Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
Hasn't anyone noticed that LiveVoip seems to happily blame just about everything on Asterisk? FWIW, I have experienced the same type of problem on a Sprint cell phone and also using a residential VOIP account with Broadvox. Both were able to correct the problem at THEIR end. Since no one else on this list seems to be complaining about the problem using provider's other than LV, I would suggest sacking them and getting DIDs from some other place. Seems like that is always the first thing they suggest too so they must not be that interested in your business. -mark On Mar 2, 2005, at 11:06 PM, Ryan Laginski wrote: Hi, I am experiencing the same problem as you. Ringback works great with the pstn or any other voip provider, but not with livevoip. I've just upgraded to 1.0.6 to see if that resolves the problem, but it has not. Please post back if you find a solution, I'll do the same. Thanks, -Ryan On Wed, 2 Feb 2005 13:25:29 -0500, Brian Dingman [EMAIL PROTECTED] wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Re: No ringback over IAX - LiveVoip
Mark Eissler wrote: Hasn't anyone noticed that LiveVoip seems to happily blame just about everything on Asterisk? FWIW, I have experienced the same type of problem on a Sprint cell phone and also using a residential VOIP account with Broadvox. Both were able to correct the problem at THEIR end. Since no one else on this list seems to be complaining about the problem using provider's other than LV, I would suggest sacking them and getting DIDs from some other place. Seems like that is always the first thing they suggest too so they must not be that interested in your business. -mark On Mar 2, 2005, at 11:06 PM, Ryan Laginski wrote: Hi, I am experiencing the same problem as you. Ringback works great with the pstn or any other voip provider, but not with livevoip. I've just upgraded to 1.0.6 to see if that resolves the problem, but it has not. Please post back if you find a solution, I'll do the same. Thanks, -Ryan On Wed, 2 Feb 2005 13:25:29 -0500, Brian Dingman [EMAIL PROTECTED] wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 is this using their asterisk city, or just a straight sip account?? Greg ___ 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] Re: No ringback over IAX - LiveVoip
It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. On Fri, 04 Mar 2005 12:23:45 -0500, Cirelle Internet Products [EMAIL PROTECTED] wrote: Mark Eissler wrote: Hasn't anyone noticed that LiveVoip seems to happily blame just about everything on Asterisk? FWIW, I have experienced the same type of problem on a Sprint cell phone and also using a residential VOIP account with Broadvox. Both were able to correct the problem at THEIR end. Since no one else on this list seems to be complaining about the problem using provider's other than LV, I would suggest sacking them and getting DIDs from some other place. Seems like that is always the first thing they suggest too so they must not be that interested in your business. -mark On Mar 2, 2005, at 11:06 PM, Ryan Laginski wrote: Hi, I am experiencing the same problem as you. Ringback works great with the pstn or any other voip provider, but not with livevoip. I've just upgraded to 1.0.6 to see if that resolves the problem, but it has not. Please post back if you find a solution, I'll do the same. Thanks, -Ryan On Wed, 2 Feb 2005 13:25:29 -0500, Brian Dingman [EMAIL PROTECTED] wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 is this using their asterisk city, or just a straight sip account?? Greg ___ 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 -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1953 ___ 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] Re: No ringback over IAX - LiveVoip
--On Friday, March 04, 2005 11:58 AM -0600 James Taylor [EMAIL PROTECTED] wrote: It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. There is clearly an issue between LiveVoip and Asterisk. The LiveVoip people claim that they have been ignored on the Asterisk List and they indeed blame Asterisk for everything from lost dtmf to other failures. That said, they are the only company I've found that offers inbound DIDs with multiple simultaneous calls, suitable for a call center or calling card application. Most others limit you to one, or a small few, inbound paths. They (Level 3, actually) also have the widest coverage for DIDs in the US. At the current level of service, LiveVoip is not going to get my business. If I can find anybody else to provide my inbound service, I'm very interested in talking to them. /edg ___ 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] Re: No ringback over IAX - LiveVoip
