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. . |
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 |
NZ PSTN Numbers
I try all 4 numbes only Auckland and Dunedin working, Wellington and Christchurch not working.
Thanks |
The only number we have here.
Just checked for you: The Netherlands, Utrecht +31-30-711-0327 works fine. :)
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ataboy |
Ok can confirm that sipbroker french numbers work fine from my PSTN.
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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 |
Vancouver PSTN
Hello all,
Just an FYI to everyone that Link2Voip has provided SIPBroker with a secondary Vancouver, Canada PSTN DID. Quote:
dwC |
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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) |
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 |
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 |
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 |
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:
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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.
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As a result, the end point needs to support a re-invite. . |
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 |
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Here are some recommended settings: Code:
Setup your adapter for use behind a NAT router . |
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 |
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 |
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..... |
Vancouver sip broker access num all down
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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? |
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. |
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 ? |
Please try the *013 number again and PM my the exact number you dialed and I will check the logs.
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the one in china keeps going to my vm now.it used to work smoothly and rings my ata.
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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! |
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@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! |
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. |
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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 |
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Thanks . |
We have been advised by LES.NET that this number is no longer supported and has now been removed it from the list.
Thanks. . |
@emoci thanks it works perfectly!
is martin so fast ? - thanks guys! btw how often ptsn is usually going down? |
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For the most part however, they have minimal problems. . |
@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! |
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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 |
@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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