[asterisk-users] iaxtel
Is iaxtel still around? I was not able to go to www.iaxtel.com . did the address changed? ___ --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] IAXTEL??
Is IAXTEL still around? I needed to call Digium and figured I would set it up to save some miinutes when talking to them but I can't get it to register. -Kerry ___ --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] IAXTEL??
From: http://www.iaxtel.com/ The IAXTel Server is currently under maintenance. Some technical difficulties, such as connection timeouts, registration timeouts, and the inability to make phone calls may be experienced. Thank you for your patience. :( b -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Tuesday, January 03, 2006 5:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] IAXTEL?? Is IAXTEL still around? I needed to call Digium and figured I would set it up to save some miinutes when talking to them but I can't get it to register. -Kerry ___ --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] IAXTEL??
That message has been there for months. On 1/3/06, Bogdan Moldovan [EMAIL PROTECTED] wrote: From: http://www.iaxtel.com/ The IAXTel Server is currently under maintenance. Some technical difficulties, such as connection timeouts, registration timeouts, and the inability to make phone calls may be experienced. Thank you for your patience. :( b -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Tuesday, January 03, 2006 5:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] IAXTEL?? Is IAXTEL still around? I needed to call Digium and figured I would set it up to save some miinutes when talking to them but I can't get it to register. -Kerry ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] IAXTEL??
Yeah, saw that, and it had said that for like six months if I recall. You would figure that since Digium features IAXTEL phone numbers so prominently, that it would be a service that was actually capable of connecting to them. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bogdan Moldovan Sent: Tuesday, January 03, 2006 8:01 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IAXTEL?? From: http://www.iaxtel.com/ The IAXTel Server is currently under maintenance. Some technical difficulties, such as connection timeouts, registration timeouts, and the inability to make phone calls may be experienced. Thank you for your patience. :( b -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Tuesday, January 03, 2006 5:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] IAXTEL?? Is IAXTEL still around? I needed to call Digium and figured I would set it up to save some miinutes when talking to them but I can't get it to register. -Kerry ___ --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] IAXTEL??
I know, this is the sad part :( b -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: Tuesday, January 03, 2006 6:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXTEL?? That message has been there for months. On 1/3/06, Bogdan Moldovan [EMAIL PROTECTED] wrote: From: http://www.iaxtel.com/ The IAXTel Server is currently under maintenance. Some technical difficulties, such as connection timeouts, registration timeouts, and the inability to make phone calls may be experienced. Thank you for your patience. :( b -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Tuesday, January 03, 2006 5:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] IAXTEL?? Is IAXTEL still around? I needed to call Digium and figured I would set it up to save some miinutes when talking to them but I can't get it to register. -Kerry ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] IAXTEL??
Iaxtel has been down for some time now. But to get in contact with digium via your asterisk box all you need is to set this dialing rule up. exten = 500,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ;Call Digium exten = 500,2,Congestion Kerry Garrison wrote: Is IAXTEL still around? I needed to call Digium and figured I would set it up to save some miinutes when talking to them but I can't get it to register. -Kerry ___ --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] IAXTEL??
Is IAXTEL still around? I needed to call Digium and figured I would set it up to save some miinutes when talking to them but I can't get it to register. That hasn't worked for many many months. Much easier to reach digium by using the Demo that is/was installed in all asterisk installs. When the voice prompt indicates its connecting to a demonstation server at digium, it is a real * server that can connect you to tech support, etc, etc. Try it. ___ --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] IAXTEL??
Ariel Batista wrote: Iaxtel has been down for some time now. But to get in contact with digium via your asterisk box all you need is to set this dialing rule up. exten = 500,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ;Call Digium exten = 500,2,Congestion Cool, I didn't think of that. It has been a long time since I've installed Asterisk for the first time. I'll keep that in mind for next time. Hopefully there isn't one though. ;) Kevin ___ --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] IAXtel update!
Rich Adamson wrote: Any chance that we could get someone to implement the milliwatt generator and echo test number. Would be kind of handy for testing various items (eg, jitterbuffer). It's running CVS HEAD (which means it has the new jb since we didn't disable it, but then again it's all VOIP so the jb doesn't get enabled anyway), with Realtime for IAX2 friends and the experimental hashtable config parsing code. If you can email me or Russell with what you think should be enabled there we'll see what we can do. ___ 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] IAXtel update!
Hello Everyone! Over this weekend, we have updated IAXtel. Before the update, it was running at almost 100% cpu load at an idle state because of the massive amount of database transactions. We enabled realtime caching and the box immediately crashed. We were able to expose a serious bug related to realtime caching in chan_iax2. Kevin Fleming was able to fix this issue, and also added some experimental code to further enhance performace. As I write this message, Asterisk is using about 4 percent CPU load on IAXtel. We are hoping that it will become usable again. May all of your calls have full-duplex audio! -- Mark Spencer Russell ___ 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] IAXtel update!
Over this weekend, we have updated IAXtel. Before the update, it was running at almost 100% cpu load at an idle state because of the massive amount of database transactions. We enabled realtime caching and the box immediately crashed. We were able to expose a serious bug related to realtime caching in chan_iax2. Kevin Fleming was able to fix this issue, and also added some experimental code to further enhance performace. As I write this message, Asterisk is using about 4 percent CPU load on IAXtel. We are hoping that it will become usable again. May all of your calls have full-duplex audio! -- Mark Spencer Good. Which version of * is running on that system now? Any chance that we could get someone to implement the milliwatt generator and echo test number. Would be kind of handy for testing various items (eg, jitterbuffer). ___ 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] IAXTEl down
Figures... So... Everybody went to FWD :) ? It mostly works, does IAX, so, yeah. ___ 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] IAXTEl down
Is iaxtel down? Ive been getting this: May 21 19:23:42 NOTICE[29984]: chan_iax2.c:2782 auto_congest: Auto-congesting call due to slow response -- IAX2/Iaxtel-12 is circuit-busy -- Hungup 'IAX2/Iaxtel-12' is it down or am I doing something wrong? Its been doing that for months. No one is actually maintaining the site. ___ 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] IAXTEl down
Figures... So... Everybody went to FWD :) ? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Domingo, 22 de Mayo de 2005 08:23 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] IAXTEl down | | Is iaxtel down? | | Ive been getting this: | May 21 19:23:42 NOTICE[29984]: chan_iax2.c:2782 auto_congest: | Auto-congesting call due to slow response | -- IAX2/Iaxtel-12 is circuit-busy | -- Hungup 'IAX2/Iaxtel-12' | | is it down or am I doing something wrong? | |Its been doing that for months. No one is actually maintaining |the site. | | |___ |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] IAXTEl down
Is iaxtel down? Ive been getting this: May 21 19:23:42 NOTICE[29984]: chan_iax2.c:2782 auto_congest: Auto-congesting call due to slow response -- IAX2/Iaxtel-12 is circuit-busy -- Hungup 'IAX2/Iaxtel-12' is it down or am I doing something wrong? ___ 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] Iaxtel
Is iaxtel down? Im trying to dial Echo test: 1700613 and I get a busy signal... Also, is the gw to FWD users down too? ___ 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] IAXTel problems
Duane wrote: Marco Supino wrote: Hi, I tried to add the IAXTel config to my asterisk, so i can dial free numbers inside the US from my SIP softphone (X-lite), everything seems to be working, but the sound quality is terrible, the other side sounds like a digitized voice, and the voice is cut, i cant hear a full word, You could always just use e164 for toll free numbers, we have sip urls for about 11 countries and international toll free in our zone, and I've never had an issue with call quality to the US toll free numbers... How can you use that? bye Ronald -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you 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
[Asterisk-Users] IAXTEL Passord
Hi, I should be missing something. The password that go with my IAXTEL registration include an @. It seem that I can't use it because it thing that the second part of the password is the host name. I just don't know how to solve this one. regards, JYL ___ 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] IAXTel problems
Hi, I tried to add the IAXTel config to my asterisk, so i can dial free numbers inside the US from my SIP softphone (X-lite), everything seems to be working, but the sound quality is terrible, the other side sounds like a digitized voice, and the voice is cut, i cant hear a full word, I tried using FWD IAX interface, and no problem there, it works great. Now, although this is in a testing phase, i wanted to know if i am missing something, or IAXTel is just problematic . I am dialing from Israel, over a E1 line, dont know exactly how much of my E1 reaches the US, but should be sufficent for one session (for which FWD works fine with) Any help appriciated. Marco. ___ 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] IAXTel problems
Marco Supino wrote: Hi, I tried to add the IAXTel config to my asterisk, so i can dial free numbers inside the US from my SIP softphone (X-lite), everything seems to be working, but the sound quality is terrible, the other side sounds like a digitized voice, and the voice is cut, i cant hear a full word, You could always just use e164 for toll free numbers, we have sip urls for about 11 countries and international toll free in our zone, and I've never had an issue with call quality to the US toll free numbers... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ 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] IAXTEL is dead/dying?
