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Audio issues - dropped packets


Chan_Sip VS PJSIP

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As far as the Polycoms are concerned, this BLF issue actually might be known. They had a bug in previous versions (I think 5.4 or older).

I picked up a new client last year, they said BLF never worked at their previous carrier and the carrier couldn’t figure out way. I upgraded their Polycoms to 5.7 (current last year) and boom, all their BLF issues went away.

That could be related to this. People tend to forget to keep their phones firmware current.

BLF's not working right with Yealink 83.0.X firmware. Response from Yealink

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What software is the Polycom VV410 running? Ive told you before I’m 90% Polycom / 10% Yealink when it comes to deploying phones. I have yet to see these Polycom issues you keep referring to.

I can’t 100% speak on Yealink, as I haven’t moved them to the latest firmware yet.

Can't forward call outside to cell phone number

Can't forward call outside to cell phone number

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So this number pattern can be dialed normally without issue: 049681222? You can pick up a phone a call it? Because you’re carrier is saying it’s bad.

Dynamic On-Call Destination

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Wrong. Tony said that having agents just as a number breaks things. Meaning normal extensions shouldn’t be any issue. And I already said that:

No way of logging in/off a queue without custom code.

Incorrect. It was removed in FreePBX 14 which was released a little more than a year ago (It still exists in FreePBX 13)

Let me explain.
Lorne said here that:

I responded that with the new feature codes you cannot login remotely.

Tony later said, that having an agent just as numbers will break things. I understand that.
So I asked Tony how about keeping this feature just for regular extensions, forget about cellphones.

Then Dave replied to me of ways you can add cellphones in a queue. Why? idk. We were talking about keeping this feature code with regular extensions.

Then you asked why it’s an issue of not having this feature code in place.

Don’t forget. I’m not alone on this issue:

Dynamic On-Call Destination

Dynamic On-Call Destination

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A feature code to add GUI destinations for Queue toggle would probably not go amiss. In the mean time, these lines will get you what you need:

[custom-queue-user-toggle]
; Use with FreePBX Custom Destination with dial string format
;      custom-queue-user-toggle,<Queue_No>,1

exten => _X.,1,noop(Entering user defined context custom-queue-user-toggle in extensions_custom.conf)
exten => _X.,n,Set(QUEUENO=${EXTEN})
exten => _X.,n,Set(QUEUEUSER=${CALLERID(number)})
exten => _X.,n,Goto(app-queue-toggle,s,1)

BLF's not working right with Yealink 83.0.X firmware. Response from Yealink

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When I did a brief test with Yealink 84 firmware, I experienced the same issue described above. If you reload asterisk, the BLF keys did stop working until you restarted the phone.

In regards to Polycom, just searching for RFC4235 on Polycoms site yields a reference in Firmware 6.3.0:


“To notify the call states of a monitored extension, your phone expects the NOTIFY message body to include
a dialog-info XML (Content-Type: dialog-info+xml) that is compatible with RFC4235.”

Maybe a lot more of the phone manufacturers are beginning to force that standard?

I agree with Tony’s statement that manufacturers shouldn’t just go changing basic functionality without giving some type of notice. On the same token, if there is a defined standard, I would assume everyone, over time would move to that standard for compatibility. The question is why was there no notification about changing that standard?

Seems like the next step would be for Asterisk to look at the RFC4235 standard. Or work through the manufacturer channels and notify them of the issue and work together toward a resolution.

Any chance someone from the Asterisk team could chime in on their thoughts on the issue?

On a side-note, I do appreciate the Sangoma team at least responding to the issue and commenting on the forums. This reflects the fact that the company is taking time to listen to the community. Thank you for any help!

BLF's not working right with Yealink 83.0.X firmware. Response from Yealink

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This is for the Obihai phones. Remember, Polycom bought them. I’m referring to the Polycom UC software that runs the VVX series phones, 301, 401/411, 501/511, 601/611. That is on 5.8 right now.

Two different phone lines, two different firmware releases, one company (now).

