Re: [asterisk-users] pay as you go t.t38 fax termination and origination
At 12:15 AM 9/28/2006 -0700, you wrote: i can't for the life of me find a pay as you go termination and origination service. there's garfachi, but they don't offer DID's in anywhere else other than CA. Any suggestions? Thanks. They do offer tollfree DID. What about using one of those? Regards, John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Now that Nufone is dead...
At 02:49 PM 7/5/2006 -0500, Rich wrote: John Kington wrote: At 09:29 AM 7/5/2006 +0300, you wrote: I have tollfree numbers with Nufone working OK. But what I like most is the regular numbers with charge/month but no charge/min on incoming calls... Did your tollfree number(s) with Nufone get cut-off in April? Did you keep the same number or did you signup for another number? I requested Nufone transfer my tollfree number in May and it is still not working (code is 77-4). I am wondering if this has happened to everyone or if my number fell through the cracks. Nope, same issue here. Luckily I hadn't advertised the number, so no big deal. Rich, Kind of good news (that I am not alone) and bad news (that you are in the same situation). I have had good availability with Nufone and did not want to go elsewhere unless it was a lost cause. I wish the company would provide some update(s) either on their website or at least in response to emails. Regards, John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Now that Nufone is dead...
At 11:09 PM 7/5/2006 +0200, Jens wrote: On 5 Jul 2006, at 19:00, John Kington wrote: At 09:29 AM 7/5/2006 +0300, you wrote: I have tollfree numbers with Nufone working OK. But what I like most is the regular numbers with charge/month but no charge/min on incoming calls... Did your tollfree number(s) with Nufone get cut-off in April? Did you keep the same number or did you signup for another number? I requested Nufone transfer my tollfree number in May and it is still not working (code is 77-4). I am wondering if this has happened to everyone or if my number fell through the cracks. I kept my toll-free number, it just took an extra step to get it working. Apparently their old provider is screwing them and taking whatever time they want to do the porting. To speed it up there's an option called force-porting, as they explained to me. The old provider will act faster for a fee of $60. NuFone is willing to eat that cost if you put those $60 into your NuFone account and put For QUICK TFN port in the payment comments, they will initiate the force porting request but leave that amount in your account so you can use it for your normal call activity. Pretty fair I'd say. The only problem is that they might not have advertised this option to every customer stuck with that particular problem. Other than that, any new toll-free DID as well as standard DID you order works instantly. jens Jens, Thanks for the force-porting tip. I use the TF for personal reasons and do not have a pressing need at the moment. My usage is so low that I suspect Nufone is losing money on me. I tried to get an update from Nufone but got no response. It does not look like a lost cause yet and I would like to stay because their service has always been available when I needed it. I even stated in my email to Nufone that I would switch to their current terms that have a monthly charge. I still got no response!?! Regards, John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Now that Nufone is dead...
At 09:29 AM 7/5/2006 +0300, you wrote: I have tollfree numbers with Nufone working OK. But what I like most is the regular numbers with charge/month but no charge/min on incoming calls... Did your tollfree number(s) with Nufone get cut-off in April? Did you keep the same number or did you signup for another number? I requested Nufone transfer my tollfree number in May and it is still not working (code is 77-4). I am wondering if this has happened to everyone or if my number fell through the cracks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nufone Tollfree Port
I tried to get an update from NuFone but Has anyone gotten their tollfree number ported to another provider by NuFone? Should I just forget it and move on? Regards, John ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone Tollfree Port
At 11:46 AM 7/1/2006 -0400, you wrote: John Kington wrote: I tried to get an update from NuFone but Has anyone gotten their tollfree number ported to another provider by NuFone? Should I just forget it and move on? Regards, John Yes we have ported our number out of there service. You need to go and sign some papers with the other provider you want and they take care of the rest for you. I planned to stay with NuFone. It looks like you chose a different provider, right? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comments on Gafachi