As someone that's just recently setup an * server I agree. I thought about setting up an Iaxtel account as well but couldn't see the point in it because I had setup FWD for testing. I continue to use FWD for all my toll free calls and the quality is just awesome. I can't see how Iaxtel would provide any additional benefit. Perhaps the time for Iaxtel has come and gone. There are plenty of IAX2 providers these days, * has become quite popular, so the need for a separate telecom network doesn't make a whole lot of sense; not that FWD isn't separate, it's just more popular IMHO. -mark On Jan 21, 2005, at 6:12 PM, Michael Graves wrote: Yeah, FWD has been pretty good about their beta of the IAX2 support. My * server has been on it for 6 months without too much trouble. I even use it to bridge out to Signate.co.uk where my boss has an account. It was crystal clear last night from Houston TX to Cambridge UK. Dead reliable. I'm dropping my IAXTel registration when next I get around to such things. -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXTEL is dead/dying?
I didn't get any response at all to my last request for status on IAXTEL. So, when this happens, I attribute it to one of a number of things: 1. No-one knows. 2. No-one cares. 3. Everyone knows, but are too busy to reply. At any rate, my investigative side kicks in and I began searching thru the digest's I've gotten, looking for references to IAXTEL. Mostly it is mentioned in snippets of extensions.conf files submitted, or in people's sigs, but I've copy/pasted below a few relevant snippets I've spotted on this mailing list going backwards in time... Based on what I see, just before and at christmas, I don't see complaints, and people are suggesting to use it. But, after Christmas, the tone changes, and people are now advising against using IAXTEL. So, is there anybody out there at Digium, who can give the party line as to what the status of IAXTEL is, and if this is temporary? It might be best to make an announcement, as I'm sure folks will be scratching their heads and asking on the list what they are doing wrong when they can't fire up their IAXTEL connection. The impression I have at the moment is that IAXTEL is overloaded and can't keep up with registration traffic, let alone phone call bandwidth. Maybe it's just me and I've got a lousy internet path to it. But FWD is working very well, I just got my first call (besides me testing it) today! And sound quality was fine. I was looking forward to boasting my cool 1-700 number, but it looks not to be... And, lest anyone mistake this post as derogatory against IAXTEL, let me clearly state that I appreciate the efforts of Digium, and VoicePulse, to provide this free service, and also acknowledge the obvious benefits the entire asterisk community has reaped from its existence! murf --- From: ... On Behalf Of Christopher Dobbs Sent: Saturday, January 19, 2002 8:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXTEL errors ! Use FWDNET.NET. It is far better on call quality!! -- Christopher Dobbs Manjit Riat wrote: I am testing IAXTEL and routing 800 number to them.. Sometimes the call goes through and the other times it get the following error. WARNING[20502]: chan_iax2.c:1477 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/iaxtel/3 (type = 6, subclass = 9, ts=631, seqno=1) PS. It seems to me iaxtel has a problem with connection today, can anybody confirm it? I just tried to place a call via iaxtel and watched the packets with ethereal. The iaxtel server is very very slow to respond to _any_ packet, indicating its not feeling very well. Could not complete the call at all, and 'iax2 show registry' indicates instability as well. What is the best codex for iaxtel? When I set in iax.conf bandwidth=high disallow=all allow=ulaw The call will not go through, if I set allow=all it sets the format to ADPCM and the first 15sec. or so the voice is choppy, it is hard to understand anything. Is it reliable/practical to terminate 1800 calls via iaxtel? IAXtel only officially supports the GSM codec. Use that codec and no other codec. I've tried gsm but the call doesn't go through. Looks like iaxtel is down again. Just tried dialing my number and nothing happens. Not uncommon. --- I've tried gsm but the call doesn't go through. bandwidth=high could be screwing it up. Post the CLI output of the failed call. Executing Dial(SIP/11-0b9e, IAX2/joseph:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called joseph:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Hungup 'IAX2[iaxtel]/3' == No one is available to answer at this time That is all I see when I try to call iaxtel. --- I have in iax.conf register = name:[EMAIL PROTECTED] but I can not make a call, it hangs up on me. How can I check if I'm registered with iaxtel? What do I have to have in iax.conf in order to register? -- #Joseph --- 28 Dec 2004: Hi List, While trying to calling Toll Free numbers using IAXTEL, the call connects, Iear about 2 seconds of voice and then the voice drops off and I get the following error message which keeps scrolling across my console screen. WARNING[-167797840]: chan_iax2.c:5967 socket_read: Received mini frame before first full voice frame Asterisk shows that the format of this call is GSM. However if I make a call to the same TOLL Free number using the FWD network from the asterisk box then the call goes through fine
Re: [Asterisk-Users] IAXTEL is dead/dying?
Mark had made a post recently (last week or so maybe) -- could have been in IRC too... (it starts to blur together) that he was aware of the IAXTEL problems and that they were working on the issues. Details are hazy... But then I drink alot too, so everything is hazy... (that's the point) Tom Walsh -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 1/17/2005 ___ 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] IAXTEL is dead/dying?
On Fri, 21 Jan 2005 11:26:12 -0700, Steve Murphy [EMAIL PROTECTED] wrote: I didn't get any response at all to my last request for status on IAXTEL. So, when this happens, I attribute it to one of a number of things: 1. No-one knows. 2. No-one cares. 3. Everyone knows, but are too busy to reply. I didn't happen to see that message you sent before, but even if I did, I was probably too busy to reply :) Anyways, I used to use IAXTEL with great success. However, for almost nearly a year or so, I've been having significant latency problems with it. My qualify times were anywhere between 1500 and 5000ms, and since I wasn't really using it that much, or receiving many calls on it, I basically just dropped it. I know a few others who have also done the same. Basically if you need a free VoIP service, I recommend using FreeWorldDialup with either SIP of IAX2 (I use IAX2) as it has what I believe to be the largest user base, so might as well just use that. Plus they have break outs to lots of the other free services (including IAXtel). Thanks, Leif Madsen. http://www.leifmadsen.com FWD: 18924 :) ___ 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] IAXTEL is dead/dying?
On Fri, 21 Jan 2005 13:28:46 -0600, Leif Madsen wrote: On Fri, 21 Jan 2005 11:26:12 -0700, Steve Murphy [EMAIL PROTECTED] wrote: I didn't get any response at all to my last request for status on IAXTEL. So, when this happens, I attribute it to one of a number of things: 1. No-one knows. 2. No-one cares. 3. Everyone knows, but are too busy to reply. I didn't happen to see that message you sent before, but even if I did, I was probably too busy to reply :) Anyways, I used to use IAXTEL with great success. However, for almost nearly a year or so, I've been having significant latency problems with it. My qualify times were anywhere between 1500 and 5000ms, and since I wasn't really using it that much, or receiving many calls on it, I basically just dropped it. I know a few others who have also done the same. Basically if you need a free VoIP service, I recommend using FreeWorldDialup with either SIP of IAX2 (I use IAX2) as it has what I believe to be the largest user base, so might as well just use that. Plus they have break outs to lots of the other free services (including IAXtel). Thanks, Leif Madsen. http://www.leifmadsen.com FWD: 18924 :) Yeah, FWD has been pretty good about their beta of the IAX2 support. My * server has been on it for 6 months without too much trouble. I even use it to bridge out to Signate.co.uk where my boss has an account. It was crystal clear last night from Houston TX to Cambridge UK. Dead reliable. I'm dropping my IAXTel registration when next I get around to such things. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] IAXTEL errors !
I am testing IAXTEL and routing 800 number to them.. Sometimes the call goes through and the other times it get the following error. WARNING[20502]: chan_iax2.c:1477 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/iaxtel/3 (type = 6, subclass = 9, ts=631, seqno=1) ___ 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] IAXTEL errors !
Use FWDNET.NET. It is far better on call quality!! -- Christopher Dobbs Manjit Riat wrote: I am testing IAXTEL and routing 800 number to them.. Sometimes the call goes through and the other times it get the following error. WARNING[20502]: chan_iax2.c:1477 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/iaxtel/3 (type = 6, subclass = 9, ts=631, seqno=1) ___ 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 No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.1 - Release Date: 1/19/2005 ___ 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] IAXTEL errors !