Dynamic On-Call Destination

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I actually tried this a long time ago. It added the agent as invalid. No calls were processed.

[agent-number]
exten => s,1,Answer()
exten => s,n,Read(agent,agent-login,11,,1,5)
exten => s,n,Playback(you-entered)
exten => s,n,SayDigits(${agent})
exten => s,n,Read(digi,if-correct-press&digits/1&otherwise-press&digits/2,,1)
exten => s,n,ExecIf($["${digi}"="1"]?goto(agent-login,s,1))
exten => s,n,Playback(please-try-again)
exten => s,n,Goto(agent-number,s,1)

[agent-login]
exten => s,1,Answer()
exten => s,n,Set(QUEUENO=500)
exten => s,n,Set(QUEUEUSER=${agent})
exten => s,n,Goto(app-queue-toggle,s,start)

The only way I was able to accomplish it is by using something like this.

[agent-login-number]
exten => s,1,Answer()
exten => s,n,Read(agent,agent-login,11,,1,5)
exten => s,n,Playback(you-entered)
exten => s,n,SayDigits(${agent})
exten => s,n,Read(digi,if-correct-press&digits/1&otherwise-press&digits/2,,1)
exten => s,n,ExecIf($["${digi}"="1"]?AddQueueMember(500,Local/${agent}@from-queue/n)
exten => s,n,ExecIf($["${digi}"="1"]?playback(agent-loginok))
exten => s,n,ExecIf($["${digi}"="1"]?SayDigits(${agent}))
exten => s,n,ExecIf($["${digi}"="1"]?playback(goodbye))
exten => s,n,ExecIf($["${digi}"="1"]?Hangup())
exten => s,n,Playback(please-try-again)
exten => s,n,Goto(agent-login-number,s,1)

No Call history in SuiteCRM

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I am running the 13.0.6.13 version of CRM and still no call history in SuiteCRM. How long is the period of time between a call ending and it appearing in the CRM?

Configuring TEO IP Phone 7810 TSG Series

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

Dynamic On-Call Destination

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If you do a queue show xxx you may well see the word ‘invalid’ in parentheses, but that refers to the fact that Asterisk is unable to get the device status for the external queue agent (i.e. there is no hint), but the agent should still be able to receive queue calls.

Issue with Freepbx when loosing Internet connecvity

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I have notice a strange behavior that cause loosing all phone calls on freepbx system based on distro when Internet connection is down. for some reason system become unstable and very slow plus all phone calls are being not answered and a huge lag on all extensions

Is there any kind of activation or online check that runs in background and may cause this ? we notice this because switching to another gateway with Internet connection solved the issue immediately

Asterisk version: Asterisk 13.18.4 built by mockbuild @ jenkins7 on a x86_64
linux version 3.10.0-693.11.1.el7.x86_64
freepbx version FreePBX 14.0.4.5


Issue with Freepbx when loosing Internet connecvity

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If I remember asterisk slows down in an attempt to resolve the urls you have for sip hosts. One way to handle this is to replace the url with an ip. There are obvious drawbacks to this as well but asterisk is not a fan of not having the internet and trying to resolve web addresses.

Issue with Freepbx when loosing Internet connecvity

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THanks for the quick reply but actually all phones are local and we use only PSTN lines, there is not sip addresses, for sure we want to keep Internet but you know this call center is active for 24hours and sometimes ISP has failures

Dynamic On-Call Destination

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I’ll try it again, but I remember from previous time that the queue did not call the agent.

Issue with Freepbx when loosing Internet connecvity

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What happens if you remove the PBX’s access to the Internet? If you are using POTS lines for PSTN connectivity (or PRI) and the phones are local, what happens when the PBX is only allowed to be local?

Internet provider changed - problems with remote extensions

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Did you make sure or check that the routers/firewalls where the remote phones are have the proper information due to the ISP change?

People tend to forget that if the PBX is in one location behind NAT and the phones are at another location behind NAT that they have to make the needed NAT rules/updates on both sides.

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