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No login box on main admin login page
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Freepbx h.323 trunk to cisco
Hello.
Configured h.323 to cisco. Freepbx configuration without nat.
h .323 is used for short numbers between offices. When you call, you are redirected to h.323 trunk, the ip address of the provider appears in the call, not the internal ip address.
OOH323 Debugging Enabled
— ooh323_request - data Cisco/6001718 format (g722)
— ooh323_alloc
+++ ooh323_alloc
— find_peer “Cisco”
comparing with “192.168.47.254”
found matching peer
+++ find_peer “Cisco”
— ooh323_new - Cisco
+++ h323_new
— onNewCallCreated 7f3088019a58: ooh323c_o_5
— find_call
+++ find_call
Outgoing call Cisco(ooh323c_o_5) - Codec prefs - (ulaw|alaw|g729|g723|gsm)
Adding capabilities to call(outgoing, ooh323c_o_5)
Adding g711 ulaw capability to call(outgoing, ooh323c_o_5)
Adding g711 alaw capability to call(outgoing, ooh323c_o_5)
Adding g729A capability to call(outgoing, ooh323c_o_5)
Adding g729 capability to call(outgoing, ooh323c_o_5)
Adding g729B capability to call(outgoing, ooh323c_o_5)
Adding g7231 capability to call (outgoing, ooh323c_o_5)
Adding gsm capability to call(outgoing, ooh323c_o_5)
— configure_local_rtp
+++ configure_local_rtp
+++ onNewCallCreated ooh323c_o_5
+++ ooh323_request
----- ooh323_queryoption 16 on channel OOH323/Cisco-4
+++++ ooh323_queryoption 16 on channel OOH323/Cisco-4
— ooh323_call- Cisco/6001718
+++ ooh323_call
— onOutgoingCall 7f3088019a58: ooh323c_o_5
— find_call
+++ find_call
setting callid number 170
+++ onOutgoingCall ooh323c_o_5
— onCallCleared ooh323c_o_5
— find_call
+++ find_call
+++ onCallCleared
— ooh323_hangup
+++ ooh323_hangup
[2019-05-24 12:02:18] WARNING[2599][C-00000131]: chan_sip.c:24055 handle_response_invite: Received response: “Forbidden” from ‘“FIO” <sip-170@10.60.x.x:5160>;tag=as094f5551’
— ooh323_destroy
Destroying Cisco
Destroying ooh323c_o_5
+++ ooh323_destroy
Must be '“FIO” <sip-170@192.168.x.x:5160>
How can I fix this?
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FreePbx user loggin module
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How to configure the new Property Manager app
Have you looked at the Wiki?
https://wiki.freepbx.org/display/FPG/Property+Manager+Getting+Started
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How to configure the new Property Manager app
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What is the default dialplan for VEGA-100G
Hello:
we are testing with other E1 connection with FreePBX and VEGA-100G. we can make inbound call through VEGA-100G, and forward to FreePBX with PJSIP. But when we make outbound calls, it looks VEGA-100G can not route to out. the VEGA log shows:
LOG: 29/05/2019 04:49:21.595 (null) ©R01C00 Login attempt username: intern locally authenticated
LOG: 29/05/2019 04:49:21.595 CONSOLE (I)R02C00 AUTOEXEC-started
LOG: 29/05/2019 04:49:21.595 LAN (A)Rb4C00 TFTP services have not been configured
LOG: 29/05/2019 04:49:21.595 CONSOLE (A)RbbC14 AUTOEXEC-TFTP not configured
LOG: 29/05/2019 04:51:14.955 ROUTER (A)Rb0C00 FINDROUTE: rejected; no route
<-- SIP [2,1] dest=NAME:300,TEL:300
LOG: 29/05/2019 04:51:14.955 ROUTER (I)R07C00 no route to destination
call ref=[f1000533]
LOG: 29/05/2019 04:51:14.955 SIP (I)R04C10 disconnect(disc req) 1
call ref=[f1000533]
LOG: 29/05/2019 04:51:48.425 WEBSERV ©R01C00 Login attempt username: admin locally authenticated
I check VEGA dialplan, there is To_SIP only. How do I add a other dialplan to support FreePBX->E1? Could you give me some default examples？
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How can I see the registration string being sent to my trunk provider
I have set up a chan_sip trunk for use with voip.ms. The outbound settings are:
username=127304_SANMIGUEL
type=peer
trustrpid=yes
sendrpid=yes
secret=
qualify=yes
nat=yes
insecure=invite
host=losangeles.voip.ms
fromuser=127304_SANMIGUEL
disallow=all
context=from-trunk
canreinvite=nonat
allow=ulaw
The incoming string is:
127304_SANMIGUEL:mysecret@losangeles.voip.ms:5060
The provider tells me that I am only sending the character ‘G’ so the registration is failing. Their support gave me a pcap file which doesn’t show this string anywhere in it so I’d like to know exactly how to determine what is being sent as a registration string. The support from voip.ms has not been of much help.
