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-   -   SIP Broker Access Numbers - Status Thread (https://forum.sipbroker.com/showthread.php?t=2156)

martin 08-23-2007 02:23 AM

SIP Broker Access Numbers - Status Thread
 
Hi Guys,

We are creating this thread to allow members to report any problems they are having with any of the SIP Broker access numbers.

As part of your report, please provide a brief description outlining the problem and the time you were attempted to make a call.

Thanks.
.

ataboy 08-23-2007 09:10 AM

Hi Martin,

Both numbers in France are currently returning the "we're sorry etc." message for me. The last attempt was at 10:00 UK time, but I found the same yesterday. I double-checked with a UK gateway to ensure that I was using the right destination number (*010myvoxalotid#) and it worked fine.

ataboy

vijaynz 08-23-2007 09:53 AM

NZ PSTN Numbers
 
I try all 4 numbes only Auckland and Dunedin working, Wellington and Christchurch not working.

Thanks

KroontjesPen 08-23-2007 11:55 AM

The only number we have here.
 
Just checked for you: The Netherlands, Utrecht +31-30-711-0327 works fine. :)

.

ataboy 09-04-2007 07:36 AM

Quote:

Originally Posted by ataboy (Post 11983)

Both numbers in France are currently returning the "we're sorry etc." message for me.

I wrote the above but I have since read your posting that VoIP-originated calls might not work too well. I ought to point out that I had in fact been testing the French numbers over Voipdiscount from the UK prior to a pending trip to France.

ataboy

foneboxx 09-04-2007 06:01 PM

Ok can confirm that sipbroker french numbers work fine from my PSTN.

richard 09-10-2007 02:33 AM

Fort Lauderdale, USA 954-607-4525 - One Way Audio
 
As of 10:10 PM EDT today, when I recieve a call from the Fort Lauderdale US access number 954-607-4525, the calling party can hear me fine, but I cannot hear them.

Calls via the Toronto Access number does not have this problem.

Thanks

Richard

dwC 09-10-2007 07:26 AM

Vancouver PSTN
 
Hello all,

Just an FYI to everyone that Link2Voip has provided SIPBroker with a secondary Vancouver, Canada PSTN DID.

Quote:

Hello,

We have hooked up 16046309700 for use with sipbroker, I have let
sipbroker know about this new DID. Once they add it to their system it
will start working.

Steven
Cheers,

dwC

martin 09-10-2007 10:03 AM

Quote:

Originally Posted by richard (Post 12540)
As of 10:10 PM EDT today, when I recieve a call from the Fort Lauderdale US access number 954-607-4525, the calling party can hear me fine, but I cannot hear them.

Calls via the Toronto Access number does not have this problem.

Thanks

Richard

Some of the PSTN access numbers changed to allow re-invites. This was done to improve call quality due to bandwidth saturation.

I just ran a test of this number connecting to a Voxalot number and got 2 way audio. The problem you describe suggests that the VoIP destination is not handling a re-invite

What SIP-Code are you trying to reach.

A good test is to try *010600 (echo test)

richard 09-10-2007 02:12 PM

Hi Martin

I was dialing via an EMUM (e164.org) lookup. 1416907XXXX.

I didn't try using *010 + the Voxalot extension yet. Its for my Dad and he's not there to test right now.

I'ved PMed you the number.

Thanks

Richard

richard 09-12-2007 06:22 PM

Update

Hi Martin

Okay, I tried calling the Fort Lauderdale (1-954-607-4525) number again, but this time instead of using the ENUM (e164.org) alias, I used *010XXXXXX and I got the same problem :-( One-way Audio. I can hear the called party (Voxalot Subscriber - My Dad), but they can't hear me from the Fort Lauderdale PSTN gateway.

Now here is where it gets interesting.

I called, the same number using my Dad's ENUM number (1-416-907-XXXX) and using his Voxalot Extension *010XXXXXX and the called party didn't answer. Instead it went to their Voxalot Voicemail service and I left a message. The message was recorded fine. So when Voxalot's Voicemail answers a call coming in from the Fort Launderdale number, there is TWO-WAY audio, but when the Voxalot VoIP extension answers the phone, there is ONE-WAY audio.

I then tried calling via the the Fremont-Newawk number (1-510-248-0397) and using both the ENUM alias and the Voxalot Extension, I got TWO-WAY audio to the SAME Voxalot Subscriber that I am getting One-Way audio with above.

Finally I ran a test by calling MY Voxalot number (a totally new Voxalot Extension) from the Fort Launderdale number and it resulted in One-Way Audio.

I hope these tests are helpful in diagnosing the problem with the Fort Lauderdale number.

Thanks in advance for your help with this issue.