On Fri, 04 Mar 2005 10:12:05 -0800 Ed Greenberg [EMAIL PROTECTED] wrote: --On Friday, March 04, 2005 11:58 AM -0600 James Taylor [EMAIL PROTECTED] wrote: It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. There is clearly an issue between LiveVoip and Asterisk. The LiveVoip people claim that they have been ignored on the Asterisk List and they indeed blame Asterisk for everything from lost dtmf to other failures. That said, they are the only company I've found that offers inbound DIDs with multiple simultaneous calls, suitable for a call center or calling card application. Most others limit you to one, or a small few, inbound paths. They (Level 3, actually) also have the widest coverage for DIDs in the US. At the current level of service, LiveVoip is not going to get my business. If I can find anybody else to provide my inbound service, I'm very interested in talking to them. /edg Seems kind of starnge that they are the only ones having this problem. I am pulling an account from Voicepulse using IAX and not have a problem at all. Maybe they need to call Digium, or some other contractor, and pay someone to set it up for them correctly since it is obviously they cannot accomplish this. ___ 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] Re: No ringback over IAX - LiveVoip
Seems kind of starnge that they are the only ones having this problem. I am pulling an account from Voicepulse using IAX and not have a problem at all. Maybe they need to call Digium, or some other contractor, and pay someone to set it up for them correctly since it is obviously they cannot accomplish this. I had some issues with VoicePulse as well with IAX. Don't remember exactly what they were... but I believe it may had been an IAX trunking issue. -forrest ___ 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] Re: No ringback over IAX - LiveVoip
Ok, time for me to ask my own newbie question. :) I've done some digging on ringback, and if I'm understanding it correctly, it's the ring tone that the caller hears when dialing another person. What exactly is it that people are finding now working with LiveVoip? Everyone says 'ringback isn't working', but nobody's really explained exactly what's happening. At least not that I've been able to find. I have a DID with them, and it works just fine. Dialing out works fine, when people call in it works fine. I'm interested in knowing what it is that isn't working, and if I can re-create it on my system... regards, Paul - Original Message - From: Ed Greenberg [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 11:12 AM Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip --On Friday, March 04, 2005 11:58 AM -0600 James Taylor [EMAIL PROTECTED] wrote: It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. There is clearly an issue between LiveVoip and Asterisk. The LiveVoip people claim that they have been ignored on the Asterisk List and they indeed blame Asterisk for everything from lost dtmf to other failures. That said, they are the only company I've found that offers inbound DIDs with multiple simultaneous calls, suitable for a call center or calling card application. Most others limit you to one, or a small few, inbound paths. They (Level 3, actually) also have the widest coverage for DIDs in the US. At the current level of service, LiveVoip is not going to get my business. If I can find anybody else to provide my inbound service, I'm very interested in talking to them. /edg ___ 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 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] Re: No ringback over IAX - LiveVoip
On Fri, 04 Mar 2005 11:35:55 -0700 Paul Fielding [EMAIL PROTECTED] wrote: Ok, time for me to ask my own newbie question. :) I've done some digging on ringback, and if I'm understanding it correctly, it's the ring tone that the caller hears when dialing another person. What exactly is it that people are finding now working with LiveVoip? Everyone says 'ringback isn't working', but nobody's really explained exactly what's happening. At least not that I've been able to find. I have a DID with them, and it works just fine. Dialing out works fine, when people call in it works fine. I'm interested in knowing what it is that isn't working, and if I can re-create it on my system... regards, Paul Setup your * box to not answer the call right away. Allow for say 5 seconds of ringing. Then call into it on one of your DID's. From the calling end all you will get is dead air. No ringing. At least this is the issue I am having.. ___ 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] Re: No ringback over IAX - LiveVoip