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Change Find Me/Follow me in extension from CLI
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Would this unusual analog setup be possible?
Agreed. Possibly
The above is just an example. You can probably find it cheaper elsewhere, especially if you are buying five at once.
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Bandwidth.com + freepbx + grandstream ata + nec dsx 80 pbx = DTMF Problems
This should not be hard to troubleshoot. To start, temporarily disconnect one port from the NEC and connect the HT port to an analog phone. Call in, answer the analog phone, confirm that you have clean audio in both directions, then press keys on the calling phone and confirm that you hear clean DTMF tones on the analog set. If not, debugging tools include packet capture on the PBX (look at RTP being sent to the HT) and syslog on the HT.
If the DTMF sounds good, reconnect to the NEC and monitor a call with a butt set. If you hear nothing wrong and still have trouble, one possibility is high frequency noise messing with the NEC – try a DSL filter between them (try it both ways). If no luck, perhaps changing the FXS Tx gain will help.
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Problems recognizing distinctive ring using Sangoma B600E
Assuming that you have
dring1context=5035551212
Then in /etc/asterisk/extensions_custom.conf put something like
[5035551212]
exten => s,1,Goto(from-pstn,5035551212,1)
Finally, set up an Inbound Route with DID 5035551212 and Destination as desired.
Sorry, I know nothing about the detection reliability issue. Does this line have caller ID? If so, does the number get received correctly?
I’m curious why you don’t just port some or all of these numbers to VoIP. What city are they in? Unless it’s very rural, there is probably a CLEC with a presence in the rate center.
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Vega 3050G distingtive ringtone, alert-info issue
Hi everyone and welcome (my first post)!
Enviroment is:
IP PBX in this system is 3CX Phone System,
IP Phones are Snom devices,
IP Gateways are Sangoma Vega 3050G
I have problem with Vega 3050G Gateways to get different ringers for external calls and all other calls.
I followed Vega manual for alert-info but without any result.
My PBX (3CX) is sending:
-
not sending Alert-Info for internal calls
-
for external calls is sending
Alert-Info: <http://127.0.0.1>;info=external -
for queue calls is sending
Alert-Info: <http://127.0.0.1>;info=queue
There is no problem with distingtive ringtones on IP Phones but Vega looks like it has problem to recognize alert-info message.