Christopher, Any idea what causing Max retries exceeded to happen? Regards, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Dobbs Sent: Saturday, January 19, 2002 8:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXTEL errors ! Use FWDNET.NET. It is far better on call quality!! -- Christopher Dobbs Manjit Riat wrote: I am testing IAXTEL and routing 800 number to them.. Sometimes the call goes through and the other times it get the following error. WARNING[20502]: chan_iax2.c:1477 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/iaxtel/3 (type = 6, subclass = 9, ts=631, seqno=1) ___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] iaxtel - best codec
What is the best codex for iaxtel? When I set in iax.conf bandwidth=high disallow=all allow=ulaw The call will not go through, if I set allow=all it sets the format to ADPCM and the first 15sec. or so the voice is choppy, it is hard to understand anything. Is it reliable/practical to terminate 1800 calls via iaxtel? -- #Joseph ___ 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] iaxtel - best codec
Joseph wrote: What is the best codex for iaxtel? When I set in iax.conf bandwidth=high disallow=all allow=ulaw The call will not go through, if I set allow=all it sets the format to ADPCM and the first 15sec. or so the voice is choppy, it is hard to understand anything. Is it reliable/practical to terminate 1800 calls via iaxtel? IAXtel only officially supports the GSM codec. Use that codec and no other 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] iaxtel - best codec
On Mon, 2005-01-17 at 12:20 -0600, Eric Wieling aka ManxPower wrote: Joseph wrote: What is the best codex for iaxtel? When I set in iax.conf bandwidth=high disallow=all allow=ulaw The call will not go through, if I set allow=all it sets the format to ADPCM and the first 15sec. or so the voice is choppy, it is hard to understand anything. Is it reliable/practical to terminate 1800 calls via iaxtel? IAXtel only officially supports the GSM codec. Use that codec and no other codec. I've tried gsm but the call doesn't go through. -- #Joseph ___ 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] iaxtel - best codec
Joseph wrote: On Mon, 2005-01-17 at 12:20 -0600, Eric Wieling aka ManxPower wrote: Joseph wrote: What is the best codex for iaxtel? When I set in iax.conf bandwidth=high disallow=all allow=ulaw The call will not go through, if I set allow=all it sets the format to ADPCM and the first 15sec. or so the voice is choppy, it is hard to understand anything. Is it reliable/practical to terminate 1800 calls via iaxtel? IAXtel only officially supports the GSM codec. Use that codec and no other codec. I've tried gsm but the call doesn't go through. bandwidth=high could be screwing it up. Post the CLI output of the failed call. ___ 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] iaxtel - best codec
What is the best codex for iaxtel? When I set in iax.conf bandwidth=high disallow=all allow=ulaw The call will not go through, if I set allow=all it sets the format to ADPCM and the first 15sec. or so the voice is choppy, it is hard to understand anything. Is it reliable/practical to terminate 1800 calls via iaxtel? IAXtel only officially supports the GSM codec. Use that codec and no other codec. I've tried gsm but the call doesn't go through. Looks like iaxtel is down again. Just tried dialing my number and nothing happens. Not uncommon. ___ 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] iaxtel - best codec
I've tried gsm but the call doesn't go through. bandwidth=high could be screwing it up. Post the CLI output of the failed call. Executing Dial(SIP/11-0b9e, IAX2/joseph:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called joseph:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Hungup 'IAX2[iaxtel]/3' == No one is available to answer at this time That is all I see when I try to call iaxtel. -- #Joseph ___ 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] iaxtel - best codec
Iaxtel only supports gsm. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, January 17, 2005 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] iaxtel - best codec What is the best codex for iaxtel? When I set in iax.conf bandwidth=high disallow=all allow=ulaw The call will not go through, if I set allow=all it sets the format to ADPCM and the first 15sec. or so the voice is choppy, it is hard to understand anything. Is it reliable/practical to terminate 1800 calls via iaxtel? IAXtel only officially supports the GSM codec. Use that codec and no other codec. I've tried gsm but the call doesn't go through. Looks like iaxtel is down again. Just tried dialing my number and nothing happens. Not uncommon. ___ 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] iaxtel - -- Format for call is ADPCM
When I try to call iaxtel it goes to codec ADPCM even though I have define in iax.conf gsm Call accepted by 69.73.19.178 (format ADPCM) -- Format for call is ADPCM My settings: [general] port=4569 register = :[EMAIL PROTECTED] bandwidth=high jitterbuffer=no tos=lowdelay [voipjet] type=peer host= xxx.xxx.xxx.xx secret= xxx auth=md5 notransfer=yes context=incoming disallow=all ; Prevent all codecs... allow = ulaw ; ...except G.711 ulaw [iaxtel] type=friend host=iaxtel.com secret= auth=rsa context=incoming inkeys=iaxtel disallow=all allow=gsm Why is it switching me to Codec: ADPCM? PS. It seems to me iaxtel has a problem with connection today, can anybody confirm it? -- #Joseph ___ 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] iaxtel - -- Format for call is ADPCM
Why is it switching me to Codec: ADPCM? PS. It seems to me iaxtel has a problem with connection today, can anybody confirm it? I just tried to place a call via iaxtel and watched the packets with ethereal. The iaxtel server is very very slow to respond to _any_ packet, indicating its not feeling very well. Could not complete the call at all, and 'iax2 show registry' indicates instability as well. ___ 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] iaxtel - -- Format for call is ADPCM
There was a bug with codecs for a very long time with Asterisk. In [general] remove the bandwidth= line (all it does is allow specific codecs) and disallow=all and allow= for eac codec you want. Joseph wrote: When I try to call iaxtel it goes to codec ADPCM even though I have define in iax.conf gsm Call accepted by 69.73.19.178 (format ADPCM) -- Format for call is ADPCM My settings: [general] port=4569 register = :[EMAIL PROTECTED] bandwidth=high jitterbuffer=no tos=lowdelay [voipjet] type=peer host= xxx.xxx.xxx.xx secret= xxx auth=md5 notransfer=yes context=incoming disallow=all ; Prevent all codecs... allow = ulaw ; ...except G.711 ulaw [iaxtel] type=friend host=iaxtel.com secret= auth=rsa context=incoming inkeys=iaxtel disallow=all allow=gsm Why is it switching me to Codec: ADPCM? PS. It seems to me iaxtel has a problem with connection today, can anybody confirm it? ___ 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] Iaxtel directory
I cannot find the directory of 1700 numbers (iaxtel), nor where I can edit my own entry. Can anybody publish the link, please? bye Ronald ___ 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] iaxtel
I have registered to iaxtel.com! I forgot my iaxtel.com number, and cannot find the white pages of it. As I see, you should setup in extensions.conf all 1700*,1888*, 1877*, 1866* and 1800* for this connection. please correct me: 1700* is only other iaxtel.com users 1888* are tollfree numbers in USA 1877* are 1866* are 1800* are tollfree numbers in USA thanks for your help bye Ronald ___ 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] iaxtel
1800,1866,1877,1888 are all toll free numbers in the us ___ 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] IAXTEL Configuration
Seems I was looking in all the wrong places.
The problem was that I was stripping the leading '1' off of the outbound
IAXTEL phone number.
exten = _91700NXX,1,Dial(${IAXNET}/${EXTEN:[EMAIL PROTECTED]) will not work
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anders F
Eriksson
Sent: Tuesday, December 21, 2004 7:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] IAXTEL Configuration
Hi,
I think you should remove the [iaxtel_out] from iax.conf
This is a snip from mine iax.conf:
[general]
register = user:[EMAIL PROTECTED]
[iaxtel]
type=user
context=incoming
auth=rsa
inkeys=iaxtel
You then can modify extensions.conf to handle outgoing calls. See
http://www.iaxtel.com/setup.html (which is where I got my settings).
/Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: den 21 december 2004 22:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAXTEL Configuration
I signed up for an IAXTEL account and have been trying,
unsuccessfully, to get it working. In IAX.CONF I have:
[iaxtel_out]
type=peer
host=iaxtel.com
username=USERNAME
secret=SECRET
auth=rsa
inkeys=iaxtel
[iaxtel]
type=friend
context=incoming
host=iaxtel.com
auth=rsa
inkeys=iaxtel
However, when I start Asterisk, I get the following warning:
[chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
== Manager registered action IAXpeers
== Parsing '/etc/asterisk/iax.conf': Found Dec 21 15:44:04
WARNING[24873]: chan_iax2.c:6602 build_user: Cannot allow unknown
format 'iaxtel.com'
Dec 21 15:44:04 WARNING[24873]: chan_iax2.c:6497 build_peer:
Cannot allow unknown format 'iaxtel.com'
For some reason, it does not like the host= lines. I've replaced
'iaxtel.com' with their IP, but that gives the same error.
