[asterisk-users] Voicepulse down
Starting around 10:00 AM EST. All services from them whether I connect by IP or DNS (both east coast and west). Anyone else? Fred Posner f...@teamforrest.com Main: +1 (212) 937-7844 Direct: +1 (503) 914-0999 www.teamforrest.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicepulse down
Quoting Fred Posner f...@teamforrest.com: Starting around 10:00 AM EST. All services from them whether I connect by IP or DNS (both east coast and west). Anyone else? Yes, I'm experiancing the same problem. Their www.voicepulse.com and connect.voicepulse.com seem to be offline as well. --Shane ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicepulse down
Yeah, they finally updated via their twitter account... Seems a generated exploded. http://www.voiptechchat.com/voip/165/voip-carrier-voicepulse-suffers-outages-uses-twitter/ Fred Posner www.teamforrest.com On Dec 22, 2008, at 11:15 AM, Shane Young wrote: Quoting Fred Posner f...@teamforrest.com: Starting around 10:00 AM EST. All services from them whether I connect by IP or DNS (both east coast and west). Anyone else? Yes, I'm experiancing the same problem. Their www.voicepulse.com and connect.voicepulse.com seem to be offline as well. --Shane ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicepulse down
Fred Posner wrote: Yeah, they finally updated via their twitter account... Seems a generated exploded. http://www.voiptechchat.com/voip/165/voip-carrier-voicepulse-suffers-outages-uses-twitter/ Fred Posner www.teamforrest.com On Dec 22, 2008, at 11:15 AM, Shane Young wrote: Quoting Fred Posner f...@teamforrest.com: Starting around 10:00 AM EST. All services from them whether I connect by IP or DNS (both east coast and west). Anyone else? Yes, I'm experiancing the same problem. Their www.voicepulse.com and connect.voicepulse.com seem to be offline as well. --Shane What's truly interesting is that JerJer from NuFone reared his unappealing head on the linked page, and commented on single points of failure. In my experience, NuFone IS a single point of failure in and of itself. Does anyone still deal with those idiots? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoicePulse Time out?
Anyone else having timeouts to Voicepulse? Fred Posner [EMAIL PROTECTED] Tel: +1 (212) 937-7844 x501 Fax: +1 (954) 252-4187 www.teamforrest.com smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
I have the same issue with the ringing currently, so I force a ring. Stephen Bosch wrote: Wes Baehr wrote: I had a lot of problems with garbled IAX calls (even when calling into just the IVR). The problem was compacted when I would bridge an incoming IAX call to an outgoing SIP call, though that may be a fault of Asterisk. Since using SIP everything has been working perfectly. I never had any real problems with dropping calls (that weren’t on my end). However, I don’t use IAX anymore, so I am not aware of any current issues. This is interesting information -- I've had similar problems with IAX trunks on totally different carriers. Example: Callers do not hear the remote ringing, or only some of the rings, or don't hear the beep tone for voice mail. IAX is easier if you're behind a firewall :( -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
I have been a VP connect customer for a few years, mow traffic, outgoing only. I have had very good experiences and they are usually the lowest cost for a USA route, often less than .01/min retail. /r On 8/8/07, John Meksavan [EMAIL PROTECTED] wrote: Has anybody use Voicepulse Connect for Asterisk? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoicePulse Connect
Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
John, Voicepulse Connect has been great to me. I've been using it for over a year now, and do not have any major complaints, except that there are no printable receipts for credit card transactions. SIP is also the preferable protocol, as IAX seems to have some issues. Customer service is usually pretty good, and there have been very few (although a couple) problems with service outages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 12:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoicePulse Connect Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
Wes, What kind of service outages did you experienced? This would use for my office and I cannot afford for any dropped calls or poor audio quality, when talking to customers. -John From: Wes Baehr [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Date: Wed, 8 Aug 2007 12:55:29 -0400 John, Voicepulse Connect has been great to me. I've been using it for over a year now, and do not have any major complaints, except that there are no printable receipts for credit card transactions. SIP is also the preferable protocol, as IAX seems to have some issues. Customer service is usually pretty good, and there have been very few (although a couple) problems with service outages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 12:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoicePulse Connect Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Find a local pizza place, movie theater, and more .then map the best route! http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theatercp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=950607encType=1FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
On Wed, 2007-08-08 at 17:08 +, John Meksavan wrote: Wes, What kind of service outages did you experienced? This would use for my office and I cannot afford for any dropped calls or poor audio quality, when talking to customers. My experience with Voicepulse has been good and quality is usually very good. Most of the time when calls get distorted the problems can be traced to my ISP. Unfortunately you will never be able to get 100% reliability when using the Internet. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
If you cannot afford any dropped calls or poor audio quality, you need a PRI or POTS connection. It doesn't matter how great the carrier is, the Internet is an unreliable medium. 2-3 times VoicePulse has had issues with incomings calls ringing busy. Once incoming calls were all garbled on my end, although the customer could hear me fine. Generally, the outbound service is reliable. However, you should have a backup carrier anyway. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 1:09 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Wes, What kind of service outages did you experienced? This would use for my office and I cannot afford for any dropped calls or poor audio quality, when talking to customers. -John From: Wes Baehr [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Date: Wed, 8 Aug 2007 12:55:29 -0400 John, Voicepulse Connect has been great to me. I've been using it for over a year now, and do not have any major complaints, except that there are no printable receipts for credit card transactions. SIP is also the preferable protocol, as IAX seems to have some issues. Customer service is usually pretty good, and there have been very few (although a couple) problems with service outages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 12:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoicePulse Connect Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Find a local pizza place, movie theater, and more.then map the best route! http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theater cp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=95060 7encType=1FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
Wes, I'm working through some issues with IAX and Voicepulse right now. It was regarding dropped inbound calls. I was able to put my server into a different location though, and so far the issues have disappeared so hopefully it was a network problem somewhere between us.Just curious what problems you encountered as I would prefer to use IAX if possible. John, I've tried a few services, and Voicepulse was the clear winner for me. I still have two other services in my dialplan for failover, but Voicepulse will remain the primary for now. The voice quality has been very good, and their technical support has been absolutely fantastic for a no-charge service. Wes Baehr wrote: If you cannot afford any dropped calls or poor audio quality, you need a PRI or POTS connection. It doesn't matter how great the carrier is, the Internet is an unreliable medium. 2-3 times VoicePulse has had issues with incomings calls ringing busy. Once incoming calls were all garbled on my end, although the customer could hear me fine. Generally, the outbound service is reliable. However, you should have a backup carrier anyway. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 1:09 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Wes, What kind of service outages did you experienced? This would use for my office and I cannot afford for any dropped calls or poor audio quality, when talking to customers. -John From: Wes Baehr [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Date: Wed, 8 Aug 2007 12:55:29 -0400 John, Voicepulse Connect has been great to me. I've been using it for over a year now, and do not have any major complaints, except that there are no printable receipts for credit card transactions. SIP is also the preferable protocol, as IAX seems to have some issues. Customer service is usually pretty good, and there have been very few (although a couple) problems with service outages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 12:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoicePulse Connect Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Find a local pizza place, movie theater, and more.then map the best route! http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theater cp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=95060 7encType=1FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
I had a lot of problems with garbled IAX calls (even when calling into just the IVR). The problem was compacted when I would bridge an incoming IAX call to an outgoing SIP call, though that may be a fault of Asterisk. Since using SIP everything has been working perfectly. I never had any real problems with dropping calls (that weren't on my end). However, I don't use IAX anymore, so I am not aware of any current issues. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Robinson Sent: Wednesday, August 08, 2007 4:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoicePulse Connect Wes, I'm working through some issues with IAX and Voicepulse right now. It was regarding dropped inbound calls. I was able to put my server into a different location though, and so far the issues have disappeared so hopefully it was a network problem somewhere between us.Just curious what problems you encountered as I would prefer to use IAX if possible. John, I've tried a few services, and Voicepulse was the clear winner for me. I still have two other services in my dialplan for failover, but Voicepulse will remain the primary for now. The voice quality has been very good, and their technical support has been absolutely fantastic for a no-charge service. Wes Baehr wrote: If you cannot afford any dropped calls or poor audio quality, you need a PRI or POTS connection. It doesn't matter how great the carrier is, the Internet is an unreliable medium. 2-3 times VoicePulse has had issues with incomings calls ringing busy. Once incoming calls were all garbled on my end, although the customer could hear me fine. Generally, the outbound service is reliable. However, you should have a backup carrier anyway. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 1:09 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Wes, What kind of service outages did you experienced? This would use for my office and I cannot afford for any dropped calls or poor audio quality, when talking to customers. -John From: Wes Baehr mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com asterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' mailto:asterisk-users@lists.digium.com asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Date: Wed, 8 Aug 2007 12:55:29 -0400 John, Voicepulse Connect has been great to me. I've been using it for over a year now, and do not have any major complaints, except that there are no printable receipts for credit card transactions. SIP is also the preferable protocol, as IAX seems to have some issues. Customer service is usually pretty good, and there have been very few (although a couple) problems with service outages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 12:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoicePulse Connect Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Find a local pizza place, movie theater, and more.then map the best route! http://maps.live.com/default.aspx?v=2 http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theate r ss=yp.bars~yp.pizza~yp.movie%20theater cp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=95060 7encType=1FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
Re: [asterisk-users] VoicePulse Connect
Wes Baehr wrote: I had a lot of problems with garbled IAX calls (even when calling into just the IVR). The problem was compacted when I would bridge an incoming IAX call to an outgoing SIP call, though that may be a fault of Asterisk. Since using SIP everything has been working perfectly. I never had any real problems with dropping calls (that weren’t on my end). However, I don’t use IAX anymore, so I am not aware of any current issues. This is interesting information -- I've had similar problems with IAX trunks on totally different carriers. Example: Callers do not hear the remote ringing, or only some of the rings, or don't hear the beep tone for voice mail. IAX is easier if you're behind a firewall :( -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
I've experienced the beginning silence sensation with IAX (no firewalls). As far as audio dropping out, that sounds like firewall or intertube issues. Along with the other problems, I gave up on IAX and went pure SIP (except for that pesky iaxmodem -- however, that works fine when receiving faxes from a SIP trunk). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Wednesday, August 08, 2007 6:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoicePulse Connect Wes Baehr wrote: I had a lot of problems with garbled IAX calls (even when calling into just the IVR). The problem was compacted when I would bridge an incoming IAX call to an outgoing SIP call, though that may be a fault of Asterisk. Since using SIP everything has been working perfectly. I never had any real problems with dropping calls (that weren't on my end). However, I don't use IAX anymore, so I am not aware of any current issues. This is interesting information -- I've had similar problems with IAX trunks on totally different carriers. Example: Callers do not hear the remote ringing, or only some of the rings, or don't hear the beep tone for voice mail. IAX is easier if you're behind a firewall :( -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
On 8/8/07, John Meksavan [EMAIL PROTECTED] wrote: Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. I've had too many DTMF problems and quality of service issues with VoicePulse Connect. The DTMF was an issue from only certain mobile carriers, they could not fix it. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect 4 Channel Limit?
So, What´s your recommendation for a production environment? I waslooking for good prices, good voice quality for USA Origination and I´d like to hear about good experiences PSTN. Just can't beat the quality :-) Wait, you said good prices. Sorry. ___ --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] VoicePulse Connect 4 Channel Limit?
Good prices means (exactly) reasonable prices. I´m a newbie, so I´m asking for good experiences... Thanks in advance... R.R. Libera Lacy Moore - Aspendora escribió: So, What´s your recommendation for a production environment? I was looking for good prices, good voice quality for USA Origination and I´d like to hear about good experiences PSTN. Just can't beat the quality :-) Wait, you said good prices. Sorry. ___ --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] VoicePulse Connect 4 Channel Limit?
