Dialing Problem -- sipbroker and pstn termination
 

I anyone having dialing problems with dialing out of us.voxalot.com today?

I can dial in from callcentric via sipbroker and ring my ata, and I can do a *600 echo test, but I can't dial out at all now. This is strange because I did many sipbroker test calls to other servicies this morning and things worked great -- then it just stopped working.

I get "We're sorry, the number you have dialed could not be connected, please try again." I get this with any external test number such as sipbroker *266300 for example, or the sipbroker test number *011 188888. Same if I tried to dial through to pstn number which for me should terminate with callwithus.

I've power cycled by dsl router, and my gateway, but nothing seems to help. I'm starting to look at settings on the ATA, but just not sure where to look as it stopped working when I'd not made any changes.

Any ideas were to start looking. Is there some sort of daily call limit? Is this a voxalot issue? Is it a config issue on my end?

Thanks
Rob

Update on dialing issue
 

I re-wrote the dial plan on my spa3102 to include a special prefix to go directly to sipbroker, and another to callwithus -- and I can call out fine directly -- still not through voxalot.

Seems like maybe a voxalot issue. Thoughts?

Rob

Me too
 

Me too I'm having the same problem at least on the last 2 days.
cannot connect to many sipbroker addresses, and some times it connects but with terrible sound quality

All on us.voxalot.com

Please help

Can't dial out of voxalot today
 

Any ideas?

Still can't dial out of voxalot.

Rob

Do you notice that problem with both US proxies?
proxy01.us1.voxalot.com = 64.34.163.35
proxy02.us1.voxalot.com = us.voxalot.com = 64.34.173.199

Are you intending for your voice packets to travel through Voxalot or directly between SIP endpoints? Do you have a way to discover which of the two possibilities is actually happening?

ok
 

Hello boatman,
In my case I've only been using us.voxalot.com. Never tried the other US proxy.
I mainly use Voxalot to connect both callback legs (callback which is performed by LIV), but even when I tested calling from a softphone I had similar problems calling Sipbroker addresses, on Voxalot

If I got your last question right, so in my acc: "Symmetric NAT " is set to "No" and "Optimize Audio Path" is set to "Yes" (that's how it worked best for me) - but I've been using Voxalot under the same settings for about 1 year without problems, and nothing in my usage or destinations has changed on the last days.

It really seems to be some problem between Voxalot and Sipbroker
Many times it doesn't connect ("number you have dialed could not be connected, please try again"), and even when it finally does, it returns too bad voice quality for talking, and almost no way to send DTMF

Hope Voxalot team can debug and fix that

I made a test call through us.voxalot.com using SipBroker to reach someone on Gizmo5. The call connected quickly and the audio was fine.

Edit:
I made more test calls, this time to both Sipbroker numbers listed here. The numbers belong to Advanced Science and Technology Institute, Department of Science and Technology, Republic of the Philippines. They have a recorded message there. The audio sounded great.

Connection Update
 

As for me it seems that my issue is perhaps different from Jeannie's. I shut off my ATA and tried to connect directly with ekiga -- and if I do that -- I can call out to the sip broker test message *011 188888. I guess this means that it's some setting in my ATA.

I'm a bit baffled by that as I don't think I changed anything with the ATA when this started happening. I guess I'll go over it one more time tomorrow and see what I can find -- it's a bit late here now.

If anyone has an idea where to look in the settings let me know. I have an SPA3102. I'm thinking of playing with the dial plan (trying a really simple one), and also there are some send and recieve calls without registering selections I may try on and off.

I was on us Proxy 1 and had dial out and dial in problems. Have just changed to us Proxy 2 and all is working fine now. I recomend that you reconfigure to proxy02.us1.voxalot.com = us.voxalot.com = 64.34.173.199 Bob

also got a similar problem accessing remote spa FXO using sip broker PSTN number and calling from Australia ....reading from this post could specific devise default settings with regard to signal parameters Like bellcore for USA and china be hampering correct decoding of DTMF tone as againts PSTN devise here in australia and UK having inconpatible DTMF tones with that of USA standards

SPA3102 and Voxalot -- Solution to Calling Problem
 

I want to thank everyone for their suggestions.

I've found the problem -- and I'm going to post it here just in case anyone else runs into this. As you can imagine it took some time to track this down. It turns out that you may need Line1>Supplementary Service Subscription>Block CID Serv set to NO if your using an SPA3102 with Voxalot (I seem to require this setting).

If you have this problem the symptoms of this problem are that you can't call out via sipbroker, or via a PSTN provider ("We're sorry, the number you have dialed could not be connected, please try again."), and when you use *500 your asked for both your mailbox number and password. Basically it looks like your not really authenticated for outgoing calls. However -- you can still call in for example from Callcentric via sipbroker.

One thing I am confused by -- the voxalot faq for the 3102 shows a screen shot that shows Block CID Serv set to YES -- but that definitly does not work for me. Maybe Voxalot has been changed some since this screen shot or maybe I'm special -- don't know.

The other thing I will say -- I also set up the SPA3102 syslog service using a Linux based syslog server. It seems that the SPA implementation is a bit strange as not all of the messages have a facility/severity code on them -- and they use a variety of different facilites: LOCAL0, LOCAL2, and LOCAL3, and severities INFO and DEBUG. However, as I said some key messages like REGISTER and SIP responses are not logged with the standard format (don't have either a facility or severity). In the end I found that using wireshark to monitor the incoming udp packets was simpler. Anyway -- just in case anyone needs to do this.

Another confusing thing: In the syslog messages I found that about 1/3 to 1/2 the time the REGISTER would return a SIP 401 Unauthorized. Is this normal? Everything seems to work -- the ATA just does another REGISTER immediatly which always works -- but it seems strange unless this is just part of the DIGEST auth process?

Anyone has any thoughts let me know.

Rob

i have also the spa3102 registered with voxalot, , about the BlockCID serv. i have it : yes (default) and it's working fine, i readed on another forum that is onlyimportant for those who use some 2-Stage dialing need to keep it off same for Cfwd All Serv.

but idon't know the cause