Richard

richard 09-13-2007 06:27 PM

Further Update:

I have now confirmed the both Callcentric numbers in the Miami area (Fort Lauderdale & Miami) has the one-way audio.

When I called from a VoiceNetwork Inc. number (Toronto), I get two-way Audio.

Martin, can you help me understand what changes needs to be made to the receiving VoIP (Voxalot registered) ATA so that it will work with SipBroker/Callcentric?

Thanks in advance

Richard

kurun 09-13-2007 10:01 PM

Just checked the Fort Lauderdale Sip Broker Access Number (calling via Voipcheap, directly not through Voxalot), and called my Voxalot account.
I get 2-way audio on the first try. Not sure why you experience this problem.


Quote:

Originally Posted by richard (Post 12633)
Further Update:

I have now confirmed the both Callcentric numbers in the Miami area (Fort Lauderdale & Miami) has the one-way audio.

When I called from a VoiceNetwork Inc. number (Toronto), I get two-way Audio.

Martin, can you help me understand what changes needs to be made to the receiving VoIP (Voxalot registered) ATA so that it will work with SipBroker/Callcentric?

Thanks in advance

Richard


richard 09-14-2007 02:17 AM

I think the problem is with the termination device's settings. Martin hinted something about re-invite, but I'm not sure how to fix that. This so far is only a problem with the Callcentric numbers I mentioned.

martin 09-14-2007 04:17 AM

Quote:

Originally Posted by richard (Post 12639)
I think the problem is with the termination device's settings. Martin hinted something about re-invite, but I'm not sure how to fix that. This so far is only a problem with the Callcentric numbers I mentioned.

Yes that is right. The Voxbone, LES.NET and CallCentric numbers all have re-invite enabled to enable the best call quality.

As a result, the end point needs to support a re-invite.
.

richard 09-14-2007 05:38 PM

Martin

Is there a specific change or tweak I can make to my Dad's SPA-1001 to enable re-invite. I'm not sure which of those setting that have the words "re-invite" in it I need to change.

Any help would be appreciated.

Richard

martin 09-15-2007 03:46 AM

Quote:

Originally Posted by richard (Post 12654)
Martin

Is there a specific change or tweak I can make to my Dad's SPA-1001 to enable re-invite. I'm not sure which of those setting that have the words "re-invite" in it I need to change.

Any help would be appreciated.

Richard

To my knowledge there are no specific settings. The trick to resolve the NAT issues.

Here are some recommended settings:

Code:

Setup your adapter for use behind a NAT router

    * Setup STUN on your adapter (NOTE: STUN settings are on the SIP tab)
          o Handle VIA received: no
          o Handle VIA rport: no
          o Insert VIA received: no
          o Insert VIA rport: no
          o Substitute VIA Addr: yes
          o Send Resp To Src Port: yes
          o STUN Enable: yes
          o STUN Test Enable: no
          o STUN Server: stun.voxalot.com:3478
                + NOTE: You can replace the above STUN server with any STUN server you like...
          o EXT IP:
                + NOTE: Leave this setting blank, STUN will figure this out for you...
          o EXT RTP Port Min:
                + NOTE: Normally you can leave this blank, but you can set this if you have a specific need
          o NAT Keep Alive Intvl: 45
                + NOTE: Use an value SHORTER than the "timeout" value in your router.
    * set "NAT Mapping Enable: yes"

Also check the "Symmetric NAT" setting on the member details page within Voxalot.
.

richard 09-15-2007 05:48 PM

Hi Martin

Okay, after trying the changes about, things got really bad. I got really scared when even after a factory reset, both my SPA1001's wouldn't accept ANY type of incoming. That was finally fixed, I think, by a power cycle on both ;-)

Any chance you think Callcentric will take a direct support ticket to look into this SPA1001 issue registered to Voxalot issue? There are two SPAs involved within two different NAT environments, so it may not be that unique an issue.

Thanks

Richard

f.soto 09-18-2007 02:00 AM

Problem with access number at Porto Alegre, Brazil
 
Hi,

The access number at 'Porto Alegre, Brazil' (+55-51-3251-2840) is frequently out of order when calling from Brasil Telecom (PSTN):
The symptoms are:

* returns a message like 'sorry, this number does not exist'
* returns a busy signal
* the call is not completed (not even rings)

It is very hard to get a connection.

Please help!

Thanks,
Fernando

emoci 09-24-2007 11:19 PM

As per this thread http://forum.voxalot.com/sip-broker-...html#post12872

I tested all the numbers in Spain (except National Number):

Barcelona number responds, but you cannot reach another account or Echo Test (to be fair I managed to get it to work once in 10 tries). Message: "SipCode Entered is incorrect"

Sta. Cruz Tenerife comes up busy completely...