Ed Greenberg wrote: --On Friday, March 04, 2005 11:58 AM -0600 James Taylor [EMAIL PROTECTED] wrote: It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. There is clearly an issue between LiveVoip and Asterisk. The LiveVoip people claim that they have been ignored on the Asterisk List and they indeed blame Asterisk for everything from lost dtmf to other failures. That said, they are the only company I've found that offers inbound DIDs with multiple simultaneous calls, suitable for a call center or calling card application. Most others limit you to one, or a small few, inbound paths. They (Level 3, actually) also have the widest coverage for DIDs in the US. At the current level of service, LiveVoip is not going to get my business. If I can find anybody else to provide my inbound service, I'm very interested in talking to them. /edg ___ 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 I believe LiveVOIP is a reseller of Level 3. From what I understand, you need to buy millions of minutes to get decent pricing at Level 3 as they are a mega wholesaler... I may be wrong, but that's what I got out of it. Greg ___ 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] Re: No ringback over IAX - LiveVoip
On Fri, 04 Mar 2005 11:46:27 -0700 Paul Fielding [EMAIL PROTECTED] wrote: Hmmm. My server is currently set to let the line ring for 20 seconds, ringing several extensions internally. (I do not answer the line, it just rings the extensions). If I don't pick up after 20 seconds it then answers the line and sends to voicemail or to an auto-attendant, depending on the situation. Ringback seems to be working for me, I hear ringing on the calling end... *shrug*. Paul Ok,I have to retract my last statement and give an update. It has been a while since I had played with the DID I have from them. It is not an issue before the * box picks up. I set my incoming context to ring my VoIP phone for 20 seconds directly with using the IVR system and I had the ringing. But when I restored it to no background on hold music and issued a dial command of Dial(SIP/2001,15,r) instead of Dial(SIP/2001,15,m), after the IVR plays its intro, I got no ringing on the calling end. Just dead air from LiveVoIP. I then used this same test context by dialing in through a VP Connect account and after the initial greeting and moving to the Dial command, I got the ringing on the the calling end. Sorry for the incorrect info the first time, it had just been quite a while since I had played with the Live account. Robert - Original Message - From: Robert Webb [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Friday, March 04, 2005 11:42 AM Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip On Fri, 04 Mar 2005 11:35:55 -0700 Paul Fielding [EMAIL PROTECTED] wrote: Ok, time for me to ask my own newbie question. :) I've done some digging on ringback, and if I'm understanding it correctly, it's the ring tone that the caller hears when dialing another person. What exactly is it that people are finding now working with LiveVoip? Everyone says 'ringback isn't working', but nobody's really explained exactly what's happening. At least not that I've been able to find. I have a DID with them, and it works just fine. Dialing out works fine, when people call in it works fine. I'm interested in knowing what it is that isn't working, and if I can re-create it on my system... regards, Paul Setup your * box to not answer the call right away. Allow for say 5 seconds of ringing. Then call into it on one of your DID's. From the calling end all you will get is dead air. No ringing. At least this is the issue I am having.. ___ 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] Re: No ringback over IAX - LiveVoip
Yes, they do blame everyone else. There is another thread where I posted that I couldn't get my toll free number working. I waited days for support to get back to me, and I ended up emailing this list and then livevoip again. A LiveVoip representive blasted me on this list, stating that I was handled by their staff, and they don't moderate this list. Apparently, he didn't read my post, were I explicitly said I emailed them first, received no response, then emailed this list. Besides the ringback problem and the initial problems configuring the number, I am quite happy with the quality and reliability of the service. I have always found other providers have a slight lag, which I don't find with Livevoip. Anyways, I haven't found anyone that offers a toll free number that works in Canada for 1.29 cents a minute. If there is others, please let me know. Thanks, -Ryan On Fri, 4 Mar 2005 11:45:26 -0500, Mark Eissler [EMAIL PROTECTED] wrote: Hasn't anyone noticed that LiveVoip seems to happily blame just about everything on Asterisk? FWIW, I have experienced the same type of problem on a Sprint cell phone and also using a residential VOIP account with Broadvox. Both were able to correct the problem at THEIR end. Since no one else on this list seems to be complaining about the problem using provider's other than LV, I would suggest sacking them and getting DIDs from some other place. Seems like that is always the first thing they suggest too so they must not be that interested in your business. -mark On Mar 2, 2005, at 11:06 PM, Ryan Laginski wrote: Hi, I am experiencing the same problem as you. Ringback works great with the pstn or any other voip provider, but not with livevoip. I've just upgraded to 1.0.6 to see if that resolves the problem, but it has not. Please post back if you find a solution, I'll do the same. Thanks, -Ryan On Wed, 2 Feb 2005 13:25:29 -0500, Brian Dingman [EMAIL PROTECTED] wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Re: No ringback over IAX - LiveVoip
