Our tech support team at VoIP Supply offers great pre- and post-sales support plus provisioning, consultations, configuration, and installation help. We get a lot of VoIP hardware and software questions and would like to share the solutions with everyone.

In previous Mom’s calling Q&A series, we have discussed: What VoIP Headsets will work with Sangoma s500 and a mobile phone? Today, we have more new real questions and answers from VoIP users just like you.  

How to Set Polycom VVX Series Volume to a Persistent Level?

Q: I have a customer with 30 locations stated that the ringer is not loud enough. Is there a solution to resolve this?

A: Polycom’s VVX receiver volume resets to a nominal level after each call. This is the VVX VoIP phones’ default setting. To make the receiver volume persist across calls after you the volume setting is changed, you would need to manually edit the tx and Rx gains in the phones.

Here’s how to:

<Volume voice.volume.persist.handset=”1″/>

0 is the default number, which automatically resets your phone to a nominal level after each call ends. If you change the number to 1, the volume for each call will be staying at the same level as the previous call.

 

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Come back for more VoIP questions and answers next time! If you have VoIP questions to ask us, please submit a technical support ticket or contact our VoIP experts today at (866) 582-8591

A lot of great questions came up during the Grandstream GWN webinar this past Tuesday. We want to share the answers with all of you who may have the same questions. If you missed the webinar, don’t forget to check out the presentation deck to learn more!

Question 1: The Grandstream Cloud Management is free “as of today”.  Will there be a fee in the future?

Answer: We don’t know yet, but we will continue having a free plan.

 

Question 2: Will the Cloud be extended to control the Router as well as the Access Points?

Answer: The GWN.cloud is designed to manage APs but we may add the router and other Grandstream product in a near future.

 

Question 3: When creating a wireless bridge, let’s say between buildings, using the 7600LR, what is the maximum distance between the LRs?

Answer: It is 600 Mts (1800 Ft) but due to the nature you may want to reduce it to avoid physical elements that obstruct the signal.

 

Question 4: Are you planning to implement WPA3 support any time soon?

Answer: Yes, but no ETA at the moment. It is on our roadmap.

 

Question 5: On the router, what is the throughput of the OpenVPN?

Answer: We recently improved this spec in latest FW, the new spec will be available soon

 

Question 6: For testing the new WiFi SIP phone would you suggest the best environment would use these AP’s rather than a 3rd party AP?

Answer: We always recommend using the same brand. Grandstream is recognized for providing a hole solution as an added value. Our WP800 will support 3rd party AP’s for sure

Our tech support team at VoIP Supply offers great pre- and post-sales support plus provisioning, consultations, configuration, and installation help. We get a lot of VoIP hardware and software questions and would like to share the solutions with everyone.

In previous Mom’s calling Q&A series, we have discussed: What is the difference between Sangoma’s FreePBX system and PBXact? Today, we have more new real questions and answers from VoIP users just like you.  

What is the Password for the RenegadePBX Pro 75?

Q: What is the password for the RenegadePBX Pro75?

A: It’s “sangoma” for SSH root access, all other Renegades, the password is “voipsupply”.

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Come back for more VoIP questions and answers next time! If you have VoIP questions to ask us, please submit a technical support ticket or contact our VoIP experts today at (866) 582-8591

 

Our tech support team at VoIP Supply offers great pre- and post-sales support plus provisioning, consultations, configuration, and installation help. We get a lot of VoIP hardware and software questions and would like to share the solutions with everyone.

In previous Mom’s calling Q&A series, we have discussed: What is my password for the Grandstream UCM? Today, we have more new real questions and answers from VoIP users just like you.  

Is the Polycom IP 7000 Phone Compatible with a Shortel Phone System?

Question: We have a Shortel Phone System and would like to know if the Polycom IP 7000 conference phone is compatible.

Answer: Yes! Shortel PBX will support 3rd party open SIP phones including the Polycom IP 7000 conference phone.

STAY TUNED

Come back for more VoIP questions and answers next time! If you have VoIP questions to ask us, please submit a technical support ticket or contact our VoIP experts today at (866) 582-8591

The Sangoma FAXStation webinar was a blast! We appreciate the great interactions with our audience. We’d like to share the questions and answered that being discussed during the webinar with you.

Also, don’t forget to join our next Sangoma webinar: User Control Panel on April 5th, 2018!

