MWeb's MTalk Webcall - Settings for 3rd party devices?

Milano

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So I have been successfully using MWeb's MTalk Webcall with my asterisk-based PBX (Yeastar MyPBX) since mid-May without any issues. I never changed any settings but today (or possibly on Friday) it stopped working via my asterisk-based PBX. Now says Unregistered whereas it used to say Registered under SIP trunks. Anyone else having issues? Did any settings change on this service/product? Any settings I should double check?
 
It seems the problem was caused by another device on my network. Although internet, email, lan, etc were all working 100% - for some reason my IP-PBX was unable to register the Webcall SIP trunk. Traced the cause of the problem to the NAS device. Glad to report problem is solved, tks :)
 
Hey Milano

What on the NAS caused the problem? I have a similar problem where registration fails and I did recently install a NAS but the SIP phone is not connected to the same router. It is actually on the modem.

Not sure what to look for on the NAS or modem?

Anyone know what ports VOIP uses - perhaps I can try forwarding the ports to the phone although since it always worked before without it I can't see how that would help, still...
 
Last edited:
I have just subscribed to MTalk. I have a Yeastar MyPBX. I received a username and password, however the MyPbx requires a Hostname/IP and a Domain. Can anyone assist with these settings.
 
I have just subscribed to MTalk. I have a Yeastar MyPBX. I received a username and password, however the MyPbx requires a Hostname/IP and a Domain. Can anyone assist with these settings.

Both must be sip.mweb.net for MTalk.
 
Milano, are you happy with the Mweb voip products?

Also, do you get a "real" number people can dial you on?

Looking for a VOIP provider but real reviews are hard to get.
 
PM Sent, didn't want to derail the OP's topic.

My original question is sorted, derail away, no probs :)

Milano, are you happy with the Mweb voip products?

Also, do you get a "real" number people can dial you on?

Looking for a VOIP provider but real reviews are hard to get.

I have only used MTalk and FnbConnect for VOIP. I found the quality of both quite good & very similar using the supplied softphones. There are no additional setup or monthly fees for customers of the specific provider. FNBconnect ran a special for a few months with very good call rates however I think they are now much the same (for cellular calls +- R1 per minute now with either provider). You get an 087 number with each. If you currently rely on multiple fixed voice lines then I would still recommend multiple fixed voice lines for redundancy.

Only when it comes to using an IP-PBX (YeaStar), I find MTalk call quality better than FnbConnect. With MTalk and my IP-PBX, I find the call quality good when using all devices ie. 3rd party softphones like Bria, FXS extensions, SIP extensions. With my IP-PBX and FnbConnect the calls always break up when I use FXS extensions and 3rd party softphones, although call quality is fine with SIP extensions. Not sure why though. Guess it won't matter if you are only using the supplied softphone without an IP-PBX.
 
Milano, are you happy with the Mweb voip products?

Also, do you get a "real" number people can dial you on?

Looking for a VOIP provider but real reviews are hard to get.

I work in the industry and am happy to send my views on half a dozen or so service providers based on experience (those who offer real numbers and porting). Mail me at [email protected] or private mail me and we can discuss.
 
Settings for Asterisk and MWeb Webcall

I have installed Asterisk 1.8.4.4 and configured it to work successfully internal and outgoing on SIP and PSTN, but I am having a problem getting SIP incoming to work. PSTN incoming works fine.

Can anybody help with the settings for sip.conf and maybe extensions.conf ??
 
Hi Ammaree, I'm having the same problem.
Did you ever figure this out?
 
I had quite a few issues with MWeb Webcall, using the softphone they provide. Often calls drop after exactly 5 minutes and most of the time incoming and outgoing calls to Vodacom and Telkom numbers, the remote party cannot hear you at all. I have been phoning and emailing Mweb technical for weeks now and still no solution from them. I suspected it to be a Codec issue with their softphone, so I installed a Atcom IP01 mini Asterisk PBX. I is quite a nice little device and comes with G.729 built in.

It worked very well and resolved most of the quality issues experienced with the softphone, but a few days ago the SIP trunk registration started giving a time out error (nothing changed on my side) and I have been unable to connect since. The big frustration with Mweb is that their engineers and product specialists cannot be contacted directly for followups after you worked with them, you must always phone the call centre and that is a total waste of time.
 
I also have had problems with their MWEB Talk client and their Android client.

For the PC client I'm now using Zoiper and the iLBC codec which works pretty well. Zoiper also has a paid for version with a licence to use G729 which I'll probably end up trying out too.

For the Android I'm now using Counterpath's Bria for Android client and Csipsimple. You have to pay for Bria and then pay extra for G729. Csipsimple is free and you can use G729 as long as you confirm you've bought the codec licence (which I assume by buying the Bria add-on you've done).

For more info on G729 licensing have a look at:
http://code.google.com/p/csipsimple/wiki/Licensing
http://www.synapseglobal.com/voip_g729_codec.php

I reckon once you've bought the licence once you should be able to use it multiple times with different applications - correct me if I'm wrong..

Also got confirmation from MWeb of their currently supported codecs (as of time of writing):
PCMU
PCMA
G729
G729A
iLBC
telephone-event
T.38
G711FB
G723
GSM

Would be nice if they also implemented silk and speex but g729 and ilbc are pretty good ;)
 
Also got confirmation from MWeb of their currently supported codecs (as of time of writing):
PCMU
PCMA
G729
G729A
iLBC
telephone-event
T.38
G711FB
G723
GSM

Would be nice if they also implemented silk and speex but g729 and ilbc are pretty good ;)

Quite a few codecs there. Would be even better if they supported G722 ;)
 
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