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Help with troubleshooting phone reboot on call when using pjsip extension

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Sorry for that Dave.
Phone is Polycom IP550 Assembly: 2345-12500-001 Rev:R (updated to the latest version in my EPM)
Bootblock: 3.0.2.0024
Updater: 5.1.1.0132

PBX version: 14.0.11
System version: 12.7.6-1904.sng7
Asterisk version: 16.3.0

I agree, i have googled for hours and i couldn’t find anyone with similar issues in non of the results.


Zulu no audio when dialing through DAHDI

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That is very problematic. All external calls require a PIN. How can I download the alpha version? Or si there a workaround?

Hooking for fun and income

Zulu no audio when dialing through DAHDI

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You can switch to the Alpha track in the Preferences menu in Zulu client.

Enabled email - now get email every day for updates but no VM attachments

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Did you enter by any chance the email address under “pager”?

Polycom Alert info

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Hello,

I am currently having an issue with new versions of Freepbx and getting distinctive ring to function with polycom vvx 300 phones

Using the commercial End point manager I have modified the base file config as follows.

 <se.rt.custom18
   se.rt.custom18.name="Low Double Trill"
   se.rt.custom18.ringer="ringer3"
   >
  </se.rt.custom18>

   voIpProt.SIP.alertInfo.3.value="ebinternal"
   voIpProt.SIP.alertInfo.3.class="custom18"

Per Polycom they are looking for the following info inside alert info.

“Alert-Info: info=ebinternal”

Which is found here

I modified the inbound route and added info=ebinternal

And I can see the info is being added to the sip header but the phone even after reboot and resetup does not ever change the ring tone. Can anyone offer some advice on how to get the polycom custom ring to work. I do have it working in an older version without enpoint manager but nothing i seem to do with the licensed version will seem to work.

– Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/2001-0000085c”, “1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
– Executing [s@func-apply-sipheaders:2] NoOp(“SIP/2001-0000085c”, “Applying SIP Headers to channel”) in new stack
– Executing [s@func-apply-sipheaders:3] Set(“SIP/2001-0000085c”, “SIPHEADERKEYS=Alert-Info”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“SIP/2001-0000085c”, “1”) in new stack
– Executing [s@func-apply-sipheaders:5] Set(“SIP/2001-0000085c”, “sipheader=info=ebinternal”) in new stack
– Executing [s@func-apply-sipheaders:6] SIPAddHeader(“SIP/2001-0000085c”, “Alert-Info: info=ebinternal”) in new stack

Thank you.

Fclose 110 is not a valid stream resource

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All calls go to voicemail & firewall crashing

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If we call internally from extension to extension it goes directly to voicemail on all extensions or if an outside call comes into an extension, it also goes directly to voicemail.
So I went to the Firewall module & when I click Re-Run I get this error:

And then when I go back to the Firewall page, it’s as if the Firewall has been reset & it walks me through the wizard to reset everything.

Any help would be greatly appreciated.


Zulu no audio when dialing through DAHDI

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I’m guessing that only works on the desktop client?

I disabled PIN on a single extension and I can now make external calls but this is simply not possible to do for all extensions. Zulu was a big selling point for getting PBXact and this will prevent deployment until the Alpha version goes stable.

Preferred upgrade path from 12.0.25 - moving from hardware to hyper-v VM preferred

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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

Teléfonos IP que Soporten Codec OPUS

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Cordial Saludo.
Actualmente existe algún teléfono IP gama media que soporte el codec OPUS?

Atentamente,

Yeison Manrique
Ing. Sistemas
Cel. +57 300 217 20 29
www.voipsystem.net.co

Polycom Alert info

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What version firmware are you using?

Try removing info= from your AlertInfo field on inbound route.

This is how i have mine configured and it works:

  <se.rt.internal
   se.rt.internal.name="Internal"
   se.rt.internal.ringer="ringer2"
   se.rt.internal.type="ring"
   >

voIpProt.SIP.alertInfo.2.class="internal"
voIpProt.SIP.alertInfo.2.value="incoming"

I then select custom on the alertInfo field on the inbound route, and put incoming

Multiple Fail2Ban notifications.... where should I be looking?

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This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.

Bucket API initialization failed. ASTERISK EXITING!