I wonder if you could share your configuration (sip.conf and extensions.conf) on handling incoming calls from VoipLive, since I'm trying to set it up also. Thanks a lot, Roman Zhovtulya -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Freitag, 4. März 2005 19:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip Ok, time for me to ask my own newbie question. :) I've done some digging on ringback, and if I'm understanding it correctly, it's the ring tone that the caller hears when dialing another person. What exactly is it that people are finding now working with LiveVoip? Everyone says 'ringback isn't working', but nobody's really explained exactly what's happening. At least not that I've been able to find. I have a DID with them, and it works just fine. Dialing out works fine, when people call in it works fine. I'm interested in knowing what it is that isn't working, and if I can re-create it on my system... regards, Paul - Original Message - From: Ed Greenberg [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 11:12 AM Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip --On Friday, March 04, 2005 11:58 AM -0600 James Taylor [EMAIL PROTECTED] wrote: It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. There is clearly an issue between LiveVoip and Asterisk. The LiveVoip people claim that they have been ignored on the Asterisk List and they indeed blame Asterisk for everything from lost dtmf to other failures. That said, they are the only company I've found that offers inbound DIDs with multiple simultaneous calls, suitable for a call center or calling card application. Most others limit you to one, or a small few, inbound paths. They (Level 3, actually) also have the widest coverage for DIDs in the US. At the current level of service, LiveVoip is not going to get my business. If I can find anybody else to provide my inbound service, I'm very interested in talking to them. /edg ___ 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 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 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] Re: No ringback over IAX - LiveVoip
- Original Message - From: Ryan Laginski [EMAIL PROTECTED] Anyways, I haven't found anyone that offers a toll free number that works in Canada for 1.29 cents a minute. If there is others, please let me know. You're LiveVoip toll free number costs 1.29 c/min from Canada? My toll free number through LiveVoip costs me 5 c/min when calling from Canada (1.2 c/min from US). Hmm wonder what I need to do to get that deal... :) Paul ___ 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] Re: No ringback over IAX - LiveVoip
Hi Robert, I tried yours and Steven's scenarios and you're absolutely right. I get ringback when the initial call takes place, but if I then try to do a transfer to another extension after the fact I do not hear ringback on the line. So I absolutely agree that there is a problem. What I'm not trying to understand is how Ringback works in this context. For lack of knowing better, my first thought would be that LiveVoip would be correct - that the problem is with asterisk, since I would have assumed that once LiveVoip has connected the call and asterisk has answered, all they're doing is providing audio in and out - wouldn't be Asterisk's responsibility to provide new ringtones to the calling party at this new transfer point? However, if everyone *does* get ringback when using other providers then it makes sense that there's something happening at LiveVoip's end. *shrug*. I'm interested in what the technicals are here regards, Paul - Original Message - From: Robert Webb [EMAIL PROTECTED] To: Paul Fielding [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 12:06 PM Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip On Fri, 04 Mar 2005 11:46:27 -0700 Paul Fielding [EMAIL PROTECTED] wrote: Hmmm. My server is currently set to let the line ring for 20 seconds, ringing several extensions internally. (I do not answer the line, it just rings the extensions). If I don't pick up after 20 seconds it then answers the