I would worry about using Voicepulse as your primary provider, even if they didn't impose their draconian policies. You could have 20 numbers paying $220/month in your account and you still get only four calls,. However if you were to open 20 voicepulse connect accounts and put one number on each, you would still pay the same $220/month however you could get up to EIGHTY -- thats 20 times more! -- for the SAME EXACT PRICES Also VoicePulse DOES NOT use Tier 1 providers. We notice every week or so during peak hours a very bad degrigation of the voice quality. If you have an IVR and call it from a landline, it will sound like crap. It's the quality of service you would expect from a free provider. Aggrivated to this, when you contact them they try to blame YOU for their issues. They told me I HAD to run PingPlotter (a WINDOWS program, besides the fact this is VoicePulse Connect for Asterisk and Asterisk is software for Linux) which was not possible on a co-located machine. Also we ported a bunch of phone numbers and the DTMF does not work. If you dial 5551212 VoicePulse might recognise and pass to us 55112 and again instead of trying to troubleshoot the issue (from the SAME phone it always produced CONSISTANT behavior -- the ported number does not accet DTMF correctly, assigned # work!) they blame us and the phones we use. I went as far as going to Sprint PCS store and EVERY CDMA phone in the store would produce the same result! In the end, don't bother with VoicePulse. The quality of the service and the support and just the treatment you get is not worth the price. For $11/month per number and their draconian channels and also billing policy (I wont even get into that) I expect a PREMIUM service and they deliver something about par for a free service. Here's some typical behavior from their servers: ug 24 14:55:48 NOTICE[13944]: chan_iax2.c:7813 iax2_poke_noanswer: Peer 'voicepulse01' is now UNREACHABLE! Time: 71 Aug 24 14:55:59 NOTICE[13944]: chan_iax2.c:7145 socket_read: Peer 'voicepulse01' is now REACHABLE! Time: 1059 Aug 24 15:11:05 NOTICE[13944]: chan_iax2.c:7813 iax2_poke_noanswer: Peer 'voicepulse01' is now UNREACHABLE! Time: 40 Aug 24 15:11:15 NOTICE[13944]: chan_iax2.c:7145 socket_read: Peer 'voicepulse01' is now REACHABLE! Time: 49 Aug 21 15:33:08 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 39 Aug 21 15:33:18 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 43 Aug 22 13:40:41 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 45 Aug 22 13:40:52 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 1064 Aug 22 16:57:15 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 41 Aug 22 16:57:25 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 43 Aug 23 11:02:36 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 11:03:49 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 37 Aug 23 11:08:19 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 39 Aug 23 11:08:26 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 39 Aug 23 11:16:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 46 Aug 23 12:10:25 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 1246 Aug 23 14:01:37 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 45 Aug 23 14:04:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 258 Aug 23 15:28:36 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 44 Aug 23 15:28:37 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 39 Aug 23 15:31:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 41 Aug 23 15:31:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 56 Aug 23 15:40:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 40 Aug 23 15:43:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 15:49:42 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 42 Aug 23 15:57:22 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 40 Aug 23 16:37:49 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 44 Aug 23 16:38:00 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 48 Aug 23 17:31:07 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 17:32:17 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 42 Aug 23 17:34:21 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 42 Aug 23 17:34:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 40 Aug 23 17:36:01 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 39 Aug 23 17:36:11 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01'
Re: [asterisk-users] VoicePulse Connect 4 Channel Limit?
So, What´s your recommendation for a production environment? I was looking for good prices, good voice quality for USA Origination and I´d like to hear about good experiences thanks in advance.. R.R. Libera Andrew Joakimsen escribió: I would worry about using Voicepulse as your primary provider, even if they didn't impose their draconian policies. You could have 20 numbers paying $220/month in your account and you still get only four calls,. However if you were to open 20 voicepulse connect accounts and put one number on each, you would still pay the same $220/month however you could get up to EIGHTY -- thats 20 times more! -- for the SAME EXACT PRICES Also VoicePulse DOES NOT use Tier 1 providers. We notice every week or so during peak hours a very bad degrigation of the voice quality. If you have an IVR and call it from a landline, it will sound like crap. It's the quality of service you would expect from a free provider. Aggrivated to this, when you contact them they try to blame YOU for their issues. They told me I HAD to run PingPlotter (a WINDOWS program, besides the fact this is VoicePulse Connect for Asterisk and Asterisk is software for Linux) which was not possible on a co-located machine. Also we ported a bunch of phone numbers and the DTMF does not work. If you dial 5551212 VoicePulse might recognise and pass to us 55112 and again instead of trying to troubleshoot the issue (from the SAME phone it always produced CONSISTANT behavior -- the ported number does not accet DTMF correctly, assigned # work!) they blame us and the phones we use. I went as far as going to Sprint PCS store and EVERY CDMA phone in the store would produce the same result! In the end, don't bother with VoicePulse. The quality of the service and the support and just the treatment you get is not worth the price. For $11/month per number and their draconian channels and also billing policy (I wont even get into that) I expect a PREMIUM service and they deliver something about par for a free service. Here's some typical behavior from their servers: ug 24 14:55:48 NOTICE[13944]: chan_iax2.c:7813 iax2_poke_noanswer: Peer 'voicepulse01' is now UNREACHABLE! Time: 71 Aug 24 14:55:59 NOTICE[13944]: chan_iax2.c:7145 socket_read: Peer 'voicepulse01' is now REACHABLE! Time: 1059 Aug 24 15:11:05 NOTICE[13944]: chan_iax2.c:7813 iax2_poke_noanswer: Peer 'voicepulse01' is now UNREACHABLE! Time: 40 Aug 24 15:11:15 NOTICE[13944]: chan_iax2.c:7145 socket_read: Peer 'voicepulse01' is now REACHABLE! Time: 49 Aug 21 15:33:08 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 39 Aug 21 15:33:18 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 43 Aug 22 13:40:41 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 45 Aug 22 13:40:52 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 1064 Aug 22 16:57:15 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 41 Aug 22 16:57:25 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 43 Aug 23 11:02:36 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 11:03:49 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 37 Aug 23 11:08:19 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 39 Aug 23 11:08:26 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 39 Aug 23 11:16:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 46 Aug 23 12:10:25 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 1246 Aug 23 14:01:37 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 45 Aug 23 14:04:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 258 Aug 23 15:28:36 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 44 Aug 23 15:28:37 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 39 Aug 23 15:31:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 41 Aug 23 15:31:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 56 Aug 23 15:40:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 40 Aug 23 15:43:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 15:49:42 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 42 Aug 23 15:57:22 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 40 Aug 23 16:37:49 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 44 Aug 23 16:38:00 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 48 Aug 23 17:31:07 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 17:32:17 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 42 Aug 23 17:34:21 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 42
Re: [asterisk-users] VoicePulse Connect 4 Channel Limit?
voipstreet allows 20 concurrent calls. On 10/16/06, Nate Kapi [EMAIL PROTECTED] wrote: Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. ___ --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] VoicePulse Connect 4 Channel Limit?
I use myphonecompany.com. They have DID's for $5.00 a month and they 'let you' use 2 channels for per did (you can use more but they dont like it if you abuse it). I had a client that needed 4 concurent channels so they told him to just purchase 2 did's. So if you need 8 concurent incoming channels it will cost you a total of $20.00 for inbound services :) - Original Message - From: Nate Kapi To: asterisk-users@lists.digium.com Sent: Monday, October 16, 2006 7:13 AM Subject: [asterisk-users] VoicePulse Connect 4 Channel Limit? Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound and outbound channels. If you wanted to be able to have 8 concurrent channels then this could get costly. Too costly in my opinion. I meanthat seems like a LOT to me, when you can go with other providers who don't limit you to 4 channels, like Voxee, NuFone or SixTel, for around the same price. I can understand the channel restrictions for inbound calls, but not for outbound calls. VoicePulse, I know you read these lists! You should be able to provide us VoicePulse Connect users with more than 4 concurrent channels for free! ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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
[asterisk-users] VoicePulse Connect 4 Channel Limit?
Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound and outbound channels. If you wanted to be able to have 8 concurrent channels then this could get costly. Too costly in my opinion. I meanthat seems like a LOT to me, when you can go with other providers who don't limit you to 4 channels, like Voxee, NuFone or SixTel, for around the same price. I can understand the channel restrictions for inbound calls, but not for outbound calls. VoicePulse, I know you read these lists! You should be able to provide us VoicePulse Connect users with more than 4 concurrent channels for free! ___ --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] Voicepulse Connect! and SIP settings
Is anyone out there using Voicepulse Connect! Service under SIP with Asterisk successfully? And if so, can you please post the sip.conf settings. I know that Voicepulse recommends for the service to be used in IAX2, but it would be nice to use as an alternate SIP when they are having problems with their IAX2 servers. Thanks in advance Juan. ___ --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] Voicepulse not accepting new customers. (FCC E911)
Hello All, It seems that voicepulse is not taking any new orders on the standard service plans (though vp connect seems unaffected) due to the fcc rulings. We'll see what happens, anyone having similar problems with other services as of today? Greg ___ --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] Voicepulse not accepting new customers. (FCC E911)
On Tue, November 29, 2005 18:36, [EMAIL PROTECTED] said: Hello All, It seems that voicepulse is not taking any new orders on the standard service plans (though vp connect seems unaffected) due to the fcc rulings. We'll see what happens, anyone having similar problems with other services as of today? Greg WHat fcc rulings? What did I miss? :-o -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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] Voicepulse not accepting new customers. (FCC E911)
On Tue, Nov 29, 2005 at 06:52:21PM +0100, Francesco Peeters wrote: On Tue, November 29, 2005 18:36, [EMAIL PROTECTED] said: Hello All, It seems that voicepulse is not taking any new orders on the standard service plans (though vp connect seems unaffected) due to the fcc rulings. We'll see what happens, anyone having similar problems with other services as of today? Greg WHat fcc rulings? What did I miss? :-o -- Francesco Peeters VOIP service providers are now required to provide E911 service to their customers or they may not sign up additional customers. This is in the US only. One provider has about 95% of their customers set up for E911 service. I know that CNN had a couple of articles on it on their website in the Tech section. The big stink was the FCC gave everyone 120 days to implement. Next stop will probably be US District Court in Washington, D.C. Patrick ___ --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] Voicepulse not accepting new customers. (FCC E911)
Patrick May wrote: On Tue, Nov 29, 2005 at 06:52:21PM +0100, Francesco Peeters wrote: On Tue, November 29, 2005 18:36, [EMAIL PROTECTED] said: Hello All, It seems that voicepulse is not taking any new orders on the standard service plans (though vp connect seems unaffected) due to the fcc rulings. We'll see what happens, anyone having similar problems with other services as of today? Greg WHat fcc rulings? What did I miss? :-o -- Francesco Peeters VOIP service providers are now required to provide E911 service to their customers or they may not sign up additional customers. This is in the US only. One provider has about 95% of their customers set up for E911 service. I know that CNN had a couple of articles on it on their website in the Tech section. The big stink was the FCC gave everyone 120 days to implement. Next stop will probably be US District Court in Washington, D.C. Patrick I checked the voicepulse and broadvoice websites. It looks to me like the online signup is still there for both. I didn't go as far as actually entering payment info but I would think if they were disabling it they would do that at step 1. ___ --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] Voicepulse Open Access status?