Sevilla and Valencia Numbers work fine but did have to try 2 and 3 times respectively to get to the Echo Test (it worked ok for all trials after that)

In general with the Spain Numbers:
-You cannot dial things like *010*600 (doing this, dialing another * mid-number will give message "number has been disconnected"). So one must dial *010600 for example
-You cannot add # at the end freely. Eg. *010600# will give either "Incorrect SipCode" or "Number Dialed could not be connected"

Martin may want to get in touch with Adam VoIP.....

hust 09-26-2007 07:11 AM

Vancouver sip broker access num all down
 
Quote:

Originally Posted by dwC (Post 12542)
Hello all,

Just an FYI to everyone that Link2Voip has provided SIPBroker with a secondary Vancouver, Canada PSTN DID.



Cheers,

dwC

i am in Vancouver now, call sip acccess number 604-628-4266, encountering busy tone. call 604-630-9700, it does not work. :mad:

hust 10-12-2007 01:01 PM

Quote:

Originally Posted by hust (Post 12927)
i am in Vancouver now, call sip acccess number 604-628-4266, encountering busy tone. call 604-630-9700, it does not work. :mad:

I have forgotten to say that sip broker access number 604-628-4266 worked normally when i returned to Vancouver on OCT,4,2007.:)

StanBarber 10-13-2007 10:44 PM

I am having problems dialing SIP Broker PSTN access numbers with my ENUM number.

I have tried all these numbers:

San Antonio, TX 1-210-881-0442
San Francisco, CA 1-415-376-7297
San Francisco, CA 1-415-594-0355
San Jose, CA 1-408-538-2197
San Jose, CA 1-408-625-4997
Tulsa, OK 1-918-398-9841
San Mateo, CA 1-650-523-6897
San Rafael, CA 1-415-223-4197
Michigan 1-616-608-0509

After the prompt, I enter my ENUM
*013 882 9999 nnn nnn#

This now just gives me a busy tone. (Not 1-way audio. No ringing on either side. Just a busy signal.)

I have used these PSTN numbers with my ENUM for several months with NO problems.
I have not made any changes to my firewall and ATA.

So I had my friends do the same thing with these and other SIP Broker PSTN numbers and their own ENUMs.
Same results. Busy tone.

What is going on?

martin 10-13-2007 11:00 PM

Did you try SIPBroker - ENUM Lookup

Please PM me the ENUM number you are calling.

I just tried some numbers and got timeouts against e164.org:

e164.arpa.(3ms) ---Not Found---
e164.org.(5000ms) ---Timed Out---
e164.info.(11ms) ---Not Found---
e164.televolution.net.(2ms) ---Not Found---
enum.org.(2ms) ---Not Found---


Hopefully evilbunny can shed some light on this problem.

StanBarber 10-13-2007 11:31 PM

Thanks Martin.

I tried the SIPBroker - ENUM Lookup with
882 9999 nnn nnn

I got the results:
e164.arpa.(4ms) ---Not Found---
e164.org.(1076ms) sip: (yes this is the correct lookup entry)
e164.info.(12ms) ---Not Found---
e164.televolution.net.(83ms) ---Not Found---
enum.org.(2ms) ---Not Found---


So I have been dialing
*013 882 9999 nnn nnn#
This has not been working recently.

BUT considering the above lookup, I tried:
882 9999 nnn nnn#
and it works!
Now my end points are ringing and I am talking again. No problems.

So removing the initial *013 is what it took to make it work (again).
Why is that?
And why was it working before with the *013 ?


Do you still want me to PM you my ENUM ?

martin 10-14-2007 12:25 AM

Please try the *013 number again and PM my the exact number you dialed and I will check the logs.
.

dc2007 10-17-2007 04:25 PM

the one in china keeps going to my vm now.it used to work smoothly and rings my ata.

tremmarc 10-18-2007 11:42 PM

Hello!

Ive discovered about sip broker accidantly and recently, and ive tried to dial PTSN in muinch germany, but from different country. When dialing the number from land line, i get msg all circuits are busy now, and immediately after i got disconnected.

Does that means that only ppls from Germany/Munich can call the number or the ptsn number is closed?

The number i dialed is :+49-89-121409114

If the number is dead is there a chance to get new number in Munich?

Thanks to everyone, and cheers!

emoci 10-18-2007 11:53 PM

Quote:

Originally Posted by tremmarc (Post 13481)
Hello!

Ive discovered about sip broker accidantly and recently, and ive tried to dial PTSN in muinch germany, but from different country. When dialing the number from land line, i get msg all circuits are busy now, and immediately after i got disconnected.

Does that means that only ppls from Germany/Munich can call the number or the ptsn number is closed?

