rtp dtmf delay to 40
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:33.367744 [DEBUG] switch_core_media.c:7016 sofia/internal/911@mydomain.com Set 2833 dtmf receive payload to 101
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:33.367744 [DEBUG] switch_core_media.c:7009 sofia/internal/911@mydomain.com Set 2833 dtmf send payload to 101
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:33.347737 [DEBUG] switch_rtp.c:3878 Starting timer [soft] 160 bytes per 20ms
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:6703 AUDIO RTP [sofia/internal/911@mydomain.com] 1.2.3.4 port 30004 -> 69.255.78.70 port 56150 codec: 3 ms: 20
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:33.347737 [INFO] switch_ivr_originate.c:3635 Sending early media
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:33.347737 [DEBUG] switch_ivr_originate.c:410 Setting codec string on sofia/internal/911@mydomain.com to PCMA@8000h@20i
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_channel.c:3473 (sofia/external/02123456) Callstate Change DOWN -> EARLY
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/02123456!
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:7039 sofia/external/02123456 Set rtp dtmf delay to 40
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:7016 sofia/external/02123456 Set 2833 dtmf receive payload to 101
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:7009 sofia/external/02123456 Set 2833 dtmf send payload to 101
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_rtp.c:3878 Starting timer [soft] 160 bytes per 20ms
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:6703 AUDIO RTP [sofia/external/02123456] 1.2.3.4 port 16964 -> 87.238.30.35 port 14206 codec: 8 ms: 20
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4696 sofia/external/02123456 Set 2833 dtmf send payload to 101 recv payload to 101
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4637 Set telephone-event payload to 101@8000
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_codec.c:111 sofia/external/02123456 Original read codec set to PCMA:8
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:3043 Set Codec sofia/external/02123456 PCMA/8000 20 ms 160 samples 64000 bits 1 channels
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4294 Set telephone-event payload to 101@8000
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1]
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1]
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4433 Audio Codec Compare [GSM:3:8000:20:13200:1] ++++ is saved as a match
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [GSM:3:8000:20:13200:1]/[GSM:3:8000:20:13200:1]
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4433 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4433 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
c380771f-e6fb-4470-888e-a7478c7e4f85
c380771f-e6fb-4470-888e-a7478c7e4f85 a=ptime:20
c380771f-e6fb-4470-888e-a7478c7e4f85 a=silenceSupp:off - - - -
c380771f-e6fb-4470-888e-a7478c7e4f85 a=fmtp:101 0-16
c380771f-e6fb-4470-888e-a7478c7e4f85 a=rtpmap:101 telephone-event/8000
c380771f-e6fb-4470-888e-a7478c7e4f85 a=rtpmap:3 GSM/8000
c380771f-e6fb-4470-888e-a7478c7e4f85 a=rtpmap:0 PCMU/8000
c380771f-e6fb-4470-888e-a7478c7e4f85 a=rtpmap:8 PCMA/8000
c380771f-e6fb-4470-888e-a7478c7e4f85 m=audio 14206 RTP/AVP 8 0 3 101
c380771f-e6fb-4470-888e-a7478c7e4f85 t=0 0
c380771f-e6fb-4470-888e-a7478c7e4f85 c=IN IP4 87.238.30.35
c380771f-e6fb-4470-888e-a7478c7e4f85 s=Asterisk PBX 1.6.2.9
c380771f-e6fb-4470-888e-a7478c7e4f85 o=root 1760269886 1760269886 IN IP4 87.238.30.35
c380771f-e6fb-4470-888e-a7478c7e4f85 v=0
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] sofia.c:7051 Remote SDP:
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:33.347737 [DEBUG] sofia.c:7041 Channel sofia/external/02123456 entering state [proceeding][183]
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.267728 [DEBUG] sofia.c:7041 Channel sofia/external/02123456 entering state [calling][0]
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] switch_core_state_machine.c:662 (sofia/external/02123456) State CONSUME_MEDIA going to sleep
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] switch_core_state_machine.c:662 (sofia/external/02123456) State CONSUME_MEDIA
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] switch_core_state_machine.c:584 (sofia/external/02123456) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 385)
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] switch_core_state_machine.c:643 (sofia/external/02123456) State ROUTING going to sleep
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] switch_ivr_originate.c:67 (sofia/external/02123456) State Change CS_ROUTING -> CS_CONSUME_MEDIA
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] mod_sofia.c:143 sofia/external/02123456 SOFIA ROUTING
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] switch_core_state_machine.c:643 (sofia/external/02123456) State ROUTING
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] sofia.c:7041 Channel sofia/external/02123456 entering state [calling][0]
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] switch_core_state_machine.c:584 (sofia/external/02123456) Running State Change CS_ROUTING (Cur 2 Tot 385)
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] switch_core_state_machine.c:627 (sofia/external/02123456) State INIT going to sleep
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] switch_core_state_machine.c:48 (sofia/external/02123456) State Change CS_INIT -> CS_ROUTING
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] switch_core_state_machine.c:40 sofia/external/02123456 Standard INIT
c380771f-e6fb-4470-888e-a7478c7e4f85
c380771f-e6fb-4470-888e-a7478c7e4f85 a=sendrecv
c380771f-e6fb-4470-888e-a7478c7e4f85 a=ptime:20
c380771f-e6fb-4470-888e-a7478c7e4f85 a=rtpmap:13 CN/8000
c380771f-e6fb-4470-888e-a7478c7e4f85 a=fmtp:101 0-16
c380771f-e6fb-4470-888e-a7478c7e4f85 a=rtpmap:101 telephone-event/8000
c380771f-e6fb-4470-888e-a7478c7e4f85 a=rtpmap:8 PCMA/8000
c380771f-e6fb-4470-888e-a7478c7e4f85 a=rtpmap:0 PCMU/8000
c380771f-e6fb-4470-888e-a7478c7e4f85 a=rtpmap:3 GSM/8000
c380771f-e6fb-4470-888e-a7478c7e4f85 m=audio 16964 RTP/AVP 3 0 8 101 13
c380771f-e6fb-4470-888e-a7478c7e4f85 t=0 0
c380771f-e6fb-4470-888e-a7478c7e4f85 c=IN IP4 1.2.3.4
c380771f-e6fb-4470-888e-a7478c7e4f85 s=FreeSWITCH
c380771f-e6fb-4470-888e-a7478c7e4f85 o=FreeSWITCH 1487156567 1487156568 IN IP4 1.2.3.4
c380771f-e6fb-4470-888e-a7478c7e4f85 v=0
c380771f-e6fb-4470-888e-a7478c7e4f85 Local SDP:
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] sofia_glue.c:1283 sofia/external/02123456 sending invite version: 1.6.14 -23-e460bf8 64bit
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] mod_sofia.c:90 sofia/external/02123456 SOFIA INIT
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] switch_core_state_machine.c:627 (sofia/external/02123456) State INIT
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] switch_core_state_machine.c:584 (sofia/external/02123456) Running State Change CS_INIT (Cur 2 Tot 385)
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [DEBUG] mod_sofia.c:4808 (sofia/external/02123456) State Change CS_NEW -> CS_INIT
c380771f-e6fb-4470-888e-a7478c7e4f85 2017-02-15 16:45:31.207729 [NOTICE] switch_channel.c:1104 New Channel sofia/external/02123456 [c380771f-e6fb-4470-888e-a7478c7e4f85]
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:31.207729 [DEBUG] switch_ivr_originate.c:2138 Parsing global variables
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:31.207729 [DEBUG] switch_channel.c:1250 sofia/internal/911@mydomain.com EXPORTING[export_vars] [origination_callee_id_name]=[02123456] to event
bb85c7fc-b9f0-4352-b152-517bb4fc3cad 2017-02-15 16:45:31.207729 [DEBUG] switch_channel.c:1250 sofia/internal/911@mydomain.com EXPORTING[export_vars] [domain_name]=[mydomain.com] to event