Sangoma FAXStation Frequently Asked Questions

Question1: Can I port or spoof my Fax number to Sangoma’s FAXStation?

Answer: Yes, you absolutely can port numbers over, however, spoofing is not allowed. Click here to learn how the number porting works: https://wiki.sangoma.com/display/ST/Number+Porting

Question2: Is FAXStation just for the USA or can it accommodate international FAX DID’s? Can FAXs be sent to international numbers or just within the USA/ North America?

Answer: Canada is the only international country where we can accommodate DIDs and also the only place we can send international faxes.

Question3: How many FAXStation appliances can I have at one location?

Answer: As many as you have room for!

Question4: How many ports can FAXStation appliances handle?

Answer: 1-4 ports

Have more questions? Contact our VoIP expert, Brian Hyrek, at bhyrek@voipsupply.com or 716-531-4318!

Yesterday’s Grandstream UCM Series webinar was a success! Our apologies if you were not able to join the webinar due to it being at our max guest capacity. But no worries, if you missed the webinar, or if you simply want to rewatch it, here is the webinar recording and the Q&A session for your convenience!

Grandstream UCM Series Technical Training Q&A

Question 1: can the UCM be connected to an existing PBX like a Panasonic ns 700 or nec sl-100 to used the feature from the UCM

Answer: Yes, you will need to use Grandstream FXO gateways model GXW4108. We have a white paper on our website

 

Question 2: How to connect a cellular line to the UCM? Is there a Grandstream device for that?

Answer: You will need a GSM gateway (non-Grandstream) and add it to your UCM as a VOIP trunk

 

Question 3: Does Grandstream Affinity need to be installed on a PC that is directly connected to the UCM on the same network, if so how does that affect security if someone compromises the PC?

Answer: Yes, Affinity is a Windows-based software that needs to be connected to a PC on the same LAN as the GXP phone. Only models GXP16xx and 21xx supports it. You need to secure your PC as you normally do on normal operations.

 

Question 4: UCM as Open VPN server?

Answer: No, only as a client

 

Question 5: Is there a way to have an extension that is outside the PBX on another Asterisk server via SIP URI?

Answer: SIP URI is not recommended for security reasons. If you are sure to use this feature you need to “Allow Guest Calls”. Look for this setting in Sip Settings.

 

Question 6: where would we go for Grandstream support and documentation?

Answer: You can find more information at Grandstream’s product page and our technical support center.

For more information on Grandstream’s UCM Series please visit the product page or contact Joe Shanahan at 716-531-4316.

Our tech support team at VoIP Supply offers great pre- and post-sales support plus provisioning, consultations, configuration, and installation help. We get a lot of VoIP hardware and software questions and would like to share the solutions with everyone.

In previous Mom’s calling Q&A series, we have discussed: 6 Simple Ways to Fix your VoIP Audio Quality Issues. Today, we have more new real questions and answers from VoIP users just like you.

 

What is my password for the Grandstream UCM?

Q: What is my admin password for Grandstream UCM?

A: The newer devices will have a sticker on the bottom showing the admin password to log in. This is a good security measure because most of all devices factory set to “admin/admin”, which can be hacked easily.

If the “admin/admin” password isn’t working, then there should be a sticker on that device. If you have forgotten the password, it can be e-mailed to you if you have set this up in the UCM’s wizard. If in fact, you did not set this in the wizard, then you will need to factory reset the device, because there is NO way around this to retrieve their password. The CX will need to start from scratch.

 

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Come back for more VoIP questions and answers next time! If you have VoIP questions to ask us, please submit a technical support ticket or contact our VoIP experts today at (866) 582-8591

We had a lot of interactions during our Cloud 9 SIPStation webinar today! We’d like to share the questions and answers with all of you. Whether you attended the webinar or not, you may find this useful!

faxstation.png

And don’t forget to register for our next Cloud 9 webinar: FAXStaion! Click here to reserve your seat today!

Question 1: How does Sangoma define a Call Path?

Answer: Each purchased High Volume Voice Trunk will provide you the ability to make one simultaneous call into or out of your system. If you require multiple simultaneous calls, then you will need multiple trunks. You do not need a trunk for every DID. The exceptions to this are as follows:

  1.      Since Toll Free inbound calls are charged on a per-minute basis, they do not count as a concurrent call for your trunks. For example, if you have two trunks with us and have five inbound calls on your Toll Free number at the same time, you are still not using any of your High Volume Voice trunks, so you can still make or receive two normal calls.
  2.      If you have bursting enabled, you can optionally choose to pay a per-minute charge to burst beyond your call paths for both inbound and outbound calling. This lets you make and receive more simultaneous calls than your purchased trunks will support.