Please assist. Thanks,
Adam
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[Asterisk-Users] IAXTEL Configuration
I signed up for an IAXTEL account and have been trying, unsuccessfully, to get it working. In IAX.CONF I have: [iaxtel_out]type=peerhost=iaxtel.comusername=USERNAMEsecret=SECRETauth=rsainkeys=iaxtel [iaxtel]type=friendcontext=incominghost=iaxtel.comauth=rsainkeys=iaxtel However, when I start Asterisk, I get the following warning: [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': FoundDec 21 15:44:04 WARNING[24873]: chan_iax2.c:6602 build_user: Cannot allow unknown format 'iaxtel.com'Dec 21 15:44:04 WARNING[24873]: chan_iax2.c:6497 build_peer: Cannot allow unknown format 'iaxtel.com' For some reason, it does not like the "host=" lines. I've replaced 'iaxtel.com' with their IP, but that gives the same error. Please assist. Thanks, Adam 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] IAXTEL Configuration
Hi, I think you should remove the [iaxtel_out] from iax.conf This is a snip from mine iax.conf: [general] register = user:[EMAIL PROTECTED] [iaxtel] type=user context=incoming auth=rsa inkeys=iaxtel You then can modify extensions.conf to handle outgoing calls. See http://www.iaxtel.com/setup.html (which is where I got my settings). /Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: den 21 december 2004 22:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAXTEL Configuration I signed up for an IAXTEL account and have been trying, unsuccessfully, to get it working. In IAX.CONF I have: [iaxtel_out] type=peer host=iaxtel.com username=USERNAME secret=SECRET auth=rsa inkeys=iaxtel [iaxtel] type=friend context=incoming host=iaxtel.com auth=rsa inkeys=iaxtel However, when I start Asterisk, I get the following warning: [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found Dec 21 15:44:04 WARNING[24873]: chan_iax2.c:6602 build_user: Cannot allow unknown format 'iaxtel.com' Dec 21 15:44:04 WARNING[24873]: chan_iax2.c:6497 build_peer: Cannot allow unknown format 'iaxtel.com' For some reason, it does not like the host= lines. I've replaced 'iaxtel.com' with their IP, but that gives the same error. Please assist. Thanks, Adam 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] IAXTel problems
Hello, I'm having problems connecting to other * boxes through IAXTel. I've seen this addressed in the list archives, and other places on the web, but haven't seen that anyone has come up with a solution. I'm dialing in to my Asterisk server using DISA, authenticating OK, then attempting to dial out and keep getting IAX2/69.73.19.178:4569/8 stopped sounds and it hangs up. I've tried switching codecs as I saw someone suggest, but get the same result. This happens with 1-700 numbers as well as 18XX numbers that I know work properly, so I don't think it's a misconfiguration on the receiving server's side as some have suggested. I've found people posting about this several times over the past year and a half, so I imagine the problem is pretty common, and is something misconfigured on my side, or some kind of common conflict. Any ideas? PF ___ 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] IAXTel problems
i had this problem last night. sometimes it would work find and then i would get errors or timeouts??? - hcir On Oct 22, 2004, at 9:07 AM, pixelFiend wrote: Hello, I'm having problems connecting to other * boxes through IAXTel. I've seen this addressed in the list archives, and other places on the web, but haven't seen that anyone has come up with a solution. I'm dialing in to my Asterisk server using DISA, authenticating OK, then attempting to dial out and keep getting IAX2/69.73.19.178:4569/8 stopped sounds and it hangs up. I've tried switching codecs as I saw someone suggest, but get the same result. This happens with 1-700 numbers as well as 18XX numbers that I know work properly, so I don't think it's a misconfiguration on the receiving server's side as some have suggested. I've found people posting about this several times over the past year and a half, so I imagine the problem is pretty common, and is something misconfigured on my side, or some kind of common conflict. Any ideas? ___ 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] IAXTel and Telesthetic
I'm trying to run some inbound test to my Asterisk box using Telesthetic's gateway in MI to my GNU/IAXtel account. Am I missing something? I set up my user account on the GNUPhonne site, configured Asterisk to talk to IAXTel. * registers fine. In fact I can make calls to other test users. I haven't tried having someone call my number. When I call into Telesthetic's exchange it answers, tells me "transferring to VOIP", I enter my number 1-700-, 1 second later I'm back at the VOIP prompt. If I leave of the 700 I get a response stating the user is offline. I tried a few other numbers that people had publish and the results were similar. thanks in advance, Dan___ 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] iaxtel and jitterbuffer
On Sun, 29 Aug 2004, Kris Boutilier wrote: Is timestamp information calculated purely from the relative timestamps of each frame of the current incoming stream or is there some degree of RTC synchronization expected between the two endpoints? No sync is needed; its all relative. Similarly, are jitter calculations made seperately for each discrete channel (ie. the IAX level) or are they based on an aggregate of all channels between each pair of two endpoints (ie. the TCP/IP level)? De-jtter is done for each call independently. Steve ___ 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] iaxtel and jitterbuffer
On Sat, 28 Aug 2004, Michael George wrote: So even with X11 eliminated the sound is still bad to Digium. I tried another's 1700 number, and it sounded the same, so it's not something unique to digium and me. Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to work with my ISP only giving me 1/2 duplex service? If you think that the jitter buffer isn't working right and should fix this, then please capture debug from the buffer and send over to me. To do that, in /etc/asterisk/logger.conf edit the debug line to be: debug = notice,warning,error,debug,verbose Then run asterisk like so: /usr/sbin/asterisk -vv -g -dd -c Then go iax2 debug at the CLI prompt. Do a test call, then send me the resulting /var/log/asterisk/debug file. THanks, Steve ___ 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] iaxtel and jitterbuffer
On Sat, 28 Aug 2004, Andrew Kohlsmith wrote: Please note that it seems impossible to disable jitter buffer between 20040806 CVS HEAD endpoints. The jitterbuffer numbers in iax2 show channels look live. The numbers look right (jitbuf 0ms) between 20040806 and RC1 (Nufone). I haven't upgraded since then. The numbers get reported still in the older version, but the buffer IS turned off. Steve ___ 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] iaxtel and jitterbuffer
On Sunday 29 August 2004 02:06, [EMAIL PROTECTED] wrote: On Sat, 28 Aug 2004, Andrew Kohlsmith wrote: Please note that it seems impossible to disable jitter buffer between 20040806 CVS HEAD endpoints. The jitterbuffer numbers in iax2 show channels look live. The numbers look right (jitbuf 0ms) between 20040806 and RC1 (Nufone). I haven't upgraded since then. The numbers get reported still in the older version, but the buffer IS turned off. Ok so the disparity between iax2 show channels between two 20040806 (looks live) and 20040806 and RC1 (shows 0s) is expected? Just making sure, as between the two 'new' versions it is live, but between the new and old, it looks dead, whereas your reply said the numbers are still reported in the older version and that's not what I'm seeing. :-) -A. ___ 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] iaxtel and jitterbuffer
On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote: If you think that the jitter buffer isn't working right and should fix this, then please capture debug from the buffer and send over to me. To do that, in /etc/asterisk/logger.conf edit the debug line to be: debug = notice,warning,error,debug,verbose Then run asterisk like so: /usr/sbin/asterisk -vv -g -dd -c Then go iax2 debug at the CLI prompt. Do a test call, then send me the resulting /var/log/asterisk/debug file. Is there any way to do this 'live'? I get it intermittently and capturing debug for days before the problem is manifest is probably not the best way to do it. I've tried leaving the debug line in and not invoking any kind of -d in the asterisk startup but the debug log still grows. I can't comment out the debug line in logger.conf because a logger reload or reload will NOT create the debug file, only a restart will. Ideally some way to create the debug file but write very litte to it until I connect with asterisk -rc or something would be best I imagine. Also, is are logs of problem conversations already in progress any use to you? You nailed down the dead audio after 65535ms problem but every now and again (very very rare) we will have a conversation where the incoming audio goes totally dead for about 2-4 seconds and then continues just fine. This occurs usually several minutes into the conversation, and I've never seen it occur twice in a conversation. Obviously this is next to impossible to catch. :-( I haven't heard a complaint about it since turning off jitter buffer to nufone. As always, thank you for your knowledge and input. -A. ___ 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] iaxtel and jitterbuffer
On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote: If you think that the jitter buffer isn't working right and should fix this, then please capture debug from the buffer and send over to me. I notice that the timing measurements are still showing wild values at times - here is a partial grab of an iax2 show channels: Lag Jitter JitBuf Format 00020ms 6291456ms ms ALAW 00012ms 6291440ms ms ALAW 00017ms 0004ms ms ALAW 00012ms 286523393ms ms ALAW 00012ms 0025ms ms ALAW -978714621ms 6293280ms ms ALAW ___ 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] iaxtel and jitterbuffer
Those wild times especially occur before any audio is sent. (e.g. while ringing or pre ringing). At 17:10 29/08/2004, you wrote: On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote: If you think that the jitter buffer isn't working right and should fix this, then please capture debug from the buffer and send over to me. I notice that the timing measurements are still showing wild values at times - here is a partial grab of an iax2 show channels: Lag Jitter JitBuf Format 00020ms 6291456ms ms ALAW 00012ms 6291440ms ms ALAW 00017ms 0004ms ms ALAW 00012ms 286523393ms ms ALAW 00012ms 0025ms ms ALAW -978714621ms 6293280ms ms ALAW ___ 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] iaxtel and jitterbuffer