I have not had my 2 voicepulse open access numbers work for about 3 weeks now. I have even tried to make them work on a new server build using asterisk 1.2 I have several SIP and IAX DID's working fine. They worked fine on asterisk 1.0.x and easily were moved to the 1.2 setup. sip show registry indicates they are registered oaky but calls to both numbers go to voicepulse voice mail. If I setup hunt and fileters at the voicepulse web portal, that seems to work. For example, I can make the numbers ring my cell phone instead of going to voice mail. The primary number on the account works with the SPA-2000 ATA fine. I just used it to call the vonage number that comes into the asterisk system and I am using echo test as I type this. If anyone here has working voicepulse open access numbers could you please post sample lines from sip.conf and extensions.conf? If anyone here has been experiencing the same type of extended outage, I would like to hear about it because I am going to ask them to waive charges for at least one month of these softphone numbers. I was very tempted to put Please help!!! in the subject line today. If I don't get any replies I guess that means voicepulse sucks and I should cancel these numbers :) ___ --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] Voicepulse Open Access status?
Debug info and posting your .confs will help to get replys. -Original Message- From: Paul [mailto:[EMAIL PROTECTED] Sent: Sunday, November 27, 2005 9:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Voicepulse Open Access status? I have not had my 2 voicepulse open access numbers work for about 3 weeks now. I have even tried to make them work on a new server build using asterisk 1.2 I have several SIP and IAX DID's working fine. They worked fine on asterisk 1.0.x and easily were moved to the 1.2 setup. sip show registry indicates they are registered oaky but calls to both numbers go to voicepulse voice mail. If I setup hunt and fileters at the voicepulse web portal, that seems to work. For example, I can make the numbers ring my cell phone instead of going to voice mail. The primary number on the account works with the SPA-2000 ATA fine. I just used it to call the vonage number that comes into the asterisk system and I am using echo test as I type this. If anyone here has working voicepulse open access numbers could you please post sample lines from sip.conf and extensions.conf? If anyone here has been experiencing the same type of extended outage, I would like to hear about it because I am going to ask them to waive charges for at least one month of these softphone numbers. I was very tempted to put Please help!!! in the subject line today. If I don't get any replies I guess that means voicepulse sucks and I should cancel these numbers :) ___ --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] Voicepulse Open Access status?
I thought it might make more sense to start with an example config from someone who currently has incoming calls working. As I mentioned already sip show registry lists the 2 voicepulse numbers as registered. I have a console open with -rv and get no messages when I dial the numbers. In the past I have always been able to get some diagnostic info on the console when registered if something like context or codecs was amiss. Steve Totaro wrote: Debug info and posting your .confs will help to get replys. -Original Message- From: Paul [mailto:[EMAIL PROTECTED] Sent: Sunday, November 27, 2005 9:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Voicepulse Open Access status? I have not had my 2 voicepulse open access numbers work for about 3 weeks now. I have even tried to make them work on a new server build using asterisk 1.2 I have several SIP and IAX DID's working fine. They worked fine on asterisk 1.0.x and easily were moved to the 1.2 setup. sip show registry indicates they are registered oaky but calls to both numbers go to voicepulse voice mail. If I setup hunt and fileters at the voicepulse web portal, that seems to work. For example, I can make the numbers ring my cell phone instead of going to voice mail. The primary number on the account works with the SPA-2000 ATA fine. I just used it to call the vonage number that comes into the asterisk system and I am using echo test as I type this. If anyone here has working voicepulse open access numbers could you please post sample lines from sip.conf and extensions.conf? If anyone here has been experiencing the same type of extended outage, I would like to hear about it because I am going to ask them to waive charges for at least one month of these softphone numbers. I was very tempted to put Please help!!! in the subject line today. If I don't get any replies I guess that means voicepulse sucks and I should cancel these numbers :) ___ --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] Voicepulse Open Access status?
I thought it might make more sense to start with an example config from someone who currently has incoming calls working. As I mentioned already sip show registry lists the 2 voicepulse numbers as registered. I have a console open with -rv and get no messages when I dial the numbers. In the past I have always been able to get some diagnostic info on the console when registered if something like context or codecs was amiss. At the CLI, type 'sip debug' and call the numbers again. There should be something in the debug messages that point to the problem. Post the results. ___ --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] Voicepulse Open Access status?
I thought it might make more sense to start with an example config from someone who currently has incoming calls working. As I mentioned already sip show registry lists the 2 voicepulse numbers as registered. I have a console open with -rv and get no messages when I dial the numbers. In the past I have always been able to get some diagnostic info on the console when registered if something like context or codecs was amiss. At the CLI, type 'sip debug' and call the numbers again. There should be something in the debug messages that point to the problem. Post the results. IAX debug as well. ___ --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] Voicepulse Open Access status?
Steve Totaro wrote: I thought it might make more sense to start with an example config from someone who currently has incoming calls working. As I mentioned already sip show registry lists the 2 voicepulse numbers as registered. I have a console open with -rv and get no messages when I dial the numbers. In the past I have always been able to get some diagnostic info on the console when registered if something like context or codecs was amiss. At the CLI, type 'sip debug' and call the numbers again. There should be something in the debug messages that point to the problem. Post the results. IAX debug as well. This is voicepulse retail open access softphone - SIP only. I appreciate the replies but I don't see any from people reporting they are able to use voicepulse softphone accounts. Interesting. Anyway, I saw the following type thing: Looking for s0022 in default where s0022 is the username. So I added extensions that match and incoming now works. Looks to me like they have changed something at voicepulse with the usual policy of not notifying subscribers. ___ --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] Voicepulse Open Access problems
Paul wrote: snacktime wrote: On 11/13/05, *Paul* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have 2 voicepulse open access numbers coming in over SIP. I use them for some testing and at other times I just comment out the register lines and let them go to the voicepulse mailboxes. I went to use them yesterday and they are not working. Calls go to the voicemail and if I enable their unavailability forwarding that works. Anyone else having this problem? I didn't change anything on my end. It just stopped working. Since then I have tried a few things but nothing helped so I reverted to the config that worked once upon a time. One of their gateways took a dive about a week or so ago. Look at the gateways they have listed and use the second one, that one works. Nice of them to send us out a notice though after being down for that long. I've talked to them on the phone and they were easy to get ahold of, but they don't seem to pay much attention to their website or to notifying customers of things we should know. Quality has always been pretty good though. You know one provider that has always been really proactive with this kind of stuff is Teliax. They consistantly send me email messages about any changes, and it's a nice way of letting customers know that someone is actually there. Just the other day I got a notice about an old gateway they were phasing out. It reminded me to check all my setups and sure enough I had one with the old gateway still in my system. Chris Thanks. I can't seem to find anything listing additional gateways. I am using retail SIP. I suppose I can sniff traffic on the SPA-2000 to see what addresses/ports it is talking with. I ordered a vonage softphone and had it working quick. I built 1.2 rc2 for debian sarge and installed it on another box. I migrated the vonage did to it, called and did echo test. Experimenting a bit I found only 2 lines are needed if you only need incoming for a vonage softphone: in sip.conf register=1207433:[EMAIL PROTECTED]:5061/1207433 in extensions.conf exten = 1207433,1,Goto(default,s,1) Nothing working for vp. In my trouble ticket I added: I search the knowledge base and only find config samples for IAX with the connect product. I need SIP for open access examples. ___ --Bandwidth and Colocation sponsored by Easynews.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] Voicepulse Open Access problems
I have 2 voicepulse open access numbers coming in over SIP. I use them for some testing and at other times I just comment out the register lines and let them go to the voicepulse mailboxes. I went to use them yesterday and they are not working. Calls go to the voicemail and if I enable their unavailability forwarding that works. Anyone else having this problem? I didn't change anything on my end. It just stopped working. Since then I have tried a few things but nothing helped so I reverted to the config that worked once upon a time. ___ --Bandwidth and Colocation sponsored by Easynews.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] Voicepulse Open Access problems
On 11/13/05, Paul [EMAIL PROTECTED] wrote: I have 2 voicepulse open access numbers coming in over SIP. I use themfor some testing and at other times I just comment out the registerlines and let them go to the voicepulse mailboxes.I went to use them yesterday and they are not working. Calls go to the voicemail and if I enable their unavailability forwarding that works.Anyone else having this problem? I didn't change anything on my end. Itjust stopped working. Since then I have tried a few things but nothing helped so I reverted to the config that worked once upon a time. One of their gateways took a dive about a week or so ago. Look at the gateways they have listed and use the second one, that one works. Nice of them to send us out a notice though after being down for that long. I've talked to them on the phone and they were easy to get ahold of, but they don't seem to pay much attention to their website or to notifying customers of things we should know. Quality has always been pretty good though. You know one provider that has always been really proactive with this kind of stuff is Teliax. They consistantly send me email messages about any changes, and it's a nice way of letting customers know that someone is actually there. Just the other day I got a notice about an old gateway they were phasing out. It reminded me to check all my setups and sure enough I had one with the old gateway still in my system. Chris ___ --Bandwidth and Colocation sponsored by Easynews.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] Voicepulse Open Access problems
snacktime wrote: On 11/13/05, *Paul* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have 2 voicepulse open access numbers coming in over SIP. I use them for some testing and at other times I just comment out the register lines and let them go to the voicepulse mailboxes. I went to use them yesterday and they are not working. Calls go to the voicemail and if I enable their unavailability forwarding that works. Anyone else having this problem? I didn't change anything on my end. It just stopped working. Since then I have tried a few things but nothing helped so I reverted to the config that worked once upon a time. One of their gateways took a dive about a week or so ago. Look at the gateways they have listed and use the second one, that one works. Nice of them to send us out a notice though after being down for that long. I've talked to them on the phone and they were easy to get ahold of, but they don't seem to pay much attention to their website or to notifying customers of things we should know. Quality has always been pretty good though. You know one provider that has always been really proactive with this kind of stuff is Teliax. They consistantly send me email messages about any changes, and it's a nice way of letting customers know that someone is actually there. Just the other day I got a notice about an old gateway they were phasing out. It reminded me to check all my setups and sure enough I had one with the old gateway still in my system. Chris Thanks. I can't seem to find anything listing additional gateways. I am using retail SIP. I suppose I can sniff traffic on the SPA-2000 to see what addresses/ports it is talking with. ___ --Bandwidth and Colocation sponsored by Easynews.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] VoicePulse DTMF Problems Anyone?
I too have been having inbound dtmf problems with VP Connect using iax2 for inbound. When I switched to sip, and added the relaxdtmf=yes, all 10 inbound test calls I did seemed to work fine for dtmf. I'm going to leave my config set up to use sip for inbound VP Connect calls for a while and see how if functions. thanks for the relaxdtmf tip Umair. ___ 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] VoicePulse Connect down Sunday evening?
It appears that incoming calls (IAX) through voicepulse are being rejected... anyone else experiencing this? -Trent Tuggle ___ 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] VoicePulse DTMF Problems Anyone?
http://bugs.digium.com/view.php?id=4631 FYI I am experiencing the same, but due to lack of cooperation from ITSP am not able to proceed with debugging it. Feel free to pursue... regards, mark On 8/8/05, Justin Richards [EMAIL PROTECTED] wrote: I too have been having inbound dtmf problems with VP Connect using iax2 for inbound. When I switched to sip, and added the relaxdtmf=yes, all 10 inbound test calls I did seemed to work fine for dtmf. I'm going to leave my config set up to use sip for inbound VP Connect calls for a while and see how if functions. thanks for the relaxdtmf tip Umair. ___ 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 -- regards, Mark P. Edwards FWD: 667917 ___ 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] Voicepulse down?
no DNS resolution to begin with. Anyone heard anything about this? jlm --- John L Magee [EMAIL PROTECTED] http://adamaircraft.com US Office: +1(303)406-5959 US Mobile: +1(917)855-7109 US Facsimile: +1(646)349-2741 US Home Office: +1(720)227-0137 ___ 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] Voicepulse down?