The number i dialed is :+49-89-121409114

If the number is dead is there a chance to get new number in Munich?

Thanks to everyone, and cheers!

None of the numbers are limited to people from a certain region...however you are right that the Munich number is down (just tested).... Maybe Martin can get in touch with SipService

tremmarc 10-19-2007 12:53 AM

@emoci thanks for prompt answer, really apprechiate it. That would be perfect if Martin can get ahold of Sipservice that would be perfect! Lets wait for Martin.

I believe he will have great and positive answer.

Thanks, for clearing this up for me!

Cheers!

StanBarber 10-20-2007 09:14 AM

Twin Cities, MN 1-612-235-6188 not working
 
Twin Cities, MN 1-612-235-6188

This Local Access Number is not working.

I called this number from my mobile phone, 4 different times. While I was roaming.
I never heard the "Welcome to the SIP Broker PSTN Gateway..." prompt.

emoci 10-20-2007 11:46 PM

Quote:

Originally Posted by tremmarc (Post 13483)
@emoci thanks for prompt answer, really apprechiate it. That would be perfect if Martin can get ahold of Sipservice that would be perfect! Lets wait for Martin.

I believe he will have great and positive answer.

Thanks, for clearing this up for me!

Cheers!

Test +49-89-121409114 now, it should be back up and running

emoci 10-20-2007 11:53 PM

Quote:

Originally Posted by StanBarber (Post 13507)
Twin Cities, MN 1-612-235-6188

This Local Access Number is not working.

I called this number from my mobile phone, 4 different times. While I was roaming.
I never heard the "Welcome to the SIP Broker PSTN Gateway..." prompt.

You are right the MN number from Les.net is not working ( a message "23" is given then hangs up)

In the meantime use 1-612-604-4274

Martin or someone else from the team may want to get in touch with Les.net...and will probably update on this further

martin 10-21-2007 12:20 AM

Quote:

Originally Posted by emoci (Post 13526)
Martin or someone else from the team may want to get in touch with Les.net...and will probably update on this further

Email sent.

Thanks
.

martin 10-21-2007 08:54 AM

We have been advised by LES.NET that this number is no longer supported and has now been removed it from the list.

Thanks.
.

tremmarc 10-21-2007 03:26 PM

@emoci thanks it works perfectly!

is martin so fast ? - thanks guys!


btw how often ptsn is usually going down?

martin 10-21-2007 10:51 PM

Quote:

Originally Posted by tremmarc (Post 13541)
btw how often ptsn is usually going down?

PSTN access numbers are sponsored and as such we have no real control over availability etc.

For the most part however, they have minimal problems.
.

tremmarc 10-22-2007 08:31 PM

@martin thanks for reply, however i must mention that again Munich number dosent work for me, but this time it dosent even ring, no message at all.

Can someone test that num for me to see if its really down again or its something from my end?

Thanks!

emoci 10-23-2007 04:22 AM

Quote:

Originally Posted by tremmarc (Post 13566)
@martin thanks for reply, however i must mention that again Munich number dosent work for me, but this time it dosent even ring, no message at all.

Can someone test that num for me to see if its really down again or its something from my end?

Thanks!

Hi,

Sorry about that....

The way things seem right now:

-Originally the Munich number was actually down (downed numbers happen but are rare)
-This was fixed and everything was back
-Then the Virginia data center had an issue that affected some of VoXalot/SipBroker and the websites themselves
-It seems after everything was restored with the servers, the morning of the 22nd the asterisk server handling SipBroker had some issues due to the powerdown (so all Sipbroker numbers were affected, there should've been a busy signal, or a message "number is not currently available")
-Martin restarted the SipBroker Asterisk server, everything should now be back to normal

Things don't usually happen to go wrong so close to each other in such a short time, so this should be a rare occassion. Just tested Munich number and it should now be up and running normally (hopefully it'll stay that way this time :D)

BTW, the events above may not neccessarily be the whole story or the official order of events, just my general understanding of what happended in the last 48 hours or so....but things should be settled for now

tremmarc 10-23-2007 12:56 PM

@emoci thanks bro, you answered more in depth. Muinch number again works without a problem.


I have few questions:
Im new to this, everything seems like fun, with knowledge. Im wondering what means call forwarding service usually? Does it mean call forward from account to landline/mobile phone or call forwarding means from one sip account to another sip account?

Is it possible for real landline number/mobile number becomes sort of real DID so that did can connect to sipbroker as a gateway for example and through that call other sip accounts?

Or also other way is it possible to link real landline/mobile phone so when ringing to that num actually direcly connect to particular voxalot account only somehow?

Im wondering is it possible to forward call from sip account to mobile phone directly?

Hope its not too many questions :)

Thanks alot for your detailed explanation.

Cheers!


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