On Sat, Aug 28, 2004 at 11:31:48PM -0400, Andrew Kohlsmith wrote: On Saturday 28 August 2004 23:01, Michael George wrote: It's a PII 266 (okay, not the fatest system) with 192MB RAM. X is not running and the Framebuffer has been turned off in /boot/grum/menu.lst. I have disabled all the servers except for sshd. I have the latest source from CVS HEAD as of about 30min ago. Should be fine. I ran * on a P90 for a while; it did everything I needed except iLBC. :-) Okay, that's a good assurance. Unfortunately, I have discovered that either the HDD or the ide controller in that system is bad because it cannot stay up overnight. When I stress it with a YaST update, it will die much more reliably. Until I can address that issue, I will have to work on my main system. I'll just have to take it down to init 3 and stop many of the other server processes that will still be running. There is no Zap card in this sytem. The only phone on it is a SIP phone. With it I dial in to digium's 1-700 number. The audio is better, but still choppy and unacceptible. Is your SIP phone doing any kind of silence suppression? It must be turned off because asterisk takes its timing from the RTP stream and if the phone's not transmitting frames continuously you'll get shitty audio. Good suggestion and I have double checked it. I am and was not doing that. I think I'd read about it in a Granstream-* page Note that latest CVS HEAD looks like they're making provisions for self-timing but without a stable clock source it's unlikely to help you. There are ztdummy modules which use the RTC or certain brands of USB controller to provide adequate timing but ideally you want some kind of Zaptel hardware in there providing a 1000Hz interrupt. Hmm, I thought that the timing source was only needed for trunking. I don't have on on the little box, but I do have a TDM400 (which seems to have faults, also, but Digium suggested moving the FXO to socket 4, we'll see if that helps) in the main box, so that should be all set for a timing source. Also -- make sure your uplink is acceptable. First test: make sure there is nothing plugged into your upstream except for your asterisk box and the phone. Some routers are known to play silly bugger with your packets which naturally wreaks havoc with asterisk. :-) The only things on the net when I run the next test will be my main server. Since I have to test on that with X turned off, I don't even need the SIP phone active. In case it might be relevant (there are SO many pieces to this puzzle that I want to mention all I can think of in case they ring a trouble-bell in someone's mind...) my router is a Netgear FVS318 acting as a NAT to my ISP. So even with X11 eliminated the sound is still bad to Digium. I tried another's 1700 number, and it sounded the same, so it's not something unique to digium and me. Perhaps something to do with your upstream or connection to IAXtel. That's why I'm recommending having nothing but asterisk and the phone on the connection, at least until we nail down what the poor audio's being caused by. That's possible. I've checked with my ISP and he said that the connection is surely half-duplex, but you say that you have 1/2 also and it works fine for you, so that's not it. I'm also inquiring about other filters they might have in place. I've heard them mention before that they had some cool router software that could detect traffic patterns usually associated with software and music piracy and then throttle that traffic into a small part of The Pipe. I haven't yet heard back, and I'm hoping that isn't the case. However, if it *is*, a VPN between offices might help. IAXtel would be shot, though. Hoever, if that *is* the case, I can probably convince them to tell their software to leave me alone on a couple specified ports. Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to work with my ISP only giving me 1/2 duplex service? It has nothing to do with IAX or GSM. Stop blaming them. My upstream is half duplex as well (pretty much anyone on DSL or cable is on a half duplex connection whether they realize it or not). There are many, many people using asterisk every day for long distance and in environments where audio quality is crucial. Let's stop blaming asterisk and take a good hard look at what's happenning, shall we? My apologies. I'm not trying to blame anyone, I love * and except for a couple glitches that we're working on (with all your gracious help), I'm very impressed. My one glitch may be with the hardware, so that's a separate issue, but the other is trying to figure out this issue with IAX/GSM. When I ask about sensitivity, I don't mean to be accusatory. IAX is open and freely available and GSM is freely usable, and I'm glad. Sometimes OSS has its limitations and I am willing to work with them. So I do not intend any condescention(sp?),
Re: [Asterisk-Users] iaxtel and jitterbuffer
At 17:10 29/08/2004, you wrote: I notice that the timing measurements are still showing wild values at times - here is a partial grab of an iax2 show channels: Lag Jitter JitBuf Format 00020ms 6291456ms ms ALAW 00012ms 6291440ms ms ALAW 00017ms 0004ms ms ALAW 00012ms 286523393ms ms ALAW 00012ms 0025ms ms ALAW -978714621ms 6293280ms ms ALAW Those wild times especially occur before any audio is sent. (e.g. while ringing or pre ringing). That maybe true, but the examples above appeared to be established calls! Linus ___ 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] iaxtel and jitterbuffer
On Sun, Aug 29, 2004 at 07:59:20AM +0200, [EMAIL PROTECTED] wrote: On Sat, 28 Aug 2004, Michael George wrote: So even with X11 eliminated the sound is still bad to Digium. I tried another's 1700 number, and it sounded the same, so it's not something unique to digium and me. Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to work with my ISP only giving me 1/2 duplex service? If you think that the jitter buffer isn't working right and should fix this, then please capture debug from the buffer and send over to me. I'm not sure what the problem is. What I am hearing does sound like the descriptions I've read w.r.t. the jitter buffer, but making jitter buffer changes haven't really changed the effect. That gives 2 possibilities: 1. That the jitter buffer isn't working and it *should* fix the problem. 2. That the problem is completely independent of the JB so there is nothing the JB can do to fix it. To do that, in /etc/asterisk/logger.conf edit the debug line to be: debug = notice,warning,error,debug,verbose Then run asterisk like so: /usr/sbin/asterisk -vv -g -dd -c Then go iax2 debug at the CLI prompt. Do a test call, then send me the resulting /var/log/asterisk/debug file. I will do that. Hopefully that will help us isolate the problem and perhaps eliminate the jitterbuffer from the equasion. :) I will try to run this test today and report back my findings. Also, on Thursday I will be going into the main office. I will take my little * box and try the IAXtel test there. That should help determine if it's my local office net connection that is the problem. Thank you! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] iaxtel and jitterbuffer
On Sun, 29 Aug 2004, Andrew Kohlsmith wrote: Also, is are logs of problem conversations already in progress any use to you? You nailed down the dead audio after 65535ms problem but every now and again (very very rare) we will have a conversation where the incoming audio goes totally dead for about 2-4 seconds and then continues just fine. This occurs usually several minutes into the conversation, and I've never seen it occur twice in a conversation. Logs of parts of a call are fine. The jitter buffer makes all its decisions about dejittering based on the timestamps of incoming frames. There a fundamental expectation that the sending side is correctly stamping each frame - 20msec, 40msec etc etc. The problem is that the sending side doesn't always do that. Sometimes for one reason or another the stamps jump. The receiver has no way of telling that the sender mangled the timestamps, and assumes that the packets with the new stamps have been delayed, or arrived early, or whatever. Either way, the jitter buffer does its thing and unknowingly makes things worse. Unfortunately, this is why you can still be better off without it - but the problem really needs to be fixed by fixing the timestamp generation on the sender. Steve ___ 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] iaxtel and jitterbuffer
On Sun, 29 Aug 2004, joachim wrote: Those wild times especially occur before any audio is sent. (e.g. while ringing or pre ringing). Yeah - because the sender does weird things to the timestamps it generates. This is the problem that needs to be resolved; the jitter buffer just shows up the issue. Steve ___ 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] iaxtel and jitterbuffer
On Sunday 29 August 2004 15:52, [EMAIL PROTECTED] wrote: The jitter buffer makes all its decisions about dejittering based on the timestamps of incoming frames. There a fundamental expectation that the sending side is correctly stamping each frame - 20msec, 40msec etc etc. Right, this makes sense. :-) The problem is that the sending side doesn't always do that. Sometimes for one reason or another the stamps jump. The receiver has no way of telling that the sender mangled the timestamps, and assumes that the packets with the new stamps have been delayed, or arrived early, or whatever. Either way, the jitter buffer does its thing and unknowingly makes things worse. Unfortunately, this is why you can still be better off without it - but the problem really needs to be fixed by fixing the timestamp generation on the sender. Hmm... I think next CVS update I'm gonna add a bit of code in chan_iax2 that tries to verify that timestamps aren't getting sent incorrectly. Fun fun fun. :-) -A. ___ 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] iaxtel and jitterbuffer
On Sun, 29 Aug 2004, Andrew Kohlsmith wrote: Hmm... I think next CVS update I'm gonna add a bit of code in chan_iax2 that tries to verify that timestamps aren't getting sent incorrectly. Fun fun fun. :-) Its not that the generation is broken. Its that various optimisations and things have been added over time. The result is that sometimes the source of the timestamps changes - and suddenly. Like - we're playing locally generated Playback() audio down the line, then the dialplan rings another IAX2/ address. Then the other end answers. First the timestamps come from the Playback, then the ring generator, then from the remote IAX2/ system... So the discontinuities get in. There is also effort in the sending IAX2code to lock the timestamps to exact intervals (20msec), but sometimes it gives up and lets it jump to get back into step... Steve ___ 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] iaxtel and jitterbuffer
On Sat, Aug 28, 2004 at 11:31:48PM -0400, Andrew Kohlsmith wrote: On Saturday 28 August 2004 23:01, Michael George wrote: It has nothing to do with IAX or GSM. Stop blaming them. My upstream is half duplex as well (pretty much anyone on DSL or cable is on a half duplex connection whether they realize it or not). There are many, many people using asterisk every day for long distance and in environments where audio quality is crucial. Let's stop blaming asterisk and take a good hard look at what's happenning, shall we? Someone suggested that perhaps the machine is too slow. If someone who uses IAX2 between offices wouldn't mind, could you please indicate how heavy a system you are using for Zap -- IAX/GSM -- VOIP. Perhaps I am underestimating the HP required for the voice coding... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] iaxtel and jitterbuffer
Is timestamp information calculated purely from the relative timestamps of
each frame of the current incoming stream or is there some degree of RTC
synchronization expected between the two endpoints?
Similarly, are jitter calculations made seperately for each discrete channel
(ie. the IAX level) or are they based on an aggregate of all channels
between each pair of two endpoints (ie. the TCP/IP level)?
k.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: August 29, 2004 12:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iaxtel and jitterbuffer
{clip}
The jitter buffer makes all its decisions about dejittering based on the
timestamps of incoming frames. There a fundamental expectation that the
sending side is correctly stamping each frame - 20msec, 40msec etc etc.