They were down and now back up. Sean John L. Magee wrote: no DNS resolution to begin with. Anyone heard anything about this? jlm ___ 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] Voicepulse connect - unable to dial out, asterisk says 9696
Hi, for some weeks now I have been unable to make calls via my voicepulse connect IAX account? When I attempt the console looks like this:- rt*CLI -- Executing Dial(SIP/2008-cf55, IAX2/NBhXX:[EMAIL PROTECTED]/12124565900) in new stack -- Called NBhX:[EMAIL PROTECTED]/12124565900 -- Call accepted by 66.234.228.160 (format ulaw) -- Format for call is ulaw -- Hungup 'IAX2/66.234.228.160:4569/1' -- Executing HasNewVoicemail(SIP/2008-cf55, 2002) in new stack rt*CLI and Asterisk speaks back to me 96 96 And thats it!? I'm not aware I changed anything at this end. Asterisk 1.07. Thanks Mike ___ 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] VoicePulse DTMF Problems Anyone?
We are developing an IVR application and when I am testing locally on my machine using a softphone (iaxcomm) the digits I press for GET DATA work every time. I am testing with a local extension that goes right into my routine. However when I try to call in to the system using an analog or cell phone GET DATA drops some digits that are pressed. There doesn't seem to be a pattern to which digits get dropped either. Digits in the beginning middle or end gets dropped equally. I am wondering if anyone else is experiencing similar issues. I believe the problem lies with VoicePulse because we are using them for IAX connections. I don't believe its a bandwidth problem on my network (cable) because I have tried the same exact system/config everything on another network (T1) and the same digit dropping continues to happen. This is happening with a load of 1 call. Is this problem with VoicePulse? Is anyone else experiencing it? Can anyone recommend a more reliable company? In most previous cases, dtmf issues have been related to how you define your interfaces. For sip definitions, use dtmfmode=rfc2833. Some itsp's have an issue with asterisk in that a completed iax call to an asterisk IVR is considered an answered call, and therefore expect dtmf tones to be passed to the endpoints. In this case, the dtmf tones are expected to be generated by the phone and passed to the IVR as inband audio tones. I'm not a voicepulse user, so don't know if they have some particular problem or not. If the dtmf digits are expected to be passed as inband audio tones, then a reasonable codec would need to be specified. Might try ulaw if you are using something different now. My system has iax trunks from multiple itsp providers, multiple iax links to other companies that we work with, a variety of sip phones (each defined with rfc2833), and multiple analog pstn lines. We don't have a problem (cvs-head) with an IVR that starts out as: [bus-ivr-main] exten = s,1,Wait,1 exten = s,2,Answer exten = s,3,Set(TIMEOUT(digit)=5) exten = s,4,Set(TIMEOUT(response)=15) exten = s,5,Background(abc-greeting) ; Thanks for calling press 1 for exten = s,6,Hangup ___ 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] VoicePulse DTMF Problems Anyone?
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: I am wondering if anyone else is experiencing similar issues. I believe the problem lies with VoicePulse because we are using them for IAX connections. I don't believe its a bandwidth problem on my network (cable) because I have tried the same exact system/config everything on another network (T1) and the same digit dropping continues to happen. This is happening with a load of 1 call. Is this problem with VoicePulse? Is anyone else experiencing it? Can anyone recommend a more reliable company? In most previous cases, dtmf issues have been related to how you define your interfaces. For sip definitions, use dtmfmode=rfc2833. Some itsp's have an issue with asterisk in that a completed iax call to an asterisk IVR is considered an answered call, and therefore expect dtmf tones to be passed to the endpoints. In this case, the dtmf tones are expected to be generated by the phone and passed to the IVR as inband audio tones. I'm not a voicepulse user, so don't know if they have some particular problem or not. My system has iax trunks from multiple itsp providers, multiple iax links to other companies that we work with, a variety of sip phones (each defined with rfc2833), and multiple analog pstn lines. We don't have a problem (cvs-head) with an IVR that starts out as: Rich, Thanks for the input. I am just using the default Asterisk settings for IAX so I would think in that case I wouldn't be the only person experiencing this. What I did was set up an account with BroadVoice and setup a SIP connection. After trying about 15 times, this new connection has gotten every digit pressed. When we started developing 3 weeks ago the VoicePulse IAX setup I have was catching all the digits I would press. It seems only lately that the same setup has gotten worse (although at certain times it works well). It does seem to me the problem was probably due to some network issues at VoicePulse. Thanks, Michael ___ 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] VoicePulse DTMF Problems Anyone?
I am wondering if anyone else is experiencing similar issues. I believe the problem lies with VoicePulse because we are using them for IAX connections. I don't believe its a bandwidth problem on my network (cable) because I have tried the same exact system/config everything on another network (T1) and the same digit dropping continues to happen. This is happening with a load of 1 call. Is this problem with VoicePulse? Is anyone else experiencing it? Can anyone recommend a more reliable company? In most previous cases, dtmf issues have been related to how you define your interfaces. For sip definitions, use dtmfmode=rfc2833. Some itsp's have an issue with asterisk in that a completed iax call to an asterisk IVR is considered an answered call, and therefore expect dtmf tones to be passed to the endpoints. In this case, the dtmf tones are expected to be generated by the phone and passed to the IVR as inband audio tones. I'm not a voicepulse user, so don't know if they have some particular problem or not. My system has iax trunks from multiple itsp providers, multiple iax links to other companies that we work with, a variety of sip phones (each defined with rfc2833), and multiple analog pstn lines. We don't have a problem (cvs-head) with an IVR that starts out as: Rich, Thanks for the input. I am just using the default Asterisk settings for IAX so I would think in that case I wouldn't be the only person experiencing this. What I did was set up an account with BroadVoice and setup a SIP connection. After trying about 15 times, this new connection has gotten every digit pressed. When we started developing 3 weeks ago the VoicePulse IAX setup I have was catching all the digits I would press. It seems only lately that the same setup has gotten worse (although at certain times it works well). It does seem to me the problem was probably due to some network issues at VoicePulse. That's entirely possible. Had something similar with livevoip.com (with the answered iax call issue). You should be able to determine whether its a voicepulse issue by either doing a iax debug (look for the dtmf digits), or, using ethereal to trace the packets. Both methods should show the pressed dtmf digits as values passed in the iax frame. If you don't see those, then its likely voicepulse is passing the dtmf tones as audio (try different codec). ___ 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] VoicePulse DTMF Problems Anyone?
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: That's entirely possible. Had something similar with livevoip.com (with the answered iax call issue). You should be able to determine whether its a voicepulse issue by either doing a iax debug (look for the dtmf digits), or, using ethereal to trace the packets. Both methods should show the pressed dtmf digits as values passed in the iax frame. If you don't see those, then its likely voicepulse is passing the dtmf tones as audio (try different codec). Thanks! I'll try that. Michael ___ 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] VoicePulse DTMF Problems Anyone?
Michael, try relaxdtmf=yes in your iax.conf, or if you are using sip, then in sip.conf regards, Umair bari Michael Stearne wrote: On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: That's entirely possible. Had something similar with livevoip.com (with the answered iax call issue). You should be able to determine whether its a voicepulse issue by either doing a iax debug (look for the dtmf digits), or, using ethereal to trace the packets. Both methods should show the pressed dtmf digits as values passed in the iax frame. If you don't see those, then its likely voicepulse is passing the dtmf tones as audio (try different codec). Thanks! I'll try that. Michael ___ 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 -- Regards, Umair Bari Tech Support Dept. Super Technologies Inc. http://www.supertec.com Voice : 1-408-884-1966 ___ 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] VoicePulse DTMF Problems Anyone?
I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf. (per the most recent sample configs) Michael, try relaxdtmf=yes in your iax.conf, or if you are using sip, then in sip.conf regards, Umair bari Michael Stearne wrote: On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: That's entirely possible. Had something similar with livevoip.com (with the answered iax call issue). You should be able to determine whether its a voicepulse issue by either doing a iax debug (look for the dtmf digits), or, using ethereal to trace the packets. Both methods should show the pressed dtmf digits as values passed in the iax frame. If you don't see those, then its likely voicepulse is passing the dtmf tones as audio (try different codec). Thanks! I'll try that. Michael ___ 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 -- Regards, Umair Bari Tech Support Dept. Super Technologies Inc. http://www.supertec.com Voice : 1-408-884-1966 ___ 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 ---End of Original Message- ___ 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] VoicePulse DTMF Problems Anyone?
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf. (per the most recent sample configs) I didn't find it either. I put it in the config anyway but it didn't seem to make a difference. I also tried changing the call codec to ulaw but that had no significant change either. I am taking in 6 six digits maybe other people are experiencing this but I see it more because of the length of the digitas being taken in. Michael Michael, try relaxdtmf=yes in your iax.conf, or if you are using sip, then in sip.conf regards, Umair bari Michael Stearne wrote: On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: That's entirely possible. Had something similar with livevoip.com (with the answered iax call issue). You should be able to determine whether its a voicepulse issue by either doing a iax debug (look for the dtmf digits), or, using ethereal to trace the packets. Both methods should show the pressed dtmf digits as values passed in the iax frame. If you don't see those, then its likely voicepulse is passing the dtmf tones as audio (try different codec). Thanks! I'll try that. Michael ___ 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 -- Regards, Umair Bari Tech Support Dept. Super Technologies Inc. http://www.supertec.com Voice : 1-408-884-1966 ___ 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 ---End of Original Message- ___ 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] VoicePulse DTMF Problems Anyone?
I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf. (per the most recent sample configs) I didn't find it either. I put it in the config anyway but it didn't seem to make a difference. I also tried changing the call codec to ulaw but that had no significant change either. I am taking in 6 six digits maybe other people are experiencing this but I see it more because of the length of the digitas being taken in. Did you see the Type: DTMF Subclass: 3 (for pressing the 3 digit) in the iax debug? If you're seeing those, then codec selection has nothing to do with it. We take in four digits on a regular basis with no errors at all. I would doubt the number of digits has anything to do with it; it either has accurate dtmf interpretation or you don't on a per digit basis. ___ 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] VoicePulse DTMF Problems Anyone?
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf. (per the most recent sample configs) I didn't find it either. I put it in the config anyway but it didn't seem to make a difference. I also tried changing the call codec to ulaw but that had no significant change either. I am taking in 6 six digits maybe other people are experiencing this but I see it more because of the length of the digitas being taken in. Did you see the Type: DTMF Subclass: 3 (for pressing the 3 digit) in the iax debug? I see that for SIP calls but I do not see a per digit basis for IAX calls. If you're seeing those, then codec selection has nothing to do with it. We take in four digits on a regular basis with no errors at all. I would doubt the number of digits has anything to do with it; it either has accurate dtmf interpretation or you don't on a per digit basis. How can I turn on per digit readings with IAX? Thanks, Michael ___ 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] VoicePulse DTMF Problems Anyone?
I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf. (per the most recent sample configs) I didn't find it either. I put it in the config anyway but it didn't seem to make a difference. I also tried changing the call codec to ulaw but that had no significant change either. I am taking in 6 six digits maybe other people are experiencing this but I see it more because of the length of the digitas being taken in. Did you see the Type: DTMF Subclass: 3 (for pressing the 3 digit) in the iax debug? I see that for SIP calls but I do not see a per digit basis for IAX calls. If you're seeing those, then codec selection has nothing to do with it. We take in four digits on a regular basis with no errors at all. I would doubt the number of digits has anything to do with it; it either has accurate dtmf interpretation or you don't on a per digit basis. How can I turn on per digit readings with IAX? By doing iax2 debug and arranging an inbound call where someone presses the dtmf keypad. Debug will create a fair amount of cli output and you have to look closely for Type: DTMF Subclass: 3 messages intermingled in the cli output. If you are not seeing any of those, then voicepulse is sending the dtmf via inband audio tones. The accuracy of inband audio tones will be less then if the dtmf digits are sent within iax packets (Type: dtmf). If they are arriving via inband audio, that's likely your problem as any missed or dropped iax frames will seriously distort the dtmf audio. Asterisk won't be able to detect the correct digit. Since you indicated that sometimes it works and other times it doesn't, that probably is indicative of network congestion between the two endpoints (your asterisk and voicepulse) and missed or dropped packets. ___ 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] VoicePulse DTMF Problems Anyone?
