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Hey folks, hoping to find out what others are experiencing (those who have implemented STIR/SHAKEN) As many of you also did last year, we too went through a STIR/SHAKEN implementation; getting set up with an OCN, integrating with iConectiv, choosing a CA (Sansay in my case) and building out an...

Vitelity has castrated their SMS service severely. I think this is mostly due to heavier regulation of SMS providers in the US. Their SMS API is still there, but only supports TLS v1.0 which most modern OSes will reject outright with cURL even the Python Requests module. I've stopped using it...

Here's my old guide from about 10 or so years ago: https://www.voip-info.org/forum/threads/sonicwall-firewall-with-asterisk-and-freepbx.13114/ Basically the gist is: Bump up the UDP Timeout for the VoIP Traffic as the default SonicWALL timeout of 30 seconds will cause clients to go...

First Off - Wi-Fi IP Phones need a good wireless infrastructure behind them. What I mean by this, is an infrastructure where access points use Layer-3 Roaming/Tunnelling to a controller. Cisco LWAPP, or Aruba LWAPP would be examples of this. Standard AP's where the voice SSID is Layer-2 bridged...

This is an excellent tip - I wasn't aware of the Milliwatt() function. I have a Sine Wave Sweep that I use for exactly this purpose - I needed it to do some research and reports on the effects of Jitter/Loss with various SIP Phones. I'll attach it here just in case anyone else would like to use...

I started this over on the FreePBX forum - I hate spraying different forums with the same question, but it's been some time with Zero response over there, and besides, I like you guys way better over here =) http://community.freepbx.org/t/freepbx-rest-voicemail-on-aastra/26370

I've just done a pretty in-depth comparison on Sound Quality comparing Aastra, Yealink and Cisco SPA series. Specifically I was digging for best Jitterbuffer/Packet Loss Concealment implementation due to the fact that Cable Modems are really starting to ruin VoIP quality. The Cisco seems to have...

To be honest, using the OSS EPM with the default template makes provisioning the Polycom IP6000 a snap. We don't do Polycoms specifically due to the "learning a new 'language' " factor, however after trying some 3rd party Conference Phones (ClearOne Max IP) I now will only sell Polycom...

Hmm... I'm not sure I'm following Andy with the "directory where the xml file resides." I'm essentially trying to Map a button or "DSS Key" as Yealink calls them, to a URL pointing to a PHP script on the PBX that would spit out the correct XML for the Yealink to interpret and draw out on the...

*45xxx*qqq where xxx is extension and qqq is queue number. Just set up some BLFs today to allow quick login/logout using that method.

What's funny, is that G.722 actually consumes the exact same 64kbps that G.711 does - it just runs a wider Khz band, at 16000Hz instead of the standard 8000Hz used in G.711, which gives you that lower bass and higher treble. It's just that most carriers I've run across generally only support...

Hey Yealink Guru's out there - (wardmundy) ! Help! I may be re-inventing the wheel here, I'm evaluating some Yealink phones (T46G and T42G) and wanted to port over my Cisco SPA Directory to the Yealinks, and have been running into trouble. The code has been added to GitHub...

Great thread guys, I've never used Asterisk Phonebook before (I'm guessing I should!) so I haven't tried rolling that code back into the directory script. (I'm definitely *not* a developer! Just tinkering here!) I'm a big fan of the SPA phones, I've been doing some heavy evaluation on those vs...

VoIPMonitor. http://www.voipmonitor.org

Another thing to add to Bill's excellent advise is to look at your gateway's NAT settings. You want to adjust your UDP timeout to be fairly high (600 seconds maybe) for SIP. SONICWALLS for example notoriously have a low default UDP timeout of like 30s which causes exactly what you describe.

I would not use server-side jitter buffer. What you likely will need to do is packet capture the calls at various points in the network to see where the packet loss/jitter is occurring. When you say you are running QoS everywhere, what does that mean? Are you marking/policing at your gateway...

I generally use G.722 for internal calls and ulaw on the sip trunk.

Yes I agree that this is a NAT issue. Seeing as how port 5060 is seen as a source port on x101 I suspect that a SIP/ALG type feature may be turned on on the gateway. If you do a 'sip show peer 101' or whatever on the PBX does it show the internal address of the phone? What type of gateway are...

Cool, FYI I think the setting on the Grandstreams to help prevent unauthorized calls is: Allow Incoming SIP Messages from SIP Proxy Only

+1 for Joe here. We run our setup exactly as described. Joe set us up with redundant instances of A2billing, and we have virtualized PBX instances behind that. Each customer is tracked through a2billing and invoiced separately.