Question 2: Is VGA the only way to see the local monitor?

Answer: It depends on the configuration of your hardware. Since there are various configurations that you may have I am unable to provide a definitive answer to this.

Question 3: Will SIP Station work with any PBX?

Answer: It is specifically designed for FreePBX/PBXact systems, but will work with any Asterisk based system.

It will also work with other phone systems, but we are unable to help or assist with setting the service up on those systems.

Question 4: How many data centers do you have?

Answer: We have 2 data centers (one in Milwaukee and one in Phoenix). We also have multiple providers going into each one for further resiliency.

More Questions? Contact Brian Hyrek at bhyrek@voipsupply.com or 716-531-4318!

VoIP Supply offers great pre- and post-sales support plus provisioning, consultations, configuration, and installation help. We get a lot of VoIP hardware and software questions and would like to share with all of you. In the previous Mom’s calling Q&A series, we have discussed: Are Revolabs Products Supported Through End Point Manager (EPM)? Today, we are going to address your audio quality issues.

Nothing more annoying than poor VoIP audio quality. Our tech support team came up with the top 5 simple ways to fix your audio quality issues. Let’s check them out:

6 Simple Ways to Fix Your VoIP Audio Quality Issues

#1 Firewall Ports – Make sure the firewall is not blocking ports, as mentioned above.

#2 Codec Settings – Make sure you are using the correct Codec settings. A lot of providers prefer you to use G.711Alaw. using the wrong codec on any port of the call can stop audio from working in a direction if the VoIP provider is not Transcoding between the codecs.

#3 Check LAN – Make sure you have no internal LAN problems like Loops or programs that send out high volumes of multi-cast packets.

#4 Check Bandwidth – Make sure that on both your LAN and WAN/ISP you have enough bandwidth to suit the codec’s you are using for SIP/VoIP. As a rule, G.711Alaw uses 87.2kbps(8.7KBps). Also, make sure you have no users on the network using Bit torrent clients for seeding and downloading

#5 Check Broadband –  Check the Bandwidth you have available from your ISP to make sure you can make calls with a guaranteed bandwidth amount. As a rule, you want a minimum of 2Mb down, and 2Mb up.

#6 Check For Packet Loss – run a packet loss trace using either MTR (Linux) or WinMTR (Windows) to both your ISPs DNS Servers AND the VoIP providers servers you are connecting to, if you see any lost packets, then you know that your calls are going to have words and bits missing, and you will be either asking to repeat stuff or be asked to  repeat stuff.

Don’t forget to register for our QoS webinar to learn more about Quality of Service!


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Come back for more VoIP questions and answers next time! If you have VoIP questions to ask us, please submit a technical support ticket or contact our VoIP experts today at (866) 582-8591!

Today’s Sangoma Cloud 9 Zulu webinar was a success! We had a lot of great interactions. If you missed the webinar, we hope you can join the upcoming Cloud 9 webinars! We’d also like to share the Q&A session with all of you who may have the same questions.

Q1: Is SMS only available while using SIP Station?

Answer: Either SIPStation or PBXact Cloud

Q2: Is there a way to hide the SMS button if you’re not using SIPSTATION?

Answer: If it is showing up, you need to disable it under the user or group in the Admin Panel

Q3: What is the charges for Zulu when not using PBXact?

Answer: The following can be stacked together to get the number you need. Ie. 40 users can purchase 2 x 20 user packs.

  • 20 user (1 year license) – $199
  • 1000 user (1 year license) – $1999
  • 20 user (25 year license) – $399
  • 1000 user (25 year license) – $3999

Q4: Is there an estimated date for the Beta period to end and the version release will be offered?

Answer: We are expecting to leave Beta in 4-6 weeks.

Q5: With the chat functionality being written on top of Let’s Chat, is the full API functionality of Let’s Chat available?

Answer: Good catch… I may not have heard of it, but that is what our devs are using. They think in the future they will have a documented API that you can use, but that is not a guarantee.

Q6: Can admins view interactions for other accounts, including previous calls, chats (including messages), etc.?

Answer: We do not have admin audibility at this time.

More Questions? Contact Brian Hyrek at bhyrek@voipsupply.com or 716-531-4318!