The problem is that the sending side doesn't always do that. Sometimes
for one reason or another the stamps jump. The receiver has no way of
telling that the sender mangled the timestamps, and assumes that the
packets with the new stamps have been delayed, or arrived early, or
whatever. Either way, the jitter buffer does its thing and unknowingly
makes things worse.
Unfortunately, this is why you can still be better off without it - but
the problem really needs to be fixed by fixing the timestamp generation on
the sender.
Steve
___
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Re: [Asterisk-Users] iaxtel and jitterbuffer
Does this also effect 1.0-RC2? I am having a similar issue at a
customer site over a frame relay network that is having occasional
choppy sound over a fairly open line, with the jitter buffer enabled,
as well as trunk=yes enabled.
Thanks!
Brian
On Fri, 27 Aug 2004 12:47:05 -0700, Kris Boutilier
[EMAIL PROTECTED] wrote:
Had this problem earlier this week - ensure 'trunk=no' in iax.conf if you're
using fairly current CVS code. There is something not right w/the trunking
that causes choppy sound. See the wiki for more info.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
-Original Message-
From: Michael George [mailto:[EMAIL PROTECTED]
Sent: August 27, 2004 11:58 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] iaxtel and jitterbuffer
I am trying to work out IAX -- IAX communications with my * box. I have a
registration with iaxtel and I thought I would start there for my learning.
I am able to call the number for Digium's support line (700-428-6000), but
the
sound is horribly chopping. Some reading revealed the jitterbuffer
settings,
so I enabled them in iax.conf. I have the following now:
{clip}
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Re: [Asterisk-Users] iaxtel and jitterbuffer
Had this problem earlier this week - ensure 'trunk=no' in iax.conf if you're using fairly current CVS code. There is something not right w/the trunking that causes choppy sound. See the wiki for more info. I am using current CVS code and I have trunk=no. Still sounds crappy. I need to check with my ISP and make sure they aren't throttling in that range. I'm only getting about 4.5Kbps of throughput... Any available codecs that can use that level of bandwidth? I do a lot of work with companies throughout the US on network performance and we _frequently_ run into routers, switches, servers, etc, that are allowed to auto-negotiate their half vs full duplex nic interfaces. About 50% of the time, systems will get it wrong as there are no standards as to how the negotiation should be done. A recent case this past week indicated that data flow between two servers on the same layer-2 network was around 400 kbps when it should have been able to sustain at least 80 meg. You might double check each of your ethernet interfaces to ensure duplex settings are correct. If not at full duplex all the way through, you'll run into the strangeness you're seeing under varying traffic loads. ___ 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] iaxtel and jitterbuffer
On Aug 28, 2004, at 7:39 AM, Rich Adamson wrote: I do a lot of work with companies throughout the US on network performance and we _frequently_ run into routers, switches, servers, etc, that are allowed to auto-negotiate their half vs full duplex nic interfaces. About 50% of the time, systems will get it wrong as there are no standards as to how the negotiation should be done. No standard? Huh? You mean besides 'NWay', which is part of 802.3? http://www.scyld.com/NWay.html I've certainly seen problems, particularly with older Cisco switches and routers, but newer hardware seems to be pretty good. In fact, autonegotiation is *required* with GigE; you aren't even allowed to disable it according to the specs. Of course, that's sort of moot, because 1000/half isn't even slightly useful due to its 640-byte minimum packet size. At my previous employer, we were having tons of duplex problems. They mostly boiled down to forced duplex problems, where someone would force one end of a link, but leave the other end to autonegotiate. With most of Cisco's hardware, forcing 100/full *completely* disables autonegotiation. IMHO, it should still participate in autonegotiation, but only advertise the 100/full ability. Instead, Cisco tells the other end I don't negotiate. So, if you set one end to 100/full and fail to force the other end, then it will try to negotiate, fail, and fall back to 100/half, because that's the only reasonable thing to when negotiation fails. At this point, one end is 100/full and the other is 100/half, and you're about to have trouble. The really fun thing with this sort of link is that it works just fine with low traffic levels--a normal ping won't show problems, but it'll break when you actually try to use it for anything non-trivial. Using larger ping packets helps: ping -s 1 totally fails if the duplex is broken anywhere along the link. With newer IOS and CatOS builds, you can get around this by leaving CDP enabled; CDP v2 shares duplex information, and it'll log duplex mis-matches when both ends of the link use Cisco hardware. I wrote a small CDP listener for Linux boxes and did the same thing, logging duplex mis-matches. With 700 servers over 2 years, the only mismatches we ever found were caused by forced 100/full on the switches. One easy fix that we found, at least for IOS switches, was to set the speed to auto but force the duplex. That apparently leaves NWay negotiation running but only advertises full duplex as an option. Since nothing *ever* uses NWay to negotiate the speed of the link, this has the same result as forcing 100/full, but it fails in the right direction if you only force one end of the link. Of course, knowing Cisco, this only applies for every third model of switch running even-numbered IOS builds. Scott ___ 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] iaxtel and jitterbuffer
On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote: I do a lot of work with companies throughout the US on network performance and we _frequently_ run into routers, switches, servers, etc, that are allowed to auto-negotiate their half vs full duplex nic interfaces. About 50% of the time, systems will get it wrong as there are no standards as to how the negotiation should be done. A recent case this past week indicated that data flow between two servers on the same layer-2 network was around 400 kbps when it should have been able to sustain at least 80 meg. You might double check each of your ethernet interfaces to ensure duplex settings are correct. If not at full duplex all the way through, you'll run into the strangeness you're seeing under varying traffic loads. My ISP has a half-duplex connection between me and the world-at-large. It doesn't seem like that should be a problem, though, because we've been running VOIP with Multitech proprietary hardware for over two years now with little trouble and excellent voice quality. That was using a 9.6KBps codec. The difference between that and what I'm getting from IAX/GSM is profound, with GSM being intolerably poor quality. As a test, I ran two internal * machines with IAX/GSM between them. A conversation would consume from 7-10KBps, varying. Then I would call Digium's iaxtel number and I could see traffic from 4.5-8KBps and the voice was all choppy. I called another person's system (knowing they had IVR, of course) and the audio was also choppy, but when it got through the message and sent ring tones, they sounded fine. Then another voice message and it was choppy again. So I tried digium again. This time I could see the bandwidth being consumed, but I heard nothing on the line at all. So I tried calling my own iaxtel number. I could see my badwidth usage jump to about 10KBps, as I would expect and * told me that it was playing out the appropriate audo to the incoming caller. I heard nothing, however. Does this perhaps give any further indication of what might be wrong? I have in my [general] section: bandwidth=low disallow=all allow=gsm jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=100 minexcessbuffer=10 jittershrinkrate=1 trunk=no register = me:[EMAIL PROTECTED] tos=lowdelay I'm working towards a client install of IAX which will be used for inter-office VOIP, but I need to get this issue worked out or it's not deployable. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] iaxtel and jitterbuffer
On Sat, Aug 28, 2004 at 03:00:26PM -0400, Michael George wrote: On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote: I do a lot of work with companies throughout the US on network performance and we _frequently_ run into routers, switches, servers, etc, that are allowed to auto-negotiate their half vs full duplex nic interfaces. About 50% of the time, systems will get it wrong as there are no standards as to how the negotiation should be done. A recent case this past week indicated that data flow between two servers on the same layer-2 network was around 400 kbps when it should have been able to sustain at least 80 meg. You might double check each of your ethernet interfaces to ensure duplex settings are correct. If not at full duplex all the way through, you'll run into the strangeness you're seeing under varying traffic loads. I just saw a page on the wiki that mentions that running X11 or a VESA frame buffer can cause jittery sound. I only have this problem with IAX2, but that might be cause when I use Zap -- Zap or Zap -- SIP there is no en/decoding involved. I am running * on my main home server, which does run X and other software. Perhaps that's the problem? Maybe if I juiced it up with more RAM, might that help? It's at .5GB now, but I can easily take it to 1GB. Or, maybe a 900MHz Athlon still can't handle the coding with X11 running? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] iaxtel and jitterbuffer