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: If you are not seeing any of those, then voicepulse is sending the dtmf via inband audio tones. The accuracy of inband audio tones will be less then if the dtmf digits are sent within iax packets (Type: dtmf). If they are arriving via inband audio, that's likely your problem as any missed or dropped iax frames will seriously distort the dtmf audio. Asterisk won't be able to detect the correct digit. Since you indicated that sometimes it works and other times it doesn't, that probably is indicative of network congestion between the two endpoints (your asterisk and voicepulse) and missed or dropped packets. Using iax2 debug I can see when numbers are pressed it's just some never make it through so they don't show up in the debug. Thanks for all you input. Michael ___ 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] VoicePulse DTMF Problems Anyone?
If you are not seeing any of those, then voicepulse is sending the dtmf via inband audio tones. The accuracy of inband audio tones will be less then if the dtmf digits are sent within iax packets (Type: dtmf). If they are arriving via inband audio, that's likely your problem as any missed or dropped iax frames will seriously distort the dtmf audio. Asterisk won't be able to detect the correct digit. Since you indicated that sometimes it works and other times it doesn't, that probably is indicative of network congestion between the two endpoints (your asterisk and voicepulse) and missed or dropped packets. Using iax2 debug I can see when numbers are pressed it's just some never make it through so they don't show up in the debug. Okay, then you've just proven that its a voicepulse problem, or, you're experiencing dropped packets between your asterisk and voicepulse. If the audio on calls is good, then its probably voicepulse. ___ 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] VoicePulse DTMF Problems Anyone?
We are developing an IVR application and when I am testing locally on my machine using a softphone (iaxcomm) the digits I press for GET DATA work every time. I am testing with a local extension that goes right into my routine. However when I try to call in to the system using an analog or cell phone GET DATA drops some digits that are pressed. There doesn't seem to be a pattern to which digits get dropped either. Digits in the beginning middle or end gets dropped equally. I am wondering if anyone else is experiencing similar issues. I believe the problem lies with VoicePulse because we are using them for IAX connections. I don't believe its a bandwidth problem on my network (cable) because I have tried the same exact system/config everything on another network (T1) and the same digit dropping continues to happen. This is happening with a load of 1 call. Is this problem with VoicePulse? Is anyone else experiencing it? Can anyone recommend a more reliable company? Thanks, Michael ___ 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] Voicepulse problems?
Hi all, Is anybody else experiencing problems with voicepulse? Today and over the weekend? I've had problems with both gateways, but one usually works when the other doesn't. I'm trying to eliminate my network from the problem. Sean ___ 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] Voicepulse problems?
Yep I had a problem with their second gw; moved over to the first gw and so far so good. Hector Villalobos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Monday, May 16, 2005 9:24 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicepulse problems? Hi all, Is anybody else experiencing problems with voicepulse? Today and over the weekend? I've had problems with both gateways, but one usually works when the other doesn't. I'm trying to eliminate my network from the problem. Sean ___ 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 This e-mail and any attachments are for the authorized use by the intended recipient only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachments and all copies and inform the sender. Thank you. ___ 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] Voicepulse down?
Anyone else using Voicepulse? This morning I noticed that they seem to be doa... no dns resolution, no ping, etc. -Trevor ___ 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] Voicepulse down?
Trevor Harrison wrote: Anyone else using Voicepulse? This morning I noticed that they seem to be doa... no dns resolution, no ping, etc. -Trevor Nope, working fine here ( Modesto California ). Try reversing which gateway you are using first. I did that a while ago and things seem to work fine now. Sean ___ 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] Voicepulse down?
it works here in Chicago. you might want to check with your provider their dns may be out.. that happened with comcast about 3 weeks ago. Original Message Subject: [Asterisk-Users] Voicepulse down? From: Trevor Harrison [EMAIL PROTECTED] Date: Wed, May 11, 2005 8:57 am To: asterisk-users@lists.digium.com Anyone else using Voicepulse? This morning I noticed that they seem to be doa... no dns resolution, no ping, etc. -Trevor ___ 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] Voicepulse down?
Its working for me now also Actually, I did try from 2 different ISP's on two sides of the country with the same results. -Trevor On 5/11/05, Max W Blackmer Jr [EMAIL PROTECTED] wrote: it works here in Chicago. you might want to check with your provider their dns may be out.. that happened with comcast about 3 weeks ago. Original Message Subject: [Asterisk-Users] Voicepulse down? From: Trevor Harrison [EMAIL PROTECTED] Date: Wed, May 11, 2005 8:57 am To: asterisk-users@lists.digium.com Anyone else using Voicepulse? This morning I noticed that they seem to be doa... no dns resolution, no ping, etc. -Trevor ___ 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] Voicepulse down?
gwiaxt01.voicepulse.com working fine 5 mins ago, just used it. Mike On 5/11/05, Trevor Harrison [EMAIL PROTECTED] wrote: Anyone else using Voicepulse? This morning I noticed that they seem to be doa... no dns resolution, no ping, etc. -Trevor ___ 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] Voicepulse connect has doubled their rates
Today I received an email informing me that effective April 1, my per number charge for VOIP will almost double. This is the downside of VOIP. It is unregulated. I have published and distributed my new VOIP phone number, and now, with no warning, my monthly charge has doubled. Ouch.. Beware of which provider you choose! There is nothing to prevent them from doubling my rates again on May 1st! ___ 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] Voicepulse connect has doubled their rates
Tim Burt wrote: Today I received an email informing me that effective April 1, my per number charge for VOIP will almost double. This is the downside of VOIP. It is unregulated. I have published and distributed my new VOIP phone number, and now, with no warning, my monthly charge has doubled. Ouch.. Beware of which provider you choose! There is nothing to prevent them from doubling my rates again on May 1st! Getting a little dramatic there, aren't we? It's going from 7.95 ( 8 bucks ) a month to 11 bucks, an increase of 72%. That's hardly what I'd call doubling ( unless you're using that new math I've heard so much about ). And it's still cheaper than my land line, when you consider that all incoming calls are free, as well as all 1800 numbers. For everything else, there's voip-jet. Not that I apprecaite the raise much myself, but hey, this industry is still in it's infancy. It gets too bad, someone else will take their place. Sean ___ 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] Voicepulse connect has doubled their rates
Call me silly, but arent incoming calls on land lines also free?? -Mark On Wed, 30 Mar 2005 12:12:29 -0800, Sean Kennedy [EMAIL PROTECTED] wrote: Tim Burt wrote: Today I received an email informing me that effective April 1, my per number charge for VOIP will almost double. This is the downside of VOIP. It is unregulated. I have published and distributed my new VOIP phone number, and now, with no warning, my monthly charge has doubled. Ouch.. Beware of which provider you choose! There is nothing to prevent them from doubling my rates again on May 1st! Getting a little dramatic there, aren't we? It's going from 7.95 ( 8 bucks ) a month to 11 bucks, an increase of 72%. That's hardly what I'd call doubling ( unless you're using that new math I've heard so much about ). And it's still cheaper than my land line, when you consider that all incoming calls are free, as well as all 1800 numbers. For everything else, there's voip-jet. Not that I apprecaite the raise much myself, but hey, this industry is still in it's infancy. It gets too bad, someone else will take their place. Sean ___ 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] Voicepulse connect has doubled their rates
It's going from 7.95 ( 8 bucks ) a month to 11 bucks, an increase of 72%. That's hardly what I'd call doubling ( unless you're using that new math I've heard so much about ). h, actually it is only a 28% increase. you want to see outrageous you should see my gas bill. ___ 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] Voicepulse connect has doubled their rates
Max W Blackmer Jr wrote: It's going from 7.95 ( 8 bucks ) a month to 11 bucks, an increase of 72%. That's hardly what I'd call doubling ( unless you're using that new math I've heard so much about ). h, actually it is only a 28% increase. you want to see outrageous you should see my gas bill. My bad. *I* must be using that new math now. Sean ___ 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] Voicepulse connect has doubled their rates
Perhaps I misspoke. A land line would run me ~20 bucks a month. A VP number will run me 11 bucks a month. I only specify the free incoming calls because that's a distguishing characteristic of voip DIDs, many places do not give you free incoming. And anyway, if you are a consumer customer, your incoming calls are not free. So there you go. Sean Mark Musone wrote: Call me silly, but arent incoming calls on land lines also free?? -Mark On Wed, 30 Mar 2005 12:12:29 -0800, Sean Kennedy [EMAIL PROTECTED] wrote: Tim Burt wrote: Today I received an email informing me that effective April 1, my per number charge for VOIP will almost double. This is the downside of VOIP. It is unregulated. I have published and distributed my new VOIP phone number, and now, with no warning, my monthly charge has doubled. Ouch.. Beware of which provider you choose! There is nothing to prevent them from doubling my rates again on May 1st! Getting a little dramatic there, aren't we? It's going from 7.95 ( 8 bucks ) a month to 11 bucks, an increase of 72%. That's hardly what I'd call doubling ( unless you're using that new math I've heard so much about ). And it's still cheaper than my land line, when you consider that all incoming calls are free, as well as all 1800 numbers. For everything else, there's voip-jet. Not that I apprecaite the raise much myself, but hey, this industry is still in it's infancy. It gets too bad, someone else will take their place. Sean ___ 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] VoicePulse Issues
I don't see how it could be firewall issues. I have firewall ports 4569 and 5036 open for UDP traffic to and from the Asterisk server. Yesterday we conducted a conference call that lasted several hours without a drop, just periodic dead spots for a few seconds. Other calls disconnect entirely. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Wednesday, March 23, 2005 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoicePulse Issues Adam Robins wrote: So, I switched to IAX2. Now, everything works fine 95% of the time . . . but every once in a while, perhaps 5 seconds into a call or 20 minutes into a call, the call will simply drop. This occurs several times per week with no observable pattern. I have attached an excerpt from the log file at the end of this message. Has anyone else experienced this? Know what is causing it? Has anyone gotten VoicePulse Connect to work with SIP? Hi Admin, I use the connect service from voicepulse ( as I am sure you do, just specifying for future searches ), and I haven't had any of these problems you have mentioned. I do have a problem when the call is connected, there's about half a second of silence about half a second into the call, on every call. I mention it here in case it's related. Honestly, my first instict says this is a firewall problem. Is that at all possible with your setup? Sean ___ 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 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ 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] VoicePulse Issues
Adam Robins wrote: I recently switched from BroadVoice to VoicePulse Connect on my Asterisk box. Too many Meetme quality complaints (whether real or perceived). I had to make a choice to use IAX2 or SIP with VoicePulse. I first tried to go with SIP because I already had it working and all of our devices are SIP. Problem is that every time I turn my back, the Asterisk registration with the VoicePulse SIP server times out. So, I switched to IAX2. Now, everything works fine 95% of the time . . . but every once in a while, perhaps 5 seconds into a call or 20 minutes into a call, the call will simply drop. This occurs several times per week with no observable pattern. I have attached an excerpt from the log file at the end of this message. Has anyone else experienced this? Know what is causing it? Has anyone gotten VoicePulse Connect to work with SIP? Thanks, Adam In cases where a provider offers a choice of SIP or IAX2 I have been wondering if the provider would be using a * server in either case. I prefer IAX2 as long as I know the provider has adequate server capacity for the load. Otherwise I would think that getting my handoff directly from SIP gateways would be more reliable. ___ 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] VoicePulse Issues
Adam Robins wrote: So, I switched to IAX2. Now, everything works fine 95% of the time . . . but every once in a while, perhaps 5 seconds into a call or 20 minutes into a call, the call will simply drop. This occurs several times per week with no observable pattern. I have attached an excerpt from the log file at the end of this message. Has anyone else experienced this? Know what is causing it? Has anyone gotten VoicePulse Connect to work with SIP? Hi Admin, I use the connect service from voicepulse ( as I am sure you do, just specifying for future searches ), and I haven't had any of these problems you have mentioned. I do have a problem when the call is connected, there's about half a second of silence about half a second into the call, on every call. I mention it here in case it's related. Honestly, my first instict says this is a firewall problem. Is that at all possible with your setup? Sean ___ 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] VoicePulse Issues