Here is part of SIP TRACE:
SIP m:0060593 0650 00607<-- UA RX — From UDP(8):10.1.1.58:5060
INVITE sip:6183@10.1.1.60:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.58:5060;branch=z9hG4bK-524287-1—f17920074a2d965f;rport
Max-Forwards: 70
Contact: <sip:0XXXXXXXXX@10.1.1.58:5060>
To: <sip:XXXX@XXX.XXX.XX>
From: “XXXXXXXXX”<sip:XXXXXXXXX@10.1.1.58:5060;nf=e>;tag=6a14336e
Call-ID: xUmo6sKCaC2-VK0dFun3Ag…
CSeq: 1 INVITE
Alert-Info: <127.0.0.1>;info=external
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
User-Agent: 3CXPhoneSystem 15.5.15502.6 (15502)
Content-Length: 411
and below part of Vega config:
purge ._advanced.pots.ring
cp ._advanced.pots.ring.1
set ._advanced.pots.ring.1.frequency=“50”
set ._advanced.pots.ring.1.name=“internal”
set ._advanced.pots.ring.1.repeat=“1”
set ._advanced.pots.ring.1.ring1_on=“400”
set ._advanced.pots.ring.1.ring1_off=“200”
set ._advanced.pots.ring.1.ring2_on=“400”
set ._advanced.pots.ring.1.ring2_off=“4000”
set ._advanced.pots.ring.1.ring3_on=“0”
set ._advanced.pots.ring.1.ring3_off=“0”
cp ._advanced.pots.ring.2
set ._advanced.pots.ring.2.frequency=“50”
set ._advanced.pots.ring.2.name=“external”
set ._advanced.pots.ring.2.repeat=“1”
set ._advanced.pots.ring.2.ring1_on=“1000”
set ._advanced.pots.ring.2.ring1_off=“4000”
set ._advanced.pots.ring.2.ring2_on=“0”
set ._advanced.pots.ring.2.ring2_off=“0”
set ._advanced.pots.ring.2.ring3_on=“0”
set ._advanced.pots.ring.2.ring3_off=“0”
Any idea why it is not working?
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Yum update python conflicts
Oke, but when I try that, it also doesn’t work. I get the same error:
https://pastebin.com/QxFUKerX
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Adding an Extra Network Card
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
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Ringing tones per trunk/route
Hello Everyone,
Thanks in advance for the help. I rarely post here since the forums are so well developed and the platform is so popular. It’s amazing what a simple google search can solve (it’s possible I just did’nt keyword it right to get there!).
Anywho, I’m in an interesting position. The company I work for is opening a small customer service office in the UK and we have setup a small phone system for the locals using one of our US trunks (Twilio).
I’m assuming the ringback may be coming from the trunk, but I did not think so. This is something I never got into (who generates what, the telco did that for us in the good 'ol days on our POTS lines ;)).
So, my question is where can I define ringback tones? I set everything to UK in the configuration (I could find) and i’m still gettng US ringback tones.
It’s no big deal, but my UK guys who are on the phone ALL the time are very, very proud of their ringback’s ;).
This is not a big deal, management said to screw off in appeasing them, but I’m spending my free time to get a giggle out of a couple of brits.
I have found some recorded ringtones I can play but that seems like a cheat ;-0. I was thinking about taking one and adding something obnoxious to the end of it as one hell of a prank, however, I found those meetings with HR are pretty uncomfortable…
I am very interested in signailing tone’s, and I’m reading up on a bunch of stuff. I really don’t understand exactly what is happening anymore and any input would be greatly appreciated.
In the meantime, I promised myself to stay out of HR, but if anyone has any good idea’s I’m game.
Thanks a bunch in advance & Best Regards,
Chris W.
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Ringing tones per trunk/route
Just a guess here, because I don’t understand your post:
In Advanced Settings, set Country Indication Tones to United Kingdom.
For the incoming trunk, if pjsip, set (in PJSIP Settings -> Advanced) Inband Progress to Yes.
If chan_sip trunk, add progressinband=yes to the PEER Details (USER Details should normally be blank).
Call in to test.
If I missed the boat, please provide more details:
Is PBX dedicated to UK office, or does it serve multiple countries?
If concerned with ringback heard by customers calling in, are they calling a UK geographic number or something else (freephone, number in another country, etc.)?
How are incoming calls routed (IVR, queue, Ring Group, sent to specific extension, etc.)?
If concerned with ringback heard by the UK staff, what devices are they using (IP phone make/model, softphone name/version, SIP app name/platform, etc.)?
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[SOLVED] How to disable directmedia in all pjsip endpoints
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How can I see the registration string being sent to my trunk provider
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Call Recordign Gets corupted in SGN7 without efecting the conversation
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
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Would this unusual analog setup be possible?
Sangoma Vega gateways have nice tight integration with FreePBX.
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