On Sat, 28 Aug 2004 15:24:01 -0400, Michael George wrote: On Sat, Aug 28, 2004 at 03:00:26PM -0400, Michael George wrote: On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote: I do a lot of work with companies throughout the US on network performance and we _frequently_ run into routers, switches, servers, etc, that are allowed to auto-negotiate their half vs full duplex nic interfaces. About 50% of the time, systems will get it wrong as there are no standards as to how the negotiation should be done. A recent case this past week indicated that data flow between two servers on the same layer-2 network was around 400 kbps when it should have been able to sustain at least 80 meg. You might double check each of your ethernet interfaces to ensure duplex settings are correct. If not at full duplex all the way through, you'll run into the strangeness you're seeing under varying traffic loads. I just saw a page on the wiki that mentions that running X11 or a VESA frame buffer can cause jittery sound. I only have this problem with IAX2, but that might be cause when I use Zap -- Zap or Zap -- SIP there is no en/decoding involved. I am running * on my main home server, which does run X and other software. Perhaps that's the problem? Maybe if I juiced it up with more RAM, might that help? It's at .5GB now, but I can easily take it to 1GB. Or, maybe a 900MHz Athlon still can't handle the coding with X11 running? My understand, admittedly limited, is that the windowing system (X or other) generates a lot of interupts. This can burden the system that is also engaged in real-time tasks such as rpt for voip. That said, my Asterisk server is is an AMD2500+ with 512 MB ram. I did install the Gnome desktop with Fedora Core 1, but I don't do anything else on the server. It runs headless. I just ssh in to tweak * as needed. Michael Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 An ounce of pretention is worth a pound of manure. ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Saturday 28 August 2004 15:00, Michael George wrote: The difference between that and what I'm getting from IAX/GSM is profound, with GSM being intolerably poor quality. That's odd; every single voice call coming in and out of the company I work for is using the GSM codec with asterisk and IAX2... even the music on hold is passable. I have in my [general] section: bandwidth=low get rid of it; you're giving codecs below. disallow=all allow=gsm jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=100 minexcessbuffer=10 jittershrinkrate=1 My jitter settings are similar. max 500, maxexcess 100, minexcess 50, dropcount=2 (10, are you *insane*?!), jittershrink of 1. I'd slow down the shrink even more if I could, as even at 1 it's still noticeable. Please note that it seems impossible to disable jitter buffer between 20040806 CVS HEAD endpoints. The jitterbuffer numbers in iax2 show channels look live. The numbers look right (jitbuf 0ms) between 20040806 and RC1 (Nufone). I haven't upgraded since then. trunk=no I found 20040806 CVS HEAD to have odd little problems with trunking too. Trunking between 20040806 and RC1 (again, with nufone) work fine. I can't trunk to VPC at all or they can't hear me (I can hear them). Just to make clear: I have completely disabled the jitter buffer between myself and Nufone and the call quality has gone up slightly. I wasn't expecting this. Regards, Andrew ___ 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] iaxtel and jitterbuffer
On Saturday 28 August 2004 15:24, Michael George wrote: I just saw a page on the wiki that mentions that running X11 or a VESA frame buffer can cause jittery sound. I only have this problem with IAX2, but that might be cause when I use Zap -- Zap or Zap -- SIP there is no en/decoding involved. Asterisk is an application requiring hard realtime performance. Pretty much any telephony application is. Running *anything* in addition to asterisk is just asking for trouble. Actually I would be curious to see if asterisk performs better in a soft-realtime environment (i.e. what's actually easily possible with commodity PC hardware). -A. ___ 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] iaxtel and jitterbuffer
On Sat, Aug 28, 2004 at 05:08:30PM -0400, Andrew Kohlsmith wrote: On Saturday 28 August 2004 15:24, Michael George wrote: I just saw a page on the wiki that mentions that running X11 or a VESA frame buffer can cause jittery sound. I only have this problem with IAX2, but that might be cause when I use Zap -- Zap or Zap -- SIP there is no en/decoding involved. Asterisk is an application requiring hard realtime performance. Pretty much any telephony application is. Running *anything* in addition to asterisk is just asking for trouble. Since X11 and other daemons might be a problem on my main * server, I pulled out my little testbed and fired it up. It's a PII 266 (okay, not the fatest system) with 192MB RAM. X is not running and the Framebuffer has been turned off in /boot/grum/menu.lst. I have disabled all the servers except for sshd. I have the latest source from CVS HEAD as of about 30min ago. There is no Zap card in this sytem. The only phone on it is a SIP phone. With it I dial in to digium's 1-700 number. The audio is better, but still choppy and unacceptible. Looking at the * hardware recommendations page, this is by no means near the smallest recorded setup, so teh system shouldn't be underpowered. So even with X11 eliminated the sound is still bad to Digium. I tried another's 1700 number, and it sounded the same, so it's not something unique to digium and me. Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to work with my ISP only giving me 1/2 duplex service? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] iaxtel and jitterbuffer
On Saturday 28 August 2004 23:01, Michael George wrote: It's a PII 266 (okay, not the fatest system) with 192MB RAM. X is not running and the Framebuffer has been turned off in /boot/grum/menu.lst. I have disabled all the servers except for sshd. I have the latest source from CVS HEAD as of about 30min ago. Should be fine. I ran * on a P90 for a while; it did everything I needed except iLBC. :-) There is no Zap card in this sytem. The only phone on it is a SIP phone. With it I dial in to digium's 1-700 number. The audio is better, but still choppy and unacceptible. Is your SIP phone doing any kind of silence suppression? It must be turned off because asterisk takes its timing from the RTP stream and if the phone's not transmitting frames continuously you'll get shitty audio. Note that latest CVS HEAD looks like they're making provisions for self-timing but without a stable clock source it's unlikely to help you. There are ztdummy modules which use the RTC or certain brands of USB controller to provide adequate timing but ideally you want some kind of Zaptel hardware in there providing a 1000Hz interrupt. Also -- make sure your uplink is acceptable. First test: make sure there is nothing plugged into your upstream except for your asterisk box and the phone. Some routers are known to play silly bugger with your packets which naturally wreaks havoc with asterisk. :-) So even with X11 eliminated the sound is still bad to Digium. I tried another's 1700 number, and it sounded the same, so it's not something unique to digium and me. Perhaps something to do with your upstream or connection to IAXtel. That's why I'm recommending having nothing but asterisk and the phone on the connection, at least until we nail down what the poor audio's being caused by. Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to work with my ISP only giving me 1/2 duplex service? It has nothing to do with IAX or GSM. Stop blaming them. My upstream is half duplex as well (pretty much anyone on DSL or cable is on a half duplex connection whether they realize it or not). There are many, many people using asterisk every day for long distance and in environments where audio quality is crucial. Let's stop blaming asterisk and take a good hard look at what's happenning, shall we? -A. ___ 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] iaxtel and jitterbuffer
I am trying to work out IAX -- IAX communications with my * box. I have a registration with iaxtel and I thought I would start there for my learning. I am able to call the number for Digium's support line (700-428-6000), but the sound is horribly chopping. Some reading revealed the jitterbuffer settings, so I enabled them in iax.conf. I have the following now: ; Inter-Asterisk eXchange driver definition ; [general] ; Specify bandwidth of low, medium, or high to control which codecs are used ; in general. ; bandwidth=low ; ; You can also fine tune codecs here using allow and disallow clauses ; with specific codecs. Use all to represent all formats. ; disallow=lpc10 ; Icky sound quality... Mr. Roboto. allow=gsm ; Always allow GSM, it's cool :) jitterbuffer=yes dropcount=3 maxjitterbuffer=500 maxexcessbuffer=100 minexcessbuffer=10 jittershrinkrate=1 register = :[EMAIL PROTECTED] ; Finally, you can set values for your TOS bits to help improve ; performance. Valid values are: ; lowdelay-- Minimize delay ; throughput -- Maximize throughput ; reliability -- Maximize reliability ; mincost -- Minimize cost ; none-- No flags ; tos=lowdelay but I still have a less-than-acceptible quality connection. The bandwidth usage is right around 5.5-6kbps. I have a multivoip that ran at about that rate and sounded fine (obviously with a different codec, but my point is that my broadband connection shouldn't be the issue). Any helpful suggestions? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] iaxtel and jitterbuffer
Had this problem earlier this week - ensure 'trunk=no' in iax.conf if you're
using fairly current CVS code. There is something not right w/the trunking
that causes choppy sound. See the wiki for more info.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
-Original Message-
From: Michael George [mailto:[EMAIL PROTECTED]
Sent: August 27, 2004 11:58 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] iaxtel and jitterbuffer
I am trying to work out IAX -- IAX communications with my * box. I have a
registration with iaxtel and I thought I would start there for my learning.
I am able to call the number for Digium's support line (700-428-6000), but
the
sound is horribly chopping. Some reading revealed the jitterbuffer
settings,
so I enabled them in iax.conf. I have the following now:
{clip}
___
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Re: [Asterisk-Users] iaxtel and jitterbuffer
On Fri, Aug 27, 2004 at 12:47:05PM -0700, Kris Boutilier wrote: Had this problem earlier this week - ensure 'trunk=no' in iax.conf if you're using fairly current CVS code. There is something not right w/the trunking that causes choppy sound. See the wiki for more info. I am using current CVS code and I have trunk=no. Still sounds crappy. I need to check with my ISP and make sure they aren't throttling in that range. I'm only getting about 4.5Kbps of throughput... Any available codecs that can use that level of bandwidth? I'll have to check out the speex codec... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] iaxtel, asterisk, and sipura 1000 am having trouble with codecs
hello I am trying to set up iaxtel with asterisk and am using a sipura 1000 when my friend calls me he is sounding like he is in a metal tank that is the best way I can describe it, how ever when he calls me on my grand stream budjet phone 101 it sounds fine. is there a fix for this really anoying problem? thanks hank My Inbox is protected by SPAMfighter264 spam mails have been blocked so far.Download free SPAMfighter today!