At 02:39 PM 3/25/2006 -0500, you wrote: Has or does anyone use Gafachi s (www.gafachi.com) origination (and termination services)? If so, what can you tell me about their call quality, and have you had any problems with the service? Comments are appreciated! Thanks, Wes Baehr Ability Business Computing, Ltd. Office: 330.882.0455 x25 Cell: 330.882.0455 x35 [EMAIL PROTECTED] ___ I have used them since July 2004 to call my daughter in London, UK, occasional call to France and here (US). Quality has been good and the only outage that I can recall was October 2004 when they disappeared for a weekend. I used IAX2 until they forced a change to SIP last January. I do not have a DID and have not tried the T38 experimental option that they recently made available. While possible, I don't think I have ever attempted more than one voice call at a time. Regards, John ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Livevoip 800 Choppy Audio
Wiley, thanks for pointing me to NuFone for tollfree DID. I was planning to report on results between LiveVOIP and NuFone. The apparent bankruptsy of LiveVOIP means that my choppy audio will probably never be resolved. I set up both DID to go through DISA and I could then use the echo test application. Everytime I tested LiveVOIP, the audio was choppy. I have not experienced any choppiness with NuFone but the echo seemed to take longer to get back to me compared to LiveVOIP. I now get a message that my call can not be completed when I call the LiveVOIP DID and I see that I can not register my asterisk to them. I am glad I did not have big dollars invested in them. Regards, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID in 513 Cincinnati
Does anyone have a recommendation for a DID local to Cincinnati (513)? I am looking for a pay as you go solution for incoming calls with light usage. I would prefer IAX but can use SIP solution. Regards, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Livevoip 800 Choppy Audio
I just signed up with livevoip for 800 DID and have very choppy audio. From PSTN to my asterisk is ok but asterisk to PSTN is terrible. I am using IAX and was assigned to server iax01.nyc.*. I do not believe it is a bandwidth problem on my end and I have no problems using iax with gafachi. I opened a ticket with livevoip but no response yet. Would I be better off using sip with them? Is there a server with better response/bandwidth? I admit that I am running a cvs head may 2004 prior to 1.x.x release. Could that be the problem? Regards, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 One way audio PSTN via Gafachi
At 08:58 PM 11/9/2004 -0500, you wrote: I have exactly the same problem, since Saturday early at the morning. I emailed to gafachi, no answer so far, also a friend of mine left them a voice message - nothing so far. Bogdan Bogdan, Thanks for the reply. The problem started on Saturday for me. I am not sure if I made any calls on Friday. My email sent Sunday has not gotten a response either. This is the first problem that I have had with them. I hope it gets fixed soon. I will copy the list in case anyone else is listening. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 One way audio PSTN via Gafachi
I am experiencing one way audio when I call through Gafachi. I can hear the person that I am calling but they can not hear me. I am able to call FWD echo test and have no problems. My daughter has called me using sjphone (sip) and firefly (iax2) with no problems. I have been using Gafachi since July using IAX2 and this is the first time I have had any problem. I did recently add some users in iax.conf so it is possible that I messed something up. I ran a debug but could not see anything that would indicate why I can hear the callee but the callee can not hear me. Can anyone confirm that they are having a similar problem or see what I did wrong from the debug output? Please note that I sanitized information like userid, password ... Regards, John IAX2 Debugging Enabled -- Starting simple switch on 'Zap/1-1' -- Executing SetCallerID(Zap/1-1, 5135551212) in new stack -- Executing Dial(Zap/1-1, IAX2/user@password/1513xxx) in new stack -- Called user@password/1513xxx Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00010ms SCall: 1 DCall: 0 [64.192.112.9:4569] VERSION : 2 CALLED NUMBER : 1513xxx CALLING NUMBER : 513yyy LANGUAGE: en USERNAME: user FORMAT : 4 CAPABILITY : 6 ADSICPE : 2 DATE TIME : 157836726 Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00010ms SCall: 16384 DCall: 1 [64.192.112.9:4569] AUTHMETHODS : 2 CHALLENGE : 110164600 USERNAME: user Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00062ms SCall: 1 DCall: 16384 [64.192.112.9:4569] MD5 RESULT : xx Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00062ms SCall: 16384 DCall: 1 [64.192.112.9:4569] FORMAT : 4 -- Call accepted by 64.192.112.9 (format ULAW) -- Format for call is ULAW Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00062ms SCall: 1 DCall: 16384 [64.192.112.9:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: VOICE Subclass: 4 