02123456: is the number bound with the PSTN. So when a customer call that number the Grandstream HT-503 reply and forward the call to Fusionpbx.
02998877: is the call I made
911: is the extension from which I made the outbound call

Thanks
 

DigitalDaz

Administrator
Staff member
Sep 29, 2016
3,044
565
113
In a post above, I told you to send the calls to port 5080, you still appear to be sending to 5060.
 

Daniele

Member
Feb 1, 2017
45
0
6
43
I posted all the log I saw. Maybe I missed something? According to your suggestion I changed the configuration of my HT-503 and requests are sent to port 5080.

The sure thing is that I see arriving the call in Freeswitch but it is rejected. I guess I need something that in Asterisk is the parameter "insecure=invite".

Thanks
 

Daniele

Member
Feb 1, 2017
45
0
6
43
I made some progress. According to some post I found, I created an extension in fusionpbx for my ht-503 (not a gateway). So now I'm able to receive calls. I'm not still able to make calls because I'm trying to understand if I should make calls using this particular extension.

Seems a bit weird to me...
 

Daniele

Member
Feb 1, 2017
45
0
6
43
Hi, is there some good soul that can give me some advice? Now I tried to use a Patton 4114 FXO but the problem I'm experiencing is exactly the same.
I found this useful post http://lists.freeswitch.org/pipermail/freeswitch-users/2013-January/091703.html, I tried to do the same things but I've always the error

Code:
You must define a domain called 'mydomain.com' in your directory and add a user with the id="rtg" attribute

I added the user in the section users in FusionPbx but I'm wondering if it's correct. I'm trying do so a "simple" things: use my Patton FXO to receive and make calls.

Thanks!
 