line and sends to voicemail or to an auto-attendant, depending on the situation. Ringback seems to be working for me, I hear ringing on the calling end... *shrug*. Paul Ok,I have to retract my last statement and give an update. It has been a while since I had played with the DID I have from them. It is not an issue before the * box picks up. I set my incoming context to ring my VoIP phone for 20 seconds directly with using the IVR system and I had the ringing. But when I restored it to no background on hold music and issued a dial command of Dial(SIP/2001,15,r) instead of Dial(SIP/2001,15,m), after the IVR plays its intro, I got no ringing on the calling end. Just dead air from LiveVoIP. I then used this same test context by dialing in through a VP Connect account and after the initial greeting and moving to the Dial command, I got the ringing on the the calling end. Sorry for the incorrect info the first time, it had just been quite a while since I had played with the Live account. Robert - Original Message - From: Robert Webb [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Friday, March 04, 2005 11:42 AM Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip On Fri, 04 Mar 2005 11:35:55 -0700 Paul Fielding [EMAIL PROTECTED] wrote: Ok, time for me to ask my own newbie question. :) I've done some digging on ringback, and if I'm understanding it correctly, it's the ring tone that the caller hears when dialing another person. What exactly is it that people are finding now working with LiveVoip? Everyone says 'ringback isn't working', but nobody's really explained exactly what's happening. At least not that I've been able to find. I have a DID with them, and it works just fine. Dialing out works fine, when people call in it works fine. I'm interested in knowing what it is that isn't working, and if I can re-create it on my system... regards, Paul Setup your * box to not answer the call right away. Allow for say 5 seconds of ringing. Then call into it on one of your DID's. From the calling end all you will get is dead air. No ringing. At least this is the issue I am having.. ___ 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] Re: No ringback over IAX - LiveVoip
What I'm not trying to understand is how Ringback works in this context. err, I mean what I'm now trying to understand. Paul ___ 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] Re: No ringback over IAX - LiveVoip
Hi, I am experiencing the same problem as you. Ringback works great with the pstn or any other voip provider, but not with livevoip. I've just upgraded to 1.0.6 to see if that resolves the problem, but it has not. Please post back if you find a solution, I'll do the same. Thanks, -Ryan On Wed, 2 Feb 2005 13:25:29 -0500, Brian Dingman [EMAIL PROTECTED] wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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 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] Re: No ringback over IAX - LiveVoip
On Thu, 3 Feb 2005, Brian Dingman wrote: I took them up on their offer for a refund. IMHO they shouldn't offer * service at all. Even outgoing calls aren't handled properly. Lots of making progress - no answer results. Others have suggested iax.cc. However, they haven't repsonded to my email (over 2 days now) and I can't get through to them over the phone or IM. Not very promising. All I want is a toll free DID that works on * and isn't too expensive. Any suggestions for a provider? I don't even care if it can be ported away! Brian, I've had great luck with NuFone. I have a couple of 800 numbers from them. One for work, one for home and one for my in-laws out in New York. It's a no frills service, and it just works.. all the time.. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Re: No ringback over IAX - LiveVoip
I'm quite happy with iax.cc (Sixtel). I don't have DIDs with them but use them for outbound and have no complaints. Whenever I've contacted support I've received a reply the same day. Perhaps they prioritize their support email based on whether or not you have an account with them? They match up inbound support messages with your email address to check if its associated with an account. Anyhow, I might obtain DIDs from them sometime soon if Voicepulse doesn't get their act together and gets inbound DTMF working properly over IAX. It's been almost two months since I first reported the problem and they just reply that they have to perform several software upgrades over a 2-4 week period. Jeez. I can't recommend Voicepulse either right now as I have no intention of switching back to SIP (from IAX) for termination. The funny thing is that I have ended up using FWD the most because of their toll free gateway. I'm constantly amazed at the clarity of those calls. But then again I'm