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I restarted my FreePBX server this morning. The GUI would load, but it would say Can Not Connect to Asterisk in the corner. I can’t get it back up. This is the output of asterisk -vvvc:

[root@freepbx Default_backup]# asterisk -vvvc
Asterisk 13.22.0, Copyright © 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster…>
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

[ Initializing Custom Configuration Options ]
Couldn’t find manager DBGet in XML documentation
Couldn’t find manager DBGet in XML documentation
== Manager registered action DBGet
Couldn’t find manager DBPut in XML documentation
Couldn’t find manager DBPut in XML documentation
== Manager registered action DBPut
Couldn’t find manager DBDel in XML documentation
Couldn’t find manager DBDel in XML documentation
== Manager registered action DBDel
Couldn’t find manager DBDelTree in XML documentation
Couldn’t find manager DBDelTree in XML documentation
== Manager registered action DBDelTree
PBX UUID: eea80e2f-12e0-4395-abb7-4081b8646cd7
== Registered ‘audio’ codec ‘g723’ at sample rate ‘8000’ with id ‘1’
== Created cached format with name ‘g723’
== Registered ‘audio’ codec ‘ulaw’ at sample rate ‘8000’ with id ‘2’
== Created cached format with name ‘ulaw’
== Registered ‘audio’ codec ‘alaw’ at sample rate ‘8000’ with id ‘3’
== Created cached format with name ‘alaw’
== Registered ‘audio’ codec ‘gsm’ at sample rate ‘8000’ with id ‘4’
== Created cached format with name ‘gsm’
== Registered ‘audio’ codec ‘g726’ at sample rate ‘8000’ with id ‘5’
== Created cached format with name ‘g726’
== Registered ‘audio’ codec ‘g726aal2’ at sample rate ‘8000’ with id ‘6’
== Created cached format with name ‘g726aal2’
== Registered ‘audio’ codec ‘adpcm’ at sample rate ‘8000’ with id ‘7’
== Created cached format with name ‘adpcm’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘8000’ with id ‘8’
== Created cached format with name ‘slin’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘12000’ with id ‘9’
== Created cached format with name ‘slin12’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘16000’ with id ‘10’
== Created cached format with name ‘slin16’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘24000’ with id ‘11’
== Created cached format with name ‘slin24’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘32000’ with id ‘12’
== Created cached format with name ‘slin32’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘44100’ with id ‘13’
== Created cached format with name ‘slin44’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘48000’ with id ‘14’
== Created cached format with name ‘slin48’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘96000’ with id ‘15’
== Created cached format with name ‘slin96’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘192000’ with id ‘16’
== Created cached format with name ‘slin192’
== Registered ‘audio’ codec ‘lpc10’ at sample rate ‘8000’ with id ‘17’
== Created cached format with name ‘lpc10’
== Registered ‘audio’ codec ‘g729’ at sample rate ‘8000’ with id ‘18’
== Created cached format with name ‘g729’
== Registered ‘audio’ codec ‘speex’ at sample rate ‘8000’ with id ‘19’
== Created cached format with name ‘speex’
== Registered ‘audio’ codec ‘speex’ at sample rate ‘16000’ with id ‘20’
== Created cached format with name ‘speex16’
== Registered ‘audio’ codec ‘speex’ at sample rate ‘32000’ with id ‘21’
== Created cached format with name ‘speex32’
== Registered ‘audio’ codec ‘ilbc’ at sample rate ‘8000’ with id ‘22’
== Created cached format with name ‘ilbc’
== Registered ‘audio’ codec ‘g722’ at sample rate ‘16000’ with id ‘23’
== Created cached format with name ‘g722’
== Registered ‘audio’ codec ‘siren7’ at sample rate ‘16000’ with id ‘24’
== Created cached format with name ‘siren7’
== Registered ‘audio’ codec ‘siren14’ at sample rate ‘32000’ with id ‘25’
== Created cached format with name ‘siren14’
== Registered ‘audio’ codec ‘testlaw’ at sample rate ‘8000’ with id ‘26’
== Created cached format with name ‘testlaw’
== Registered ‘audio’ codec ‘g719’ at sample rate ‘48000’ with id ‘27’
== Created cached format with name ‘g719’
== Registered ‘audio’ codec ‘opus’ at sample rate ‘48000’ with id ‘28’
== Created cached format with name ‘opus’
== Registered ‘image’ codec ‘jpeg’ at sample rate ‘0’ with id ‘29’
== Created cached format with name ‘jpeg’
== Registered ‘image’ codec ‘png’ at sample rate ‘0’ with id ‘30’
== Created cached format with name ‘png’
== Registered ‘video’ codec ‘h261’ at sample rate ‘1000’ with id ‘31’
== Created cached format with name ‘h261’
== Registered ‘video’ codec ‘h263’ at sample rate ‘1000’ with id ‘32’
== Created cached format with name ‘h263’
== Registered ‘video’ codec ‘h263p’ at sample rate ‘1000’ with id ‘33’
== Created cached format with name ‘h263p’
== Registered ‘video’ codec ‘h264’ at sample rate ‘1000’ with id ‘34’
== Created cached format with name ‘h264’
== Registered ‘video’ codec ‘mpeg4’ at sample rate ‘1000’ with id ‘35’
== Created cached format with name ‘mpeg4’
== Registered ‘video’ codec ‘vp8’ at sample rate ‘1000’ with id ‘36’
== Created cached format with name ‘vp8’
== Registered ‘video’ codec ‘vp9’ at sample rate ‘1000’ with id ‘37’
== Created cached format with name ‘vp9’
== Registered ‘text’ codec ‘red’ at sample rate ‘0’ with id ‘38’
== Created cached format with name ‘red’
== Registered ‘text’ codec ‘t140’ at sample rate ‘0’ with id ‘39’
== Created cached format with name ‘t140’
== Registered ‘audio’ codec ‘none’ at sample rate ‘8000’ with id ‘40’
== Created cached format with name ‘none’
== Registered ‘audio’ codec ‘silk’ at sample rate ‘8000’ with id ‘41’
== Created cached format with name ‘silk8’
== Registered ‘audio’ codec ‘silk’ at sample rate ‘12000’ with id ‘42’
== Created cached format with name ‘silk12’
== Registered ‘audio’ codec ‘silk’ at sample rate ‘16000’ with id ‘43’
== Created cached format with name ‘silk16’
== Registered ‘audio’ codec ‘silk’ at sample rate ‘24000’ with id ‘44’
== Created cached format with name ‘silk24’
== Sorcery registered wizard ‘bucket’
== Sorcery registered wizard ‘bucket_file’
Cannot update type ‘bucket’ in module ‘core’ because it has no existing documentation!
Failed to register ‘bucket’ object type in Bucket sorcery
Bucket API initialization failed. ASTERISK EXITING!
== Manager unregistered action DBGet
== Manager unregistered action DBPut
== Manager unregistered action DBDel
== Manager unregistered action DBDelTree