[Asterisk-Users] IAXtel questions
I have just get an account on Iaxtel.com, and i woud like to know what can i do to receive my Iaxtel calls in my asterisk server? Actually i just can make IAX calls. Thanks -- ___ Get your free email from http://www.hackermail.com Powered by Outblaze ___ 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] IAXTel Help
I've searched WIKI and Archives but nothing. Im getting: -- Called username:[EMAIL PROTECTED]/[EMAIL PROTECTED] Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call rejected by 69.73.19.178: Unable to negotiate codec -- Hungup 'IAX2[Iaxtel]/8' == No one is available to answer at this time -- Executing Hangup(SIP/104-b8eb, ) in new stack == Spawn extension (home, h, 1) exited non-zero on 'SIP/104-b8eb' IAX.CONF [general] port=5036 register = mynumber:[EMAIL PROTECTED] register = khagan:[EMAIL PROTECTED] disallow=all allow=ulaw [iaxfwd] type=user context=fromiaxfwd ;context=local deny=0.0.0.0/0.0.0.0 permit=65.39.205.0/255.255.255.0 [Iaxtel] type=friend host=iaxtel.com secret=password auth=rsa context=from-iaxtel inkeys=iaxtel Please help me. Im working with IAX FWD. Tried putting different codec's in. extension.conf is as it said to setup. Kyle ___ 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] IAXTel Help
Kyle Hagan wrote: I've searched WIKI and Archives but nothing. Im getting: -- Called username:[EMAIL PROTECTED]/[EMAIL PROTECTED] Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call rejected by 69.73.19.178: Unable to negotiate codec -- Hungup 'IAX2[Iaxtel]/8' == No one is available to answer at this time -- Executing Hangup(SIP/104-b8eb, ) in new stack == Spawn extension (home, h, 1) exited non-zero on 'SIP/104-b8eb' IAX.CONF [general] port=5036 register = mynumber:[EMAIL PROTECTED] register = user:[EMAIL PROTECTED] disallow=all allow=ulaw Had to enable GSM for it to work. No other support? Didnt see anywhere that had to use GSM. Kyle ___ 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] iaxtel 1-800 gateway down?
Just dialed (or attempted to) a 800 number, still down At 17:20 6/8/2004, you wrote: Heh..yea, I made sure I did a search through the archives before posting it :) (not that I'm complaining) The weird thing though is that I _am_ able to call digium's iaxtel number.. -Mark ___ 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] iaxtel 1-800 gateway down?
tmpm wrote: Just dialed (or attempted to) a 800 number, still down you could always enable enum lookups and use either the freenum.org zone or e164.org zone as they both contain IAX2 and SIP URLs for north american and other countries toll free numbers... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ 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] iaxtel 1-800 gateway down?
Thanks for the tip. will look into that... At 05:47 6/9/2004, you wrote: tmpm wrote: Just dialed (or attempted to) a 800 number, still down you could always enable enum lookups and use either the freenum.org zone or e164.org zone as they both contain IAX2 and SIP URLs for north american and other countries toll free numbers... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ 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] iaxtel 1-800 gateway down?
Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (ContextExtensionPri ) State Appl. Data IAX2[iaxtel]/1 ( s1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 1476626 1 )Ring Dial IAX2/XXX:[EMAIL PROTECTED]/[EMAIL PROTECTED] But I never hear a ringing on the actual phone, and it seems to stay in this state (i.e. never gets to bridge mode) for a long time..to a point that ijust hang up. Thanks, Mark ___ 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] iaxtel 1-800 gateway down?
Do you have r on your Dial line? If so, then Asterisk will override whatever should you SHOULD be hearing and provide you with a ringing sound. On Tue, 2004-06-08 at 10:24, Mark Musone wrote: Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (ContextExtensionPri ) State Appl. Data IAX2[iaxtel]/1 ( s1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 1476626 1 )Ring Dial IAX2/XXX:[EMAIL PROTECTED]/[EMAIL PROTECTED] But I never hear a ringing on the actual phone, and it seems to stay in this state (i.e. never gets to bridge mode) for a long time..to a point that ijust hang up. Thanks, Mark ___ 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 -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] iaxtel 1-800 gateway down?
Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (ContextExtensionPri ) State Appl. Data IAX2[iaxtel]/1 ( s1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 1476626 1 )Ring Dial IAX2/XXX:[EMAIL PROTECTED]/[EMAIL PROTECTED] But I never hear a ringing on the actual phone, and it seems to stay in this state (i.e. never gets to bridge mode) for a long time..to a point that ijust hang up. Same here at least with an 800 number just tested. Registration is fine, but calls do not appear to be handled at all. ___ 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] iaxtel 1-800 gateway down?
It seems to be down, I even tried dialing for example 1-800-555-TELL. I tried yesterday and again today.. Just get dead air. Stephen Rosebush Mark Musone wrote: Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (ContextExtensionPri ) State Appl. Data IAX2[iaxtel]/1 ( s1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 1476626 1 )Ring Dial IAX2/XXX:[EMAIL PROTECTED]/[EMAIL PROTECTED] But I never hear a ringing on the actual phone, and it seems to stay in this state (i.e. never gets to bridge mode) for a long time..to a point that ijust hang up. Thanks, Mark ___ 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] iaxtel 1-800 gateway down?
Down here. It seems to be down, I even tried dialing for example 1-800-555-TELL. I tried yesterday and again today.. Just get dead air. Stephen Rosebush Mark Musone wrote: Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (ContextExtensionPri ) State Appl. Data IAX2[iaxtel]/1 ( s1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 1476626 1 )Ring Dial IAX2/XXX:[EMAIL PROTECTED]/[EMAIL PROTECTED] ___ 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] iaxtel 1-800 gateway down?
same with their 700 network w - Original Message - From: Mark Musone [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 08, 2004 11:24 AM Subject: [Asterisk-Users] iaxtel 1-800 gateway down? Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (ContextExtensionPri ) State Appl. Data IAX2[iaxtel]/1 ( s1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 1476626 1 )Ring Dial IAX2/XXX:[EMAIL PROTECTED]/[EMAIL PROTECTED] But I never hear a ringing on the actual phone, and it seems to stay in this state (i.e. never gets to bridge mode) for a long time..to a point that ijust hang up. Thanks, Mark ___ 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] iaxtel 1-800 gateway down?
Ive got similar probs Mark, and no one either here (unless I havent got thru the pile yet) or on the IRC channel last nite answered. Ive simply got no response when I try to use Iaxtel to call anywhere. My distant end is experienceing the exact same thing. I also tried FWD to Iaxtel, and it craps out too, but FWD is fine. Im showing registered on Iaxtel, and if i dial, all I get is silence till call timeout. Ive seen this for three days now, and am hesitant to post, because my email from this list maxed out a couple days before the end of the month, and as we've all seen, if you ask a question that's been previously asked, (or answered) we get some rather snap, growl, bite responses from under the rocks when we poke around. So I simply sit around and wait a while... Marc At 11:24 6/8/2004, you wrote: Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (ContextExtensionPri ) State Appl. Data IAX2[iaxtel]/1 ( s1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 1476626 1 )Ring Dial IAX2/XXX:[EMAIL PROTECTED]/[EMAIL PROTECTED] But I never hear a ringing on the actual phone, and it seems to stay in this state (i.e. never gets to bridge mode) for a long time..to a point that ijust hang up. Thanks, Mark ___ 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] iaxtel 1-800 gateway down?
Thanks for verifying that...thats what I thought...took two days to verify it... At 13:21 6/8/2004, you wrote: same with their 700 network w - Original Message - From: Mark Musone [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 08, 2004 11:24 AM Subject: [Asterisk-Users] iaxtel 1-800 gateway down? Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (ContextExtensionPri ) State Appl. Data IAX2[iaxtel]/1 ( s1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 1476626 1 )Ring Dial IAX2/XXX:[EMAIL PROTECTED]/[EMAIL PROTECTED] But I never hear a ringing on the actual phone, and it seems to stay in this state (i.e. never gets to bridge mode) for a long time..to a point that ijust hang up. Thanks, Mark ___ 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] iaxtel 1-800 gateway down?
Heh..yea, I made sure I did a search through the archives before posting it :) (not that I'm complaining) The weird thing though is that I _am_ able to call digium's iaxtel number.. -Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of tmpm Sent: Tuesday, June 08, 2004 4:37 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxtel 1-800 gateway down? Ive got similar probs Mark, and no one either here (unless I havent got thru the pile yet) or on the IRC channel last nite answered. Ive simply got no response when I try to use Iaxtel to call anywhere. My distant end is experienceing the exact same thing. I also tried FWD to Iaxtel, and it craps out too, but FWD is fine. Im showing registered on Iaxtel, and if i dial, all I get is silence till call timeout. Ive seen this for three days now, and am hesitant to post, because my email from this list maxed out a couple days before the end of the month, and as we've all seen, if you ask a question that's been previously asked, (or answered) we get some rather snap, growl, bite responses from under the rocks when we poke around. So I simply sit around and wait a while... Marc At 11:24 6/8/2004, you wrote: Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (ContextExtensionPri ) State Appl. Data IAX