constantly amazed at the clarity of any of my calls through Asterisk vs. say my residential phone services via Vonage and Broadvox. -mark On Feb 3, 2005, at 10:38 PM, Brian Dingman wrote: I took them up on their offer for a refund. IMHO they shouldn't offer * service at all. Even outgoing calls aren't handled properly. Lots of making progress - no answer results. Others have suggested iax.cc. However, they haven't repsonded to my email (over 2 days now) and I can't get through to them over the phone or IM. Not very promising. All I want is a toll free DID that works on * and isn't too expensive. Any suggestions for a provider? I don't even care if it can be ported away! On Thu, 3 Feb 2005 10:12:02 -0500, Mark Eissler [EMAIL PROTECTED] wrote: Based on the support and management responses that have been posted to this list it doesn't sound to me (at least) like LiveVoip really wants business from * users anyhow. They blame a lot of problems on * and are quick to offer a refund. There are plenty of DID providers that are more asterisk-friendly. -mark On Feb 2, 2005, at 1:25 PM, Brian Dingman wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Re: No ringback over IAX - LiveVoip
I currently use LiveVoip for service (2 DIDs, 1 Toll Free 8ZZ and outgoing). Although it took longer to get installed than was expected (about a day and a half), I find their service to be quite acceptable. Could you clarify the no ringback condition? There was a situation at first in which I got fast busies (reorder) when calling the incoming services. I suspect that it took a while to provision the service with THEIR service providers (Lever 3, Quest, etc.) I reported the problem via email, and was pleased with the support I received. At least they ARE willing to quickly refund money (unlike some other providers I have read about on the list) and seem sincere about their desire to provide quality service. Norm Zimon Globex Telecom www.globextele.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Eissler Sent: Thursday, February 03, 2005 7:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Brian Dingman Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip Based on the support and management responses that have been posted to this list it doesn't sound to me (at least) like LiveVoip really wants business from * users anyhow. They blame a lot of problems on * and are quick to offer a refund. There are plenty of DID providers that are more asterisk-friendly. -mark On Feb 2, 2005, at 1:25 PM, Brian Dingman wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 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] Re: No ringback over IAX - LiveVoip
What we have been discussing with no ringback, is if you have a caller call in through your DID line and say dials an extension, then after using the dial command, the caller hears silence and no ringing tone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, February 04, 2005 1:58 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Re: No ringback over IAX - LiveVoip I currently use LiveVoip for service (2 DIDs, 1 Toll Free 8ZZ and outgoing). Although it took longer to get installed than was expected (about a day and a half), I find their service to be quite acceptable. Could you clarify the no ringback condition? There was a situation at first in which I got fast busies (reorder) when calling the incoming services. I suspect that it took a while to provision the service with THEIR service providers (Lever 3, Quest, etc.) I reported the problem via email, and was pleased with the support I received. At least they ARE willing to quickly refund money (unlike some other providers I have read about on the list) and seem sincere about their desire to provide quality service. Norm Zimon Globex Telecom www.globextele.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Eissler Sent: Thursday, February 03, 2005 7:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Brian Dingman Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip Based on the support and management responses that have been posted to this list it doesn't sound to me (at least) like LiveVoip really wants business from * users anyhow. They blame a lot of problems on * and are quick to offer a refund. There are plenty of DID providers that are more asterisk-friendly. -mark On Feb 2, 2005, at 1:25 PM, Brian Dingman wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 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 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] Re: No ringback over IAX - LiveVoip
Based on the support and management responses that have been posted to this list it doesn't sound to me (at least) like LiveVoip really wants business from * users anyhow. They blame a lot of problems on * and are quick to offer a refund. There are plenty of DID providers that are more asterisk-friendly. -mark On Feb 2, 2005, at 1:25 PM, Brian Dingman wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Re: No ringback over IAX - LiveVoip