Daniele

Member
Feb 1, 2017
45
0
6
43
Some more information hopefully are usefull. I was able to make work the gateway (patton) with inbound calls. Just a step backward: my pbx with FusionPbx is in the cloud so it has a public IP address (no NAT), my gateway (Patton 4114) instead is behind NAT in my office.
I found interesting this page even if old: https://wiki.freeswitch.org/wiki/Traditional_Gateway_connecting

FusionPbx
I created an extension 9999 following the example of the guide.
Then I created a dialplan rule for incoming calls (for now I'm catching everything).

SmarNode Patton
I set the auth to the public ip of my FusionPbx on port 5080 with user and password of the extension 9999. My Patton is registered correctly, in fact:

Code:
SIP Registration Manager: PBX
  -----------------------------

    State:                              Registered
    Registration Inbound:               enabled

    SIP Registration:                   
      State:                            Idle/Waiting
      Registrar:                      publicIpFusionPbx:5080
      Used Registrar:                   

publicIpFusionPbx:5080
      Logical Address:                 
      Physical Address:                 
      Configured Expiration Time:       3600 s
      Actual Expiration Time:           3600 s

    SIP Registration:                   
      State:                            Registered
      Registrar:                     publicIpFusionPbx:5080
      Used Registrar:         

publicIpFusionPbx:5080
      Logical Address:                 
      Physical Address:                 
      Configured Expiration Time:       3600 s
      Actual Expiration Time:           3600 s

Outbound calls don't work (I hear the ring but the call doesn't arrive on the Patton). I natted the port 5060 in my router and sent the traffic to the local ip of the SmartNode Patton.
The dialplan is:

<action application="bridge" data="sofia/internal/$1@publicIpOfMyOffice"/>

I didn't post logs because are very very long but I'm sure someone can tell me if I'm doing something really wrong or I'm on the right path.

Thanks
 

ppenkov

New Member
Aug 10, 2018
9
0
1
39
Some more information hopefully are usefull. I was able to make work the gateway (patton) with inbound calls. Just a step backward: my pbx with FusionPbx is in the cloud so it has a public IP address (no NAT), my gateway (Patton 4114) instead is behind NAT in my office.
I found interesting this page even if old: https://wiki.freeswitch.org/wiki/Traditional_Gateway_connecting

FusionPbx
I created an extension 9999 following the example of the guide.
Then I created a dialplan rule for incoming calls (for now I'm catching everything).

SmarNode Patton
I set the auth to the public ip of my FusionPbx on port 5080 with user and password of the extension 9999. My Patton is registered correctly, in fact:

Code:
SIP Registration Manager: PBX
  -----------------------------

    State:                              Registered
    Registration Inbound:               enabled

    SIP Registration:                  
      State:                            Idle/Waiting
      Registrar:                      publicIpFusionPbx:5080
      Used Registrar:                  

publicIpFusionPbx:5080
      Logical Address:                
      Physical Address:                
      Configured Expiration Time:       3600 s
      Actual Expiration Time:           3600 s

    SIP Registration:                  
      State:                            Registered
      Registrar:                     publicIpFusionPbx:5080
      Used Registrar:        

publicIpFusionPbx:5080
      Logical Address:                
      Physical Address:                
      Configured Expiration Time:       3600 s
      Actual Expiration Time:           3600 s

Outbound calls don't work (I hear the ring but the call doesn't arrive on the Patton). I natted the port 5060 in my router and sent the traffic to the local ip of the SmartNode Patton.
The dialplan is:



I didn't post logs because are very very long but I'm sure someone can tell me if I'm doing something really wrong or I'm on the right path.

Thanks

Hello Daniele,
did you solve this ?

I'm in the exact same situation...want to use a Patton FXO (SN4112 or SN4114) to connect the PSTN access to the Fusion.

Can you tell me how you configured both Patton and Fusion for the registration ?
 

DigitalDaz

Administrator
Staff member
Sep 29, 2016
3,044
565
113
All these should be relatively simple to solve, as long as the following is possible, create a gateway out on the fusion that points to the device, that should deal with outbound, obviously on the device itself you will need to configure it to allow these calls.

For inbound, if the device is able, just send all calls to fusionpbxip:5080
 

teditn

New Member
Oct 6, 2018
17
1
1
43
All these should be relatively simple to solve, as long as the following is possible, create a gateway out on the fusion that points to the device, that should deal with outbound, obviously on the device itself you will need to configure it to allow these calls.

For inbound, if the device is able, just send all calls to fusionpbxip:5080
thank you
it works for incoming calls but I do not know how to do for outgoing calls I have a gateway linksys spa 3000 you can see the pictures I do not know what to choose from the Gateway menu
the spa3000 gateway linked with a number 999 fusionpbx account and call them enter send to number 100 it works fine but i can not direct the outgoing calls
fu1.PNGspa3000.PNG
 
Last edited:

vinicio

New Member
Sep 19, 2020
6
0
1
43
Good evening, I hope this is the right place.
I have recently switched to fusion and the problem I have now with a grandstream ht813 FXO FXS is to use the FXO I don't know what the essential configuration data are.
Pbx side it seems simple following your guides but I have the feeling of making a mistake about the grandstream. Thanks
 
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