Jared Watkins wrote: Adam Robins wrote: So, I switched to IAX2. Now, everything works fine 95% of the time . . . but every once in a while, perhaps 5 seconds into a call or 20 minutes into a call, the call will simply drop. This occurs several times per week with no observable pattern I have also seen this in the last couple of weeks... I've had some long (90 min) calls.. and had it happen a few times during the same call... though I've not taken the time to pull any logs. My net service at home can be spotty sometimes... so I thought it might have been caused by that. It's not been bad enough to terminate a call... just 10-15 seconds of silence. I have had similar problems with VoicePulse for 2 - 3 months now. Along with random dropped calls, we have also experienced dial-in problems. VoicePulse tech support has been utterly silent despite several emails and phone calls. We're now trying out SixTel and Live VoIP. Live VoIP has recently had a problem with our (and others) DID numbers, but they have been very responsive and it looks like the problem will be solved shortly. Joe ___ 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] VoicePulse Issues
Adam Robins wrote: I recently switched from BroadVoice to VoicePulse Connect on my Asterisk box. Too many Meetme quality complaints (whether real or perceived). I had to make a choice to use IAX2 or SIP with VoicePulse. I first tried to go with SIP because I already had it working and all of our devices are SIP. Problem is that every time I turn my back, the Asterisk registration with the VoicePulse SIP server times out. So, I switched to IAX2. Now, everything works fine 95% of the time . . . but every once in a while, perhaps 5 seconds into a call or 20 minutes into a call, the call will simply drop. This occurs several times per week with no observable pattern. I have attached an excerpt from the log file at the end of this message. Has anyone else experienced this? Know what is causing it? Has anyone gotten VoicePulse Connect to work with SIP? I have also seen this in the last couple of weeks... I've had some long (90 min) calls.. and had it happen a few times during the same call... though I've not taken the time to pull any logs. My net service at home can be spotty sometimes... so I thought it might have been caused by that. It's not been bad enough to terminate a call... just 10-15 seconds of silence. Jared ___ 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] VoicePulse Issues
I recently switched from BroadVoice to VoicePulse Connect on my Asterisk box. Too many Meetme quality complaints (whether real or perceived). I had to make a choice to use IAX2 or SIP with VoicePulse. I first tried to go with SIP because I already had it working and all of our devices are SIP. Problem is that every time I turn my back, the Asterisk registration with the VoicePulse SIP server times out. So, I switched to IAX2. Now, everything works fine 95% of the time . . . but every once in a while, perhaps 5 seconds into a call or 20 minutes into a call, the call will simply drop. This occurs several times per week with no observable pattern. I have attached an excerpt from the log file at the end of this message. Has anyone else experienced this? Know what is causing it? Has anyone gotten VoicePulse Connect to work with SIP? Thanks, Adam Mar 17 11:50:23 VERBOSE[9987]: -- Executing Dial(SIP/2034-771f, IAX2/[EMAIL PROTECTED]/19043317785) in new stack Mar 17 11:50:23 VERBOSE[9987]: -- Called [EMAIL PROTECTED]/19043317785 Mar 17 11:50:23 VERBOSE[24993]: -- Call accepted by 66.234.228.160 (format ulaw) Mar 17 11:50:23 VERBOSE[24993]: -- Format for call is ulaw Mar 17 11:50:23 VERBOSE[9987]: -- IAX2/voicepulse-out-01/6 stopped sounds Mar 17 11:50:23 VERBOSE[9987]: -- IAX2/voicepulse-out-01/6 is making progress passing it to SIP/2034-771f Mar 17 11:50:23 DEBUG[24993]: Ooh, voice format changed to 4 Mar 17 11:50:33 VERBOSE[9987]: -- IAX2/voicepulse-out-01/6 answered SIP/2034-771f Mar 17 11:50:33 DEBUG[24992]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Found Mar 17 11:51:03 DEBUG[24993]: Immediately destroying 7, having received INVAL Mar 17 11:51:43 DEBUG[24993]: Immediately destroying 4, having received INVAL Mar 17 11:51:43 DEBUG[24993]: Raw Hangup 69.73.19.178:4569, src=4, dst=285 Mar 17 11:52:32 DEBUG[24992]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 156: Found Mar 17 11:52:43 DEBUG[24993]: Sending VNAK Mar 17 11:52:48 DEBUG[24992]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 156: Found Mar 17 11:53:04 DEBUG[24993]: Immediately destroying 6, having received INVAL Mar 17 11:53:04 DEBUG[9987]: Didn't get a frame from channel: IAX2/voicepulse-out-01/6 Mar 17 11:53:04 DEBUG[9987]: Bridge stops bridging channels SIP/2034-771f and IAX2/voicepulse-out-01/6 Mar 17 11:53:04 DEBUG[9987]: We're hanging up IAX2/voicepulse-out-01/6 now... Mar 17 11:53:04 DEBUG[9987]: Really destroying IAX2/voicepulse-out-01/6 now... Mar 17 11:53:04 VERBOSE[9987]: -- Hungup 'IAX2/voicepulse-out-01/6' Mar 17 11:53:04 DEBUG[9987]: Exiting with DIALSTATUS=ANSWER. Mar 17 11:53:04 VERBOSE[9987]: == Spawn extension (intl-access, 919043317785, 2) exited non-zero on 'SIP/2034-771f' Mar 17 11:53:04 DEBUG[9987]: update_user_counter(2034) - decrement inUse counter Mar 17 11:53:04 DEBUG[24992]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ 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] voicepulse silence during conversations
Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off chance this could be something on my end: Asterisk 1.0.0 Connecting to voicepulse via iax, ulaw codec Polycom 500 IP SIP phone, ulaw codec I'll be honest, I don't notice it at all, but my customer does, and I'd like to make them as happy as I can with this system. Also ( I would feel silly making another thread out of this ) what are the common reasons for echo between sip phones going through two different asterisk servers? As in phone - asterisk A - asterisk B - phone. I've been searching for it, but I'm not having much luck. Thank you, any help is greatly apprecaited! Sean ___ 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] voicepulse silence during conversations
On Wed, 09 Mar 2005 13:06:24 -0800 Sean Kennedy [EMAIL PROTECTED] wrote: Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off chance this could be something on my end: Asterisk 1.0.0 Connecting to voicepulse via iax, ulaw codec Polycom 500 IP SIP phone, ulaw codec Not sure if the ploycom has it, but make sure that it is set to transmit silence. Sounds like this option is set not to which will cause that dead sound. Robert ___ 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] voicepulse silence during conversations
Yes, Ironic isn't it. That SIP/VOIP is so digitally clear that you don't hear the pops, cracks and whistles of the old analog phones. The only analog is from the human to the machine. The old analog phone humans hear it, soon there will another generation of humans who have never used an analog phone. Anyone remember the transition from long distance operators to direct dial. Or from pulse to touch tone? Back in 1992 I tried to make a calling card call using a rotary phone in Alabama, where they had 5 digit dialing. I was stumped looking at a phone with no pound/# sign on it. I first noticed this silence quirk when I was working with a 3COM SIP phone back in 2000. The crystal clear voice and silence made me feel like the phone was not working or that the other person had hung-up. You also have to be careful of background noise in the room; phones with good microphones will let the other end here everything going in the room you are in. Race The Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Wednesday, March 09, 2005 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] voicepulse silence during conversations Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off chance this could be something on my end: Asterisk 1.0.0 Connecting to voicepulse via iax, ulaw codec Polycom 500 IP SIP phone, ulaw codec I'll be honest, I don't notice it at all, but my customer does, and I'd like to make them as happy as I can with this system. Also ( I would feel silly making another thread out of this ) what are the common reasons for echo between sip phones going through two different asterisk servers? As in phone - asterisk A - asterisk B - phone. I've been searching for it, but I'm not having much luck. Thank you, any help is greatly apprecaited! Sean ___ 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] voicepulse silence during conversations
the issue is lack of sidetone u can google sidetone sidetone is feedback u get from the mike to your earpiece that the fone generates to let u know the circuit did not go dead when people stop talking i find the lace of sidetone extremely annoying and so will many customers with asterisk i have found lack of sidetone on the grandstream budgetone i have found perfect sidetone on the cisco 79xx and also on the sipuras Race Vanderdecken wrote: Yes, Ironic isn't it. That SIP/VOIP is so digitally clear that you don't hear the pops, cracks and whistles of the old analog phones. The only analog is from the human to the machine. The old analog phone humans hear it, soon there will another generation of humans who have never used an analog phone. Anyone remember the transition from long distance operators to direct dial. Or from pulse to touch tone? Back in 1992 I tried to make a calling card call using a rotary phone in Alabama, where they had 5 digit dialing. I was stumped looking at a phone with no pound/# sign on it. I first noticed this silence quirk when I was working with a 3COM SIP phone back in 2000. The crystal clear voice and silence made me feel like the phone was not working or that the other person had hung-up. You also have to be careful of background noise in the room; phones with good microphones will let the other end here everything going in the room you are in. Race The Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Wednesday, March 09, 2005 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] voicepulse silence during conversations Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off chance this could be something on my end: Asterisk 1.0.0 Connecting to voicepulse via iax, ulaw codec Polycom 500 IP SIP phone, ulaw codec I'll be honest, I don't notice it at all, but my customer does, and I'd like to make them as happy as I can with this system. Also ( I would feel silly making another thread out of this ) what are the common reasons for echo between sip phones going through two different asterisk servers? As in phone - asterisk A - asterisk B - phone. I've been searching for it, but I'm not having much luck. Thank you, any help is greatly apprecaited! Sean ___ 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 -- Allen Niven GlobalFone 350 Fifth Avenue #6206 Empire State Building New York, NY 10118 +1-212-678-4381 office +1-646-246-7415 cell http://www.GlobalFone.biz Instant Messaging Accounts ICQ 137763656 Yahoo Messenger [EMAIL PROTECTED] MSN Messenger [EMAIL PROTECTED] PLEASE NOTE I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO ___ 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] Voicepulse Open Access Asterisk Problems
I can't seem to dial out with Voicepulse Open Access service using *. Incoming works fine. Another user posted a few weeks back that they were having problems and there are some threads at dslreports.com about this as well. Maybe someone here can figure out what the issue is from the sip debug info below. I am at a loss. The audible error message from Allison is 0984 (from VP server) Here is all the pertinent info: [sip.conf] [general] port = 5060 bindaddr = 0.0.0.0 srvlookup=yes tos=lowdelay maxexpirey=3600 disallow=all allow=ulaw musicclass=default language=en relaxdtmf=yes ;useragent=Asterisk PBX ;nat=yes register = s00**:[EMAIL PROTECTED] externip=asterisk.briandingman.com localnet=192.168.1.0/255.255.0.0 [voicepulse] type=friend context=voicepulse-incoming username=s00** secret= host=access1.voicepulse.com dtmf=inband nat=yes qualify=yes canreinvite=no insecure=very [1000] type=friend host=dynamic ;callerid=Brian 1000 dtmfmode=rfc2833 mailbox=1000 context=Home ;nat=no ;qualify=yes secret= Error message from CLI: -- Executing Macro(SIP/1000-fbdb, vp-dial|16109951010) in new stack -- Executing Dial(SIP/1000-fbdb, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/voicepulse-e009 is making progress passing it to SIP/1000-fbdb Feb 17 17:08:42 WARNING[8523]: chan_sip.c:6811 handle_response: Forbidden - wrong password on authentication for INVITE to '1000 sip:[EMAIL PROTECTED];tag=as3e632d2a' -- SIP/voicepulse-e009 is circuit-busy == Everyone is busy/congested at this time -- Executing Hangup(SIP/1000-fbdb, ) in new stack == Spawn extension (macro-vp-dial, s, 2) exited non-zero on 'SIP/1000-fbdb' in macro 'vp-dial' == Spawn extension (Home, 16109951010, 1) exited non-zero on 'SIP/1000-fbdb' -- Got SIP response 481 Call Leg Does Not Exist back from 66.234.228.159 (Sorry for the length) SIP Debug info: -- Executing Macro(SIP/1000-cd47, vp-dial|16109951010) in new stack -- Executing Dial(SIP/1000-cd47, SIP/[EMAIL PROTECTED]) in new stack We're at 68.163.52.50 port 15640 Answering/Requesting with root capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport From: 1000 sip:[EMAIL PROTECTED];tag=as74c56bff To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 17 Feb 2005 22:10:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 214 v=0 o=root 8523 8523 IN IP4 68.163.52.50 s=session c=IN IP4 68.163.52.50 t=0 0 m=audio 15640 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 66.234.228.159:5060 -- Called [EMAIL PROTECTED] asterisk*CLI Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;received=68.163.52.50;rport=50210 From: 1000 sip:[EMAIL PROTECTED];tag=as74c56bff To: sip:[EMAIL PROTECTED];tag=as1ecc3219 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: VoicePulse SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=uasw001.voicepulse.com, nonce=5d626333 Content-Length: 0 11 headers, 0 lines Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport From: 1000 sip:[EMAIL PROTECTED];tag=as74c56bff To: sip:[EMAIL PROTECTED];tag=as1ecc3219 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 66.234.228.159:5060 We're at 68.163.52.50 port 15640 Answering/Requesting with root capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport From: 16109951010 sip:[EMAIL PROTECTED];tag=as74c56bff To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username=s00**, realm=uasw001.voicepulse.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=5d626333, response=HASH***, opaque= Date: Thu, 17 Feb 2005 22:10:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 214 v=0 o=root 8523 8524 IN IP4 68.163.52.50 s=session c=IN IP4 68.163.52.50 t=0 0 m=audio 15640 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 66.234.228.159:5060 asterisk*CLI Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;received=68.163.52.50;rport=50210 From: 16109951010 sip:[EMAIL PROTECTED];tag=as74c56bff To: sip:[EMAIL PROTECTED];tag=as0630cede Call-ID: [EMAIL PROTECTED]
[Asterisk-Users] Voicepulse DNID is blank - Any other options?
