Re: [Asterisk-Users] VoicePulse SIP
asterisk only supports IAX2, SIP and TEL, it will only use IAX2 and SIP entries however so it is used to route via the Net if it cannot find a route via the Net or the link isn't working it will go to the next priority in your dialplan and do whatever you want, it doesn't re-configure your dialplan or route preferences let's say it's a bypass to IAX and SIP providers as it will tell you the username and server where users may be reached directly!!! Marc At 23:50 22.05.2004, you wrote: Andres wrote: [EMAIL PROTECTED] wrote: Which providers give you a jitter buffer? In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure there are more. Clearpath gives jitter buffer as well. http://www.clearpath1.com/ 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 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] VoicePulse SIP
Hfailed to add the illustrative bit about my installationI DO have an X100p in my * box. I'm not using it for anything more than a timing source since I'm not happy with it as an FXO. I've just recently sarted playing with the Sipura SPA-3000 as an FXO. Michael On Sat, 22 May 2004 20:07:27 +0800, Lars Boegild Thomsen wrote: H - can anybody confirm this. I have generally had little luck with IAX in any case so I must admit I assumed (due to info from www.voip-info.org) that it was due to lack of timing device. I have actually not tried to do any trunking - just normal calls. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris A. Icide Sent: 22 May 2004 13:26 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoicePulse SIP Lars, I could be quite wrong, but I think you only need a 'timing' source if you want to use trunking over IAX. You can still use IAX without trunking if you don't have any sort of timing device. -Chris On 06:39 PM 5/21/2004, Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ 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 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 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 It is dangerous to be correct about matters when the established authories are wrong. - Voltaire ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] VoicePulse SIP
I use VoicePulse Connect, have done for about 6 months. I have no problems with audio quality relating to the fact that I use IAX2 as the connection protocol. I have had issues with QoS and codecs, but these were issues at my end. I've recently started trying iLBC instead of GSM. Michael On Sat, 22 May 2004 20:07:27 +0800, Lars Boegild Thomsen wrote: H - can anybody confirm this. I have generally had little luck with IAX in any case so I must admit I assumed (due to info from www.voip-info.org) that it was due to lack of timing device. I have actually not tried to do any trunking - just normal calls. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris A. Icide Sent: 22 May 2004 13:26 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoicePulse SIP Lars, I could be quite wrong, but I think you only need a 'timing' source if you want to use trunking over IAX. You can still use IAX without trunking if you don't have any sort of timing device. -Chris On 06:39 PM 5/21/2004, Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ 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 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 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 Plutocrats beware... ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] VoicePulse SIP
H - can anybody confirm this. I have generally had little luck with IAX in any case so I must admit I assumed (due to info from www.voip-info.org) that it was due to lack of timing device. I have actually not tried to do any trunking - just normal calls. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris A. Icide Sent: 22 May 2004 13:26 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoicePulse SIP Lars, I could be quite wrong, but I think you only need a 'timing' source if you want to use trunking over IAX. You can still use IAX without trunking if you don't have any sort of timing device. -Chris On 06:39 PM 5/21/2004, Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ 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 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 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] VoicePulse SIP
I'm using Coloco now, which so far is working well. Where companies like VoicePulse buy services from a patchwork of CLECs in order to cover their markets, Coloco is a CLEC. The upside is that you cut out the middleman. But if you need a number in an area they don't serve you'll need to find a different provider. Coloco serves latas 236 and 238 (NPAs 301,240,410,443,703), which works well for me since I'm in 238. If you need numbers local to DC and central Maryland give them a shout (coloco.com). I hear they're also working with some other CLECs to get numbers in other areas but I don't have any details on that. -brian David H Hickman wrote: Who do you use now? David Hickman TSG Computer Consulting - Auctions 314-865-4752 x2 On May 21, 2004, at 8:49 PM, Brian Cuthie wrote: SIP used to work fine with VoicePulse. But the funny thing is I could never detect any signs that they were doing call accounting. I could make IAX calls and see them show up in the CDR and the $$ deducted from my account balance. But when I made SIP calls they appeared, by all measures, to be free. I wrote to their support department several times about this and never received a response. But that was pretty much par for the course with those guys so I moved on to another provider. -brian Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ 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 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 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] VoicePulse SIP