Timestamp: 00129ms SCall: 1 DCall: 16384 [64.192.112.9:4569] Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00129ms SCall: 16384 DCall: 1 [64.192.112.9:4569] Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: (14?) Timestamp: 00065ms SCall: 16384 DCall: 1 [64.192.112.9:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00065ms SCall: 1 DCall: 16384 [64.192.112.9:4569] -- IAX2[a97saKWn3Ea6pX05]/1 is making progress passing it to Zap/1-1 Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 003 Type: VOICE Subclass: 4 Timestamp: 00273ms SCall: 16384 DCall: 1 [64.192.112.9:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 00273ms SCall: 1 DCall: 16384 [64.192.112.9:4569] Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 003 Type: CONTROL Subclass: RINGING Timestamp: 01916ms SCall: 16384 DCall: 1 [64.192.112.9:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 01916ms SCall: 1 DCall: 16384 [64.192.112.9:4569] -- IAX2[user]/1 is ringing Rx-Frame Retry[No] -- OSeqno: 005 ISeqno: 003 Type: CONTROL Subclass: (255?) Timestamp: 05696ms SCall: 16384 DCall: 1 [64.192.112.9:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 006 Type: IAX Subclass: ACK Timestamp: 05696ms SCall: 1 DCall: 16384 [64.192.112.9:4569] Rx-Frame Retry[No] -- OSeqno: 006 ISeqno: 003 Type: CONTROL Subclass: ANSWER Timestamp: 05699ms SCall: 16384 DCall: 1 [64.192.112.9:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 007 Type: IAX Subclass: ACK Timestamp: 05699ms SCall: 1 DCall: 16384 [64.192.112.9:4569] -- IAX2[user]/1 stopped sounds -- IAX2[user]/1 answered Zap/1-1 Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 007 Type: IAX Subclass: LAGRQ Timestamp: 10011ms SCall: 1 DCall: 16384 [64.192.112.9:4569] Rx-Frame Retry[No] -- OSeqno: 007 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10011ms SCall: 16384 DCall: 1 [64.192.112.9:4569] Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 008 Type: IAX Subclass: ACK Timestamp: 10011ms SCall: 1 DCall: 16384 [64.192.112.9:4569] Rx-Frame Retry[No] -- OSeqno: 008 ISeqno: 004 Type: IAX Subclass: LAGRQ Timestamp: 10016ms SCall: 16384 DCall: 1 [64.192.112.9:4569] Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 009 Type: IAX Subclass: LAGRP Timestamp: 10016ms SCall: 1 DCall: 16384 [64.192.112.9:4569] Rx-Frame Retry[No] -- OSeqno: 009 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10016ms SCall: 16384 DCall: 1
Re: [Asterisk-Users] iaxy vs sipura
On Thu, 09 Sep 2004 12:52:56 -0400, John Kington [EMAIL PROTECTED] wrote: What about sip softphones that use STUN? I am especially interested in UK because my daughter is going to study in London. At 02:23 AM 9/10/2004 +0900, you wrote: [EMAIL PROTECTED] If she is going to be on a residential ADSL, that shouldn't be a problem. I have friends in the UK who use both softphones and Grandstreams behind NAT on BT's ADSL service and we haven't had any major problems other than softphones locking up the PC or some silly stuff like that. Just make sure your Asterisk server is on a public IP address, or if it is behind NAT, then let her use FWD and use IAX to connect your Asterisk server to FWD. I have asterisk running on a machine with a public ip which is pointed to by dyndns. My wife's sister in France can call using sjphone (sip soft phone). I have had no problems using asterisk for these calls nor for long distance through Gafachi. (Did I just be bannished from the mailing list?) My daughter has a XP laptop with 802.11g. I don't know what Internet access she will have if any in the student housing. I am hoping she can find a wireless connection that she can share. I would like to also make it so she can walk into some place that is a hotspot and make VoIP calls. Would a soft phone using IAX be more reliable (easier) to use in this situation? I am sure she will encounter NAT at least once in either situation. Her room does have a port for telephone but I think she is on her own to get any phone service. I have signed up with FWD and also have a DID through CallUK. She can always use that to call back to the US. Regards, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxy vs sipura
At 09:54 PM 9/7/2004 +0900, Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote: For travelling, no SIP based device will be configure and forget. Perhaps if you travel only within the US, you may be lucky most of the time but pretty much anywhere else where IP addresses are scarce you will be out of luck. I have been travelling a lot on all inhabited continents, using hotel provided internet connections, internet cafes, client's office LANs, hotspots in public places, cafes, airports etc etc. The most common experience is SIP doesn't work at all and the second most common experience is SIP only works after messing around a lot. What about sip softphones that use STUN? I am especially interested in UK because my daughter is going to study in London. Regards, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