I just signed up for a second voicepulse number. I assumed that I would be able to differentiate which number the caller dialed. But DNID is empty and DIALEDPEERNUMBER is empty and CHANNEL contains the same info (almost, with the exception of a randomly assigned suffix) for both numbers. Does anyone know how I might determine which number was called? Note, this is not CALLERID. I need the number that the caller CALLED. As a last resort, I guess I could use a different provider for the second number. Can anyone shed any light? Thanks in advance! ___ 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] Voicepulse DNID is blank - Any other options?
Maybe I am missing your exact point, but what about handling this in your extensions.conf [voicepulse-incoming] exten = 2124007999,1,Goto(nyc,s,1) exten = 2124007998,1,Goto(nyc2,s,1) That will put calls to 2124007999 into context nyc and calls to 2124007998 into context nyc2. I guess the real questions is what is your ultimate goal? On Sun, 6 Feb 2005 12:52:21 -0800 (PST), Tim Burt [EMAIL PROTECTED] wrote: I just signed up for a second voicepulse number. I assumed that I would be able to differentiate which number the caller dialed. But DNID is empty and DIALEDPEERNUMBER is empty and CHANNEL contains the same info (almost, with the exception of a randomly assigned suffix) for both numbers. Does anyone know how I might determine which number was called? Note, this is not CALLERID. I need the number that the caller CALLED. As a last resort, I guess I could use a different provider for the second number. Can anyone shed any light? Thanks in advance! ___ 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] Voicepulse DNID is blank - Any other options?
Ah.. the obvious. I don't know why I missed it. I am just a newbie at this PBX stuff. Thanks for the pointer. It worked. First off. Hopefully, someday soon, I will contribute more than silly questions to this list! Thanks again! Maybe I am missing your exact point, but what about handling this in your extensions.conf [voicepulse-incoming] exten = 2124007999,1,Goto(nyc,s,1) exten = 2124007998,1,Goto(nyc2,s,1) That will put calls to 2124007999 into context nyc and calls to 2124007998 into context nyc2. I guess the real questions is what is your ultimate goal? On Sun, 6 Feb 2005 12:52:21 -0800 (PST), Tim Burt [EMAIL PROTECTED] wrote: I just signed up for a second voicepulse number. I assumed that I would be able to differentiate which number the caller dialed. But DNID is empty and DIALEDPEERNUMBER is empty and CHANNEL contains the same info (almost, with the exception of a randomly assigned suffix) for both numbers. Does anyone know how I might determine which number was called? Note, this is not CALLERID. I need the number that the caller CALLED. As a last resort, I guess I could use a different provider for the second number. Can anyone shed any light? Thanks in advance! ___ 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] Voicepulse DNID is blank - Any other options?
A lot of times we all overlook the obvious or easiest way to do things :) On Sun, 6 Feb 2005 13:26:25 -0800 (PST), Tim Burt [EMAIL PROTECTED] wrote: Ah.. the obvious. I don't know why I missed it. I am just a newbie at this PBX stuff. Thanks for the pointer. It worked. First off. Hopefully, someday soon, I will contribute more than silly questions to this list! Thanks again! ___ 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] VoicePulse OpenAccess
They have an entire knowledge base with example scripts, etc on there web site. You can also call them and reach someone in tech support during the day. - Original Message - From: Keith O'Brien To: asterisk-users@lists.digium.com Sent: Sunday, December 19, 2004 11:09 AM Subject: [Asterisk-Users] VoicePulse OpenAccess Has anyone been able to get * working with VoicePulse OpenAccess (SIP not IAX). I have found a ton of information about VoicePulse Connect but very little on the proper * settings for OpenAccess. Tried contacting VP with no response. If anyone has this working, can they share their extensions.conf and sip.conf files? Better yet, if it could be posted on the Wiki Keith ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Voicepulse down for anyone else?
Thanks,Steve Totaro[EMAIL PROTECTED]www.totarotechnologies.com ___ 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 down for anyone else?
They were for me.. But back up now.. brian From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 18, 2004 1:43 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicepulse down for anyone else? Thanks, Steve Totaro [EMAIL PROTECTED] www.totarotechnologies.com ___ 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 down for anyone else?
Does VoicePulse use Level3 ? If so there is a reported problem in the Washington DC area that seems to have been corrected. - Original Message - From: Steve Totaro To: [EMAIL PROTECTED] Sent: Monday, October 18, 2004 11:43 AM Subject: [Asterisk-Users] Voicepulse down for anyone else? Thanks,Steve Totaro[EMAIL PROTECTED]www.totarotechnologies.com ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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 down for anyone else?
Could be this: From: Jon Lewis [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: Level 3 US east coast issues On Mon, 18 Oct 2004, Grant A. Kirkwood wrote: Level 3 experiencing widespread unspecified routing issues on the US east coast. Master ticket 1086844. Anyone have more specific information? More, but not specific. We shut off our BGP session to them as lots of sites were unreachable through Level3. I'm waiting for a callback to say it might be safe to turn it back on. backbone impairment is pretty vague. Could be a fiber cut, crashed router, bad software upgrade, etc. Hopefully they know more about it than they're saying. Summary Level 3 is currently experiencing a backbone outage causing routing instability and packet loss. We are working to restore and will be sending hourly updates. Service Impact Statement Level 3 is currently experiencing a backbone impairment. Affected Locations Routing instability and packet loss are possible network wide, but concentrated on the eastern US. Steve Totaro wrote: Thanks, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.totarotechnologies.com http://www.totarotechnologies.com ___ 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 down for anyone else?
Well I am a few minutes outside of DC so thats probably it. Quite a bit of hops through Level3 for me. 110 ms 10 ms10 ms 192.168.2.1 2 10 ms10 ms10 ms 10.91.40.1 310 ms10 ms10 ms fe-3-6-ar01.howardcounty.md.md02.comcast.net [68 .87.59.69] 410 ms10 ms10 ms pos-7-1-cr01.ritchieroad.md.core.comcast.net [68 .87.19.153] 510 ms10 ms10 ms 12.118.122.9 610 ms10 ms10 ms tbr2-p011701.wswdc.ip.att.net [12.123.9.110] 710 ms10 ms10 ms ggr2-p390.wswdc.ip.att.net [12.123.9.85] 820 ms10 ms10 ms so-0-3-0.edge2.Washington1.Level3.net [4.68.127. 153] 910 ms21 ms10 ms so-1-1-0.bbr2.Washington1.Level3.net [64.159.3.6 5] 1020 ms20 ms10 ms as-1-0.bbr2.NewYork1.Level3.net [64.159.1.85] 1120 ms20 ms10 ms ge-7-2.ipcolo2.NewYork1.Level3.net [64.159.17.16 4] 1220 ms20 ms20 ms unknown.Level3.net [63.211.32.126] 1320 ms20 ms20 ms 61-224-234-66.transbeam.com [66.234.224.61] 1420 ms20 ms20 ms 134-228-234-66.cosmoweb.net [66.234.228.134] Trace complete. - Original Message - From: Steve Kann [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 18, 2004 1:50 PM Subject: Re: [Asterisk-Users] Voicepulse down for anyone else? Could be this: From: Jon Lewis [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: Level 3 US east coast issues On Mon, 18 Oct 2004, Grant A. Kirkwood wrote: Level 3 experiencing widespread unspecified routing issues on the US east coast. Master ticket 1086844. Anyone have more specific information? More, but not specific. We shut off our BGP session to them as lots of sites were unreachable through Level3. I'm waiting for a callback to say it might be safe to turn it back on. backbone impairment is pretty vague. Could be a fiber cut, crashed router, bad software upgrade, etc. Hopefully they know more about it than they're saying. Summary Level 3 is currently experiencing a backbone outage causing routing instability and packet loss. We are working to restore and will be sending hourly updates. Service Impact Statement Level 3 is currently experiencing a backbone impairment. Affected Locations Routing instability and packet loss are possible network wide, but concentrated on the eastern US. Steve Totaro wrote: Thanks, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.totarotechnologies.com http://www.totarotechnologies.com ___ 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 down for anyone else?
Yep, my DID's have been out all night and all day so far. Can't get anyone on the phone or through their ticket system. Their site was down for part of the night too. I think it has something to do with the general issues across the net. Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 18, 2004 12:43 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicepulse down for anyone else? Thanks, Steve Totaro [EMAIL PROTECTED] www.totarotechnologies.com ___ 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 down for anyone else?
