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Provider for Voip FAX to physical machine (not efax)

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HTTPS Fax Adapters- These have flaws like delays, and it will tell you a fax went through when it really didn’t. Both of these things make these adapters unreliable. I’ve still got about 20 or so out there that I will be moving to T38.

G711 will Fail with something as little as a 2% packet loss or any amount of real Jitter. I definitely don’t recommend even trying this. I have more details on G711/Modem stuff but I won’t bore you.

T38 works extremely well if setup properly, and if you can step the fax machine down to 14400, after months, years of perfecting T38. I’ve finally make a new breakthrough tonight actually and I’ve got it working really reliably with 4 different carriers. Ran about 100 test faxes through today to different numbers. I even combined this new breakthrough with the FreePBX Trunk Balancing Module. So now instead of it going through the same carrier every time, I can have the trunks round robin. So when the fax machine sends it’s retries…it will try to go out several different carriers in case one has a bad route. When you try the same route over and over again and expect a different result…LOL. Some Fax machines allow you to set retries to 14 times. If you don’t want to mess with adjusting the fax baud rate to 14400, you can use a T38 Relay like audiocodes which will step the speed down for you before sending to freepbx.

Online only HIPPA Compliant Fax Providers. Like the FaxAge and SRFax. Great if they work reliably. Pricing is much better than someone like E-Fax. Still expensive though.

Copper Lines/ISP lines…Someone else’s problem and $30-$50 for unlimited calling.


I changed the call recording location to a SMB folder but no files are being created

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

Redirect External Call to IAX Trunk to PBX2

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HI,

I gave a couple of conceptual queries to understand call handling to configure the below mentioned correctly.

I am running two FPBX systems (FPB14 and ASTX13) in two cities that connected by IAX2 trunk.

This is working well and I can make calls between two internal calls. I am now at the handling external calls stage down the trunk and are trying to get my head around some of the concepts.

Requirement: Receive external call to PBX1 and then redirect call to Conf IVR and PBX2

I have configured the inbound route on PBX1 (using DID) to the IAX trunk destination. I have then tried to configure an incoming route on PBX 2 using the same ext DID number to the Conf IVR. This is how the PBX1 used to operate but we are now shifting all of the conf rooms to PBX2.

A review of the logs on PBX2 shows the call coming into the system nicely but it is showing up as an internal call. This makes sense as the IAX2 on the PB2 is configured with “context=from-internal”:

– Accepting AUTHENTICATED call from 203.59.XX.XXX:4569:
– > requested format = gsm,
– > requested prefs = (gsm),
– > actual format = gsm,
– > host prefs = (gsm),
– > priority = mine
– Executing [612915XXXX@from-internal:1] ResetCDR(“IAX2/iaxVultrXXX-3312”, “”) in new stack
– Executing [612915XXXX@from-internal:2] NoCDR(“IAX2/iaxVultrXXX-3312”, “”) in new stack
– Executing [612915XXXX@from-internal:3] Progress(“IAX2/iaxVultrXXX-3312”, “”) in new stack
– Executing [612915XXXX@from-internal:4] Wait(“IAX2/iaxVultrXXX-3312”, “1”) in new stack

On the above basis, I believe that in inbound route on PBX2 is irrelevant in this context as its already an internal call! I would appreciate any suggestions to the best way configure this?

Thanks
Mark

Transfer to VM REST-API does not accept direct extension entry

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I just change my direct VM prefix from * to 0, and then tell people if you want to XFER to VM, just put the 0 in front of the extension. Easy enough. With phones like yealink you can also set a VMXFER prefix BLF button. So they hit XFerVM which prefixes a 0 then enter the extension, or press the blf key.

I use 0 instead of * because * interfers with too many other possible feature codes.

Provider for Voip FAX to physical machine (not efax)

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Over the years, i chose hylafax+ and t38modems bypassing any intermediary (like asterisk), it worksat least 3 nines with any real T38 provider the failures comparable to t30 to the same numbers, t38modem is almost impossible to compile on RH though , use a 5 dollar vultr/do Debian instance and just apt install t38modem You also get trivial email2fax/fax2emailVworking with a few hours of RTFM

Redirect External Call to IAX Trunk to PBX2

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For many internal extensions like ring groups, the context from-internal is not sufficient for a tie-line, ext-local would likely work, but is extremely risky and can cause conflicts between the PBI, caveat emptor

No assisted transfers are possible, Asterisk 13.22.0, FreePBX Utility 14.0.5.5

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… probably i should specify, that voip-LAN is behind a pfsense(?).