I took them up on their offer for a refund. IMHO they shouldn't offer * service at all. Even outgoing calls aren't handled properly. Lots of making progress - no answer results. Others have suggested iax.cc. However, they haven't repsonded to my email (over 2 days now) and I can't get through to them over the phone or IM. Not very promising. All I want is a toll free DID that works on * and isn't too expensive. Any suggestions for a provider? I don't even care if it can be ported away! On Thu, 3 Feb 2005 10:12:02 -0500, Mark Eissler [EMAIL PROTECTED] wrote: Based on the support and management responses that have been posted to this list it doesn't sound to me (at least) like LiveVoip really wants business from * users anyhow. They blame a lot of problems on * and are quick to offer a refund. There are plenty of DID providers that are more asterisk-friendly. -mark On Feb 2, 2005, at 1:25 PM, Brian Dingman wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. ___ 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] Re: No ringback over IAX - LiveVoip
Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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] Re: No ringback over IAX - LiveVoip
I am having an issue right now where they cannot seem to get their switch configured correctly. When I call the number I get either a fast busy or a You're call cannot be completed as dialed message. I got a response back that when they call from their switch board, they get a woman's voice saying to please leave a message. There is no woman's voices on my * box that this line should go into. Plus, I am not seeing any incoming IAX connection from LV on my * box. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Dingman Sent: Wednesday, February 02, 2005 1:25 PM To: Steven Frazier; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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 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] Re: No ringback over IAX - LiveVoip
On Sat, 29 Jan 2005, Andrew Kohlsmith wrote: On January 29, 2005 11:29 pm, Brian Dingman wrote: This is driving me crazy. I have resorted to using the m option in the Dial command just so folks don't hang up. I can't believe nobody else is having this issue. Simple test: try it with another VOIP provider. Throw $5 at a nufone account, or an iax.cc account. See what happens. Hell you're already saying it's working with other providers, so what's your data showing you? Why do people insist on staying with VOIP providers who provide spotty performance and half-assed answers to technical support issues? Same reason people stick with Gentoo after a stage one installation. ;) I have a theory about Gentoo that explains the rabid nature of Gentoo fans. I believe that people that radically defend Gentoo and it's stage one installation process are people that have fought through the process and gotten a system to work. After spending 2 days working at it, the last thing they want to do is admit that they are a total idiot for wasting 48 hours of their life getting their system to a login prompt, so in a classic case of denial, they become raging defenders of the cause. If they convince themselves, and others that Gentoo is the best thing since sliced bread, they feel better about themselves. Now, with crappy VoIP providers it may be that they just do not want to let go of the dream. Or, they just want to recover the value of the money they have deposited with that company. ;) P.S. I have no experience with Livevoip or their service, so I have no idea if it is crappy or not. However, pretty any much VoIP service delivered over the public, non-QOS controlled Internet is going to have it's share of problems at some point in time. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Re: No ringback over IAX - LiveVoip
On January 30, 2005 09:31 am, Greg Boehnlein wrote: Same reason people stick with Gentoo after a stage one installation. ;) I have a theory about Gentoo that explains the rabid nature of Gentoo fans. I believe that people that radically defend Gentoo and it's stage one installation process are people that have fought through the process and gotten a system to work. After spending 2 days working at it, the last thing they want to do is admit that they are a total idiot for wasting 48 hours of their life getting their system to a login prompt, so in a classic case of denial, they become raging defenders of the cause. If they convince themselves, and others that Gentoo is the best thing since sliced bread, they feel better about themselves. Well