Can you give me more info on general issues across the net ? Yeah, VoicePulse seems to be having issues, it's usual though. I wish they weren't the only place I knew to get flat rate incoming DID's Nationally in the U.S from. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Your Own ISP .com Sent: Monday, October 18, 2004 2:03 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Voicepulse down for anyone else? Yep, my DID's have been out all night and all day so far. Can't get anyone on the phone or through their ticket system. Their site was down for part of the night too. I think it has something to do with the general issues across the net. Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 18, 2004 12:43 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicepulse down for anyone else? Thanks, Steve Totaro [EMAIL PROTECTED] www.totarotechnologies.com ___ 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 down for anyone else?
When you go hiking in the mountains, always carry a short section of fiber-optic cable in your backpack. If you ever get lost and cannot find your way, just bury the fiber in the ground. A backhoe will arrive shortly thereafter to dig it up... Steve Kann wrote: Could be this: From: Jon Lewis [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: Level 3 US east coast issues On Mon, 18 Oct 2004, Grant A. Kirkwood wrote: Level 3 experiencing widespread unspecified routing issues on the US east coast. Master ticket 1086844. Anyone have more specific information? More, but not specific. We shut off our BGP session to them as lots of sites were unreachable through Level3. I'm waiting for a callback to say it might be safe to turn it back on. backbone impairment is pretty vague. Could be a fiber cut, crashed router, bad software upgrade, etc. Hopefully they know more about it than they're saying. Summary Level 3 is currently experiencing a backbone outage causing routing instability and packet loss. We are working to restore and will be sending hourly updates. Service Impact Statement Level 3 is currently experiencing a backbone impairment. Affected Locations Routing instability and packet loss are possible network wide, but concentrated on the eastern US. Steve Totaro wrote: Thanks, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.totarotechnologies.com http://www.totarotechnologies.com ___ 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 Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ 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 problems since new configs
Voicepulse has ignored
four emails over the course of two weeks.
Anyone have any ideas of whats wrong?
- Executing Dial("IAX2/[EMAIL PROTECTED]:4569/7",
"IAX2/acctname:[EMAIL PROTECTED]/14109649073") in new
stack
-- Called
acctname:[EMAIL PROTECTED]/14109649073
Sep 13 22:48:25 WARNING[131080]: chan_iax2.c:5375 socket_read: Call rejected by 66.234.228.160: No such
context/extension
-- Hungup 'IAX2/vpconnect-t01/8'
== No one is available to answer at this
time
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569/7",
"IAX2/acctname:[EMAIL PROTECTED]/14109649073") in new
stack
-- Called
acctname:[EMAIL PROTECTED]/14109649073
Sep 13 22:48:26 WARNING[131080]: chan_iax2.c:5375 socket_read: Call rejected by 66.234.228.166: No such
context/extension
-- Hungup
'IAX2/66.234.228.166:4569/9'
== No one is available to answer at this
time
I am running todays cvs
head
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Re: [Asterisk-Users] VoicePulse Connect DTMF with IAX2
Here's my iax.conf and extensions.conf (I have not yet made the recent changes they just emailed about a day ago, this is twice in a two month period, jeesh) I have tested inbound and outbound dtmf. I use the g.711 codec and use inband. iax.conf -- [general] port=5036 bindaddr=0.0.0.0 context=incoming ;iaxcompat=yes ; Set iaxcompat to yes if you plan to use layered switches. ; It incurs a small performance hit to enable it. delayreject=yes ; For increased security against brute force password attacks. ; Enabling this will delay the sending of authentication ; reject for REGREQ or AUTHREP if there is a password. amaflags=documentation ; global default AMA flag for iaxtel calls. These flags ; are used in the generation of call detail records. ;accountcode=1 ; default account for Call Detail Records in addition ; to specifying on a per-user basis. language=en ; Global default language for users. ; If omitted, will fallback to english bandwidth=high ; Specify bandwidth of low, medium, or high to ; control which codecs are used in general. allow=all ; Which codecs to allow, same as bandwidth=high disallow=g723.1 ; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. ; You can adjust several parameters relating to the jitter buffer. ; The jitter buffer's function is to compensate for varying network delay. ; All the jitter buffer settings except dropcount are in milliseconds. ; The jitter buffer works for INCOMING audio - the outbound audio ; will be dejittered by the jitter buffer at the other end. ; jitterbuffer=no ; Whether you want the jitter buffer at all. ;dropcount=2; The jitter buffer is sized such that no more than dropcount ; frames would have been too late over the last 2 seconds. ; Set to a small number. 3 represents 1.5% of frames dropped ;maxjitterbuffer=500; A maximum size for the jitter buffer. Setting a reasonable maximum ; here will prevent the call delay from rising to silly values in ; extreme situations. ;maxexcessbuffer=80 ; If conditions improve after a period of high jitter, the jitter buffer ; can end up bigger than necessary. If it ends up more than ; maxexcessbuffer bigger than needed, Asterisk will start gradually ; decreasing the amount of jitter buffering. ;minexcessbuffer=80 ; Sets a desired mimimum amount of headroom in the jitter buffer. ; If Asterisk has less headroom than this, then it will start gradually ; increasing the amount of jitter buffering. ;jittershrinkrate=1 ; When the jitter buffer is being gradually shrunk (or enlarged), ; how many millisecs shall we take off per 20ms frame received? ; Use a small number, or you will be able to hear it changing. ; An example: if you set this to 2, then the jitter buffer size will ; change by 100 millisec per second. ;trunkfreq=20 ; How frequently to send trunk msgs (in ms) authdebug=no; You can disable authentication debugging to reduce ; the amount of debugging traffic. tos=lowdelay; You can set values for your TOS bits to help improve performance. ; Can be lowdelay, throughput, reliability, mincost or none. ;mailboxdetail=yes ; If set to yes, the user receives the actual ; new/old message counts, not just a yes/no as to ; whether they have messages. register = in-xxx##XxX#X:[EMAIL PROTECTED] ; ### PROVIDERS ### [voicepulse]; For inbound context=VPWS type=user host=gw5.voicepulse.com accountcode=1 [vpconnect-t01] ; For outbound type=peer secret=xXx##Xxx## host=gwiaxt01.voicepulse.com auth=md5 qualify=yes accountcode=1 [vpconnect-t02] ; Outbound backup type=peer secret=xXx##Xxx## host=gwiaxt02.voicepulse.com auth=md5 qualify=yes accountcode=1 -- extensions.conf -- [VPWS] ; All Inbound Voicepulse DID numbers go here ; From here it is distributed to the propper place ;; - Some Company - exten = 1235551212,1,Goto(company,1235551212,1) [company] ;
Re: [Asterisk-Users] VoicePulse Connect DTMF with IAX2
I'm using RC2 and last weekend's changes from VoicePulse. Outbound calling and dtmf works fine. However, an inbound call to my DID cannot send dtmf digits to the IVR. Thoughts? Deon Rodden wrote: Here's my iax.conf and extensions.conf (I have not yet made the recent changes they just emailed about a day ago, this is twice in a two month period, jeesh) I have tested inbound and outbound dtmf. I use the g.711 codec and use inband. iax.conf -- [general] port=5036 bindaddr=0.0.0.0 context=incoming ;iaxcompat=yes ; Set iaxcompat to yes if you plan to use layered switches. ; It incurs a small performance hit to enable it. delayreject=yes ; For increased security against brute force password attacks. ; Enabling this will delay the sending of authentication ; reject for REGREQ or AUTHREP if there is a password. amaflags=documentation ; global default AMA flag for iaxtel calls. These flags ; are used in the generation of call detail records. ;accountcode=1 ; default account for Call Detail Records in addition ; to specifying on a per-user basis. language=en ; Global default language for users. ; If omitted, will fallback to english bandwidth=high ; Specify bandwidth of low, medium, or high to ; control which codecs are used in general. allow=all ; Which codecs to allow, same as bandwidth=high disallow=g723.1 ; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. ; You can adjust several parameters relating to the jitter buffer. ; The jitter buffer's function is to compensate for varying network delay. ; All the jitter buffer settings except dropcount are in milliseconds. ; The jitter buffer works for INCOMING audio - the outbound audio ; will be dejittered by the jitter buffer at the other end. ; jitterbuffer=no ; Whether you want the jitter buffer at all. ;dropcount=2; The jitter buffer is sized such that no more than dropcount ; frames would have been too late over the last 2 seconds. ; Set to a small number. 3 represents 1.5% of frames dropped ;maxjitterbuffer=500; A maximum size for the jitter buffer. Setting a reasonable maximum ; here will prevent the call delay from rising to silly values in ; extreme situations. ;maxexcessbuffer=80 ; If conditions improve after a period of high jitter, the jitter buffer ; can end up bigger than necessary. If it ends up more than ; maxexcessbuffer bigger than needed, Asterisk will start gradually ; decreasing the amount of jitter buffering. ;minexcessbuffer=80 ; Sets a desired mimimum amount of headroom in the jitter buffer. ; If Asterisk has less headroom than this, then it will start gradually ; increasing the amount of jitter buffering. ;jittershrinkrate=1 ; When the jitter buffer is being gradually shrunk (or enlarged), ; how many millisecs shall we take off per 20ms frame received? ; Use a small number, or you will be able to hear it changing. ; An example: if you set this to 2, then the jitter buffer size will ; change by 100 millisec per second. ;trunkfreq=20 ; How frequently to send trunk msgs (in ms) authdebug=no; You can disable authentication debugging to reduce ; the amount of debugging traffic. tos=lowdelay; You can set values for your TOS bits to help improve performance. ; Can be lowdelay, throughput, reliability, mincost or none. ;mailboxdetail=yes ; If set to yes, the user receives the actual ; new/old message counts, not just a yes/no as to ; whether they have messages. register = in-xxx##XxX#X:[EMAIL PROTECTED] ; ### PROVIDERS ### [voicepulse]; For inbound context=VPWS type=user host=gw5.voicepulse.com accountcode=1 [vpconnect-t01] ; For outbound type=peer secret=xXx##Xxx## host=gwiaxt01.voicepulse.com auth=md5 qualify=yes accountcode=1 [vpconnect-t02] ; Outbound backup type=peer secret=xXx##Xxx## host=gwiaxt02.voicepulse.com auth=md5 qualify=yes accountcode=1 -- extensions.conf
RE: [Asterisk-Users] VoicePulse Connect DTMF with IAX2
I'm using RC2 and last weekend's changes from VoicePulse. Outbound calling and dtmf works fine. However, an inbound call to my DID cannot send dtmf digits to the IVR. Thoughts? I have the same problem...my iax.conf is set up exactly as recommended per the recent Voicepulse changes and the configs they sent - my CVS is 7/14/04. Both inbound and outbound calling work, but no DTMF received on inbound calls. I found a post on the broadband reports forums regarding this issue, there where a few people who thought that it may affect VP customers who signed up for a DID in a new VP rate center/exchange...for example I've been waiting for VP to offer Colorado DID's (303 or 720) for quite awhile...so when I saw that they were available recently, I jumped on it and ordered one...so this a fairly new area code for them and I have the DTMF problem. I read other people that signed up for a fairly new area code having the same problem and emailing VP support to get it straightened out... I myself have sent them an email which they say they are checking into...I will be sure to let people know what my findings are. Marty ___ 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 Connect DTMF with IAX2
On Tue, Aug 31, 2004 at 10:15:02AM -0400, Deon Rodden wrote:
exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,3,Congestion
This would dial the number twice..?
My config is
exten = _9.,1,Dial,IAX2/voicepulse/011${EXTEN:1}
exten = _9.,2,GotoIf($[ ${DIALSTATUS} != CONGESTION ${DIALSTATUS} != CHANUNAVAIL
]?6)
exten = _9.,3,Dial,IAX2/voicepulse2/011${EXTEN:1}
exten = _9.,4,GotoIf($[ ${DIALSTATUS} != CONGESTION ${DIALSTATUS} != CHANUNAVAIL
]?6)
exten = _9.,5,Congestion
exten = _9.,6,Hangup
arkadi.
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