ELindemann

No assisted transfers are possible, Asterisk 13.22.0, FreePBX Utility 14.0.5.5

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  1. Any reason why you aren’t using the phone side transfers instead of using the DTMF feature codes? Does it happen when you use the phones built in assisted transfer?
  2. What happens if you dial *2EXT and don’t dial pound? This should still work just 2-3 second delay.
  3. What happens if you change the *2 to a custom different feature code to isolate possible feature code compatibility issues.
    4.When the dial *2EXT# are they actually waiting for someone to pickup before hanging up the call? Or are they doing a partial assisted transfer.
  4. On your queue what is your setting for Skip Busy Agents and your call waiting settings on the phones.
  5. What is your setting for queues-restrict dynamic agents?

Redirect External Call to IAX Trunk to PBX2

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Thanks, I will do some homework on this.

No assisted transfers are possible, Asterisk 13.22.0, FreePBX Utility 14.0.5.5

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This actually could be related to an issue I posted with asterisk segfaults. I asked for an upgrade to 13.24.1 for freepbx 13 but they denied. Since freepbx13 isn’t supported anymore but I really don’t want the UCP change with 14/15 as it’s still broken in my eyes without templates, also any type of upgrade would also likely reset UCP passwords for several hundreds of extensions since they are encrypted unless Freepbx 15 restore can keep them somehow. I know asterisk 13.25 or 13.26 had some major issues with segfaults so they dropped it back down to 13.22 I’m running 13.23.1 and I haven’t had any segfaults in a while after stopping use of FOP2 and getting rid of CEL and dashboard stats.

"Current levels of asterisk for freepbx 13 is 13.23.1 but a major bug was discovered that caused asterisk to randomly crash and the fix was released in 13.24.0
2018-10-22 07:47 +0000 [91630834f7] lvl <digium@lvlconsultancy.nl>
*** app_queue: Revert broken queue channel reference patch**
Revert commit 6409e7b11a2310196a9978b30a6b79e2760be592, and add
NULL checks for all app_queue event handling code.
Related issues: ASTERISK~25185, ASTERISK~27006, ASTERISK~25844

No assisted transfers are possible, Asterisk 13.22.0, FreePBX Utility 14.0.5.5

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This is sample senarios described in ASTERISK~25185. A patch was created which broke more stuff. Then they finally reverted the patch in 13.24.1 and added queue null checks so the queues didn’t break. From there I don’t know because no one has explained what the major issues were with 13.25 or 13.26 recently that made them back track down to 13.22. Did they go all the way to 13.22 to be safe>? Where did the major issues start.

Hello,

A segfault happens in asterisk in the following scenario:

Given I have 2 users, Alice and Bob, each with a SIP phone (using chan_sip)
Given I have a queue Foo, with member Bob
When Alice calls the queue Foo
And Bob answers the calls
And Alice then does a direct transfer (SIP native) to another extension
Then asterisk segfaults

There is another similar scenario where a segfault happens:

Given I have 3 users, Alice, Bob and Carol, each with a SIP phone (using chan_sip)
Given I have a queue Foo, with member Bob
When Carol calls Alice
And Alice begins an attended transfer to the queue Foo
And Bob answers the calls
And Alice finalize the attended transfer
Then asterisk segfaults

This happens in a systematic way. I’ll attach some gdb output for each scenario.

IVR Key Press / Digit Timeout can't be changed from 3 seconds

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The digit timeout it’s referring to 3 is set in one of the IVR php pages but it isn’t what you think it is. It’s the inter-digit timeout. Meaning the IVR has options
Press 1
Press 2
Press 3
Press 101
Then I press 1 it will wait 3 seconds to see if I’m going additionally enter 01 to fully enter 101
If I press 2 the interdigit timeout has no affect as long as I don’t have direct dial extensions enabled.

Are you trying to edit the IVR entry timeout where they don’t press anything, or the inter digit timeout in between button presses because increasing the interdigit timeout to 10 seconds would be very bad, users would press 1 and nothing would happen afterwards for 10 seconds. but the IVR timeout to 10 seconds is pretty normal and ok.

LetsEncrypt 'Token did not match'

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The only thing I can suggest is running a tcpdump on your machine when mirror1 reaches out to it, and make sure that your machine IS ACTUALLY RETURNING the correct data. The code (from what I can remember) is super simple - it’s basically just a if ($token === $expected).

If what you’re seeing in tcpdump matches what you expect to see, you’ll have to open a ticket on issues.freepbx.org about it.

Domain account to login softphone

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

Rendimiento de FreePbx en Equipos AMD

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


Polycom IP Series phones and HTTP/HTTPS Provisioning

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Cisco 7942g wont register with FreePBX

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Custom dialplan

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This is, but i’m using freePBX .conf files, so i need to use a macro.

I guess there is one for post-call context

S305 Rebooting on Login

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CallerID settings

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