if you're doing it for a learning experience that is one thing. I used LFS and scratchbox for those purposes. :-) Now, with crappy VoIP providers it may be that they just do not want to let go of the dream. Or, they just want to recover the value of the money they have deposited with that company. ;) Credit cards have a great feature where you can clawback any charge. Use it wisely. :-) P.S. I have no experience with Livevoip or their service, so I have no idea if it is crappy or not. However, pretty any much VoIP service delivered over the public, non-QOS controlled Internet is going to have it's share of problems at some point in time. Yes and no... Typically speaking, once you're at your upstream provider's router there are no bottlenecks. It's all in the last mile, in my experience. If your provider's oversubscribing too much then that is another issue entirely but typically if you're not on a consumer-grade connection there isn't a whole lot of trouble with QoS and the internet, barring the next worm. -A. ___ 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] Re: No ringback over IAX - LiveVoip
There is the little problem of having to switch numbers and then communicating to everyone that the number has changed. This also only seems to be a problem on inbound calls. On Sat, 29 Jan 2005 23:34:49 -0500, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On January 29, 2005 11:29 pm, Brian Dingman wrote: This is driving me crazy. I have resorted to using the m option in the Dial command just so folks don't hang up. I can't believe nobody else is having this issue. Simple test: try it with another VOIP provider. Throw $5 at a nufone account, or an iax.cc account. See what happens. Hell you're already saying it's working with other providers, so what's your data showing you? Why do people insist on staying with VOIP providers who provide spotty performance and half-assed answers to technical support issues? -A. ___ 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] Re: No ringback over IAX - LiveVoip
On January 30, 2005 12:18 pm, Brian Dingman wrote: There is the little problem of having to switch numbers and then communicating to everyone that the number has changed. This also only seems to be a problem on inbound calls. And why, praytell, did you go into production with DIDs from a provider that couldn't be ported and without adequate testing? I could be wrong and maybe it worked all along until this point, but in that case you should be able to revert to the last good config and continue. -A. ___ 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] Re: No ringback over IAX - LiveVoip
This is driving me crazy. I have resorted to using the m option in the
Dial command just so folks don't hang up. I can't believe nobody else
is having this issue.
Any ideas to work around this?
On Wed, 26 Jan 2005 12:11:42 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
Some more info. Using this exact call flow, ringback works for PSTN
callers over WIldcard, IAX Callers over VP Connect, but NOT IAX
callers over LiveVoip. Could this possibly be a bug with their new
patch?
On Wed, 26 Jan 2005 11:43:59 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
Here is the call flow:
[ivr-incoming]
exten = s,1,LookupCIDName
exten = s,2,DigitTimeout(2)
exten = s,3,ResponseTimeout(10)
exten = s,4,Wait(1)
exten = s,5,Background(custom/ivr-incoming)
exten = 1,1,Background(pls-wait-connect-call)
exten = 1,2,Dial(${RINGPHONENUMBERS},20,r)
exten = 1,3,Voicemail,u${VMBOX}
exten = 1,4,Hangup
Running * 1.0.5. The calling party hears the please wait while I
connect your call, but does not hear any ringing. I tried inserting
exten = 1,1,Ringing but that does not work either.
The same call flow from the pstn DOES generate ringback:
[fromPSTN]
exten = s,1,DigitTimeout(2)
exten = s,2,ResponseTimeout(10)
exten = s,3,Wait(1)
exten = s,4,Background(custom/ivr-greeting)
exten = 1,1,Background(pls-wait-connect-call)
exten = 1,2,Dial(${RINGPHONENUMBERS},15,r)
exten = 1,3,Voicemail,u${VMBOX}
exten = 1,4,Hangup
Any thoughts.
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Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
On January 29, 2005 11:29 pm, Brian Dingman wrote: This is driving me crazy. I have resorted to using the m option in the Dial command just so folks don't hang up. I can't believe nobody else is having this issue. Simple test: try it with another VOIP provider. Throw $5 at a nufone account, or an iax.cc account. See what happens. Hell you're already saying it's working with other providers, so what's your data showing you? Why do people insist on staying with VOIP providers who provide spotty performance and half-assed answers to technical support issues? -A. ___ 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