Can someone point me in the right direction? I have spent hours Googling with no resolution. Thank you.

Polycom Alert info

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I updated the phones to the latest to verify if that was the issue.

IP Mode IPv4
IP Address 192.168.76.62
UC Software Version 5.9.2.3446
Updater Version 5.9.7.19340

I will try to remove the info= but I originally had it that way until I found the post from polycom.

I will give it a try and see if it helps.


All calls go to voicemail & firewall crashing

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First step would be to try “fwconsole chown” from the console prompt (ssh or keyboard and screen).

Polycom Alert info

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Hmm, I just found this so maybe that is why its not working. so perhaps its because I am using class 18.

The phone supports the following ring classes: default, visual, answerMute, autoAnswer, ringAnswerMute, ringAutoAnswer, internal, external, emergency, precedence, splash, and custom<y> where y is 1 to 17.

Zulu no audio when dialing through DAHDI

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Go to the Alpha version now, it will be Beta in a couple weeks and in production shortly after that. Versions on Zulu seem to be rolling through ridiculously fast.

Polycom Alert info

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Following your method did in fact work. I think ideally i didnt realize 17 was max. I changed one of the other custom ones for my need and its in fact working!

Thanks so much!

Zulu no audio when dialing through DAHDI

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Just for laughs, I copied the [macro-pinsets] context from extensions_additional.conf, edited it replacing Progress with Answer() and put the modified version in extensions_override_freepbx.conf.

It works, but there are a few unwanted side effects:

  1. On answered calls, the CDR shows duration starting from the PIN prompt.
  2. Calls abandoned before completely entering the PIN show as answered with app Read.
  3. The calling device shows all calls as answered.

(1) could be fixed by resetting the CDR after accepting the PIN.
Possibly this workaround or something similar would be useful.

How is the PIN used? Unless it’s identifying the client on whose behalf the call is made, e.g. in a law firm, there may be an automated way to get the information, improving productivity in addition to solving the present technical issue. For example, some companies use a PIN to allocate phone charges to a specific department. But if the call costs less than 1% of the cost of the person making it, this is counterproductive.

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