Brian Cuthie wrote: I'm using Coloco now, which so far is working well. Where companies like VoicePulse buy services from a patchwork of CLECs in order to cover their markets, Coloco is a CLEC. The upside is that you cut out the middleman. But if you need a number in an area they don't serve you'll need to find a different provider. Coloco serves latas 236 and 238 (NPAs 301,240,410,443,703), which works well for me since I'm in 238. If you need numbers local to DC and central Maryland give them a shout (coloco.com). I hear they're also working with some other CLECs to get numbers in other areas but I don't have any details on that. -brian Is all above AFTER or BEFORE coloco is sent many emails asking please I would like to buy from your company? My experience with them is EXACTLY that!!! ___ 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] VoicePulse SIP
Welcome to Voicepulse and their lack of jitter buffer. This is the cause of your horrible sound. Will be just as bad with SIP. Which providers give you a jitter buffer? ___ 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] VoicePulse SIP
[EMAIL PROTECTED] wrote: Welcome to Voicepulse and their lack of jitter buffer. This is the cause of your horrible sound. Will be just as bad with SIP. Which providers give you a jitter buffer? In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure there are more. ___ 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 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] VoicePulse SIP
Lars Boegild Thomsen wrote: H - can anybody confirm this. I have generally had little luck with IAX in any case so I must admit I assumed (due to info from www.voip-info.org) that it was due to lack of timing device. I have actually not tried to do any trunking - just normal calls. That is correct. You only need it for IAX2 trunking. ___ 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] VoicePulse SIP
Andres wrote: [EMAIL PROTECTED] wrote: Which providers give you a jitter buffer? In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure there are more. Clearpath gives jitter buffer as well. http://www.clearpath1.com/ 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] VoicePulse SIP
Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ 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] VoicePulse SIP
SIP used to work fine with VoicePulse. But the funny thing is I could never detect any signs that they were doing call accounting. I could make IAX calls and see them show up in the CDR and the $$ deducted from my account balance. But when I made SIP calls they appeared, by all measures, to be free. I wrote to their support department several times about this and never received a response. But that was pretty much par for the course with those guys so I moved on to another provider. -brian Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ 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 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] VoicePulse SIP
Let me get this straight - you moved on to a different provider because the calls did NOT show up on your bill? :) :) :) Just kiddin' :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Cuthie Sent: 22 May 2004 09:50 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoicePulse SIP SIP used to work fine with VoicePulse. But the funny thing is I could never detect any signs that they were doing call accounting. I could make IAX calls and see them show up in the CDR and the $$ deducted from my account balance. But when I made SIP calls they appeared, by all measures, to be free. I wrote to their support department several times about this and never received a response. But that was pretty much par for the course with those guys so I moved on to another provider. -brian Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ 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 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 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] VoicePulse SIP
Who do you use now? x-tad-smallerDavid Hickman TSG Computer Consulting - Auctions 314-865-4752 x2 /x-tad-smaller On May 21, 2004, at 8:49 PM, Brian Cuthie wrote: SIP used to work fine with VoicePulse. But the funny thing is I could never detect any signs that they were doing call accounting. I could make IAX calls and see them show up in the CDR and the $$ deducted from my account balance. But when I made SIP calls they appeared, by all measures, to be free. I wrote to their support department several times about this and never received a response. But that was pretty much par for the course with those guys so I moved on to another provider. -brian Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ 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 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] VoicePulse SIP
Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Welcome to Voicepulse and their lack of jitter buffer. This is the cause of your horrible sound. Will be just as bad with SIP. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ 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 -- Andres Network Admin http://www.telesip.net Providing Wholesale Florida SIP/IAX2 Termination for US$0.01/minute ___ 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] VoicePulse SIP
Lars, I could be quite wrong, but I think you only need a 'timing' source if you want to use trunking over IAX. You can still use IAX without trunking if you don't have any sort of timing device. -Chris On 06:39 PM 5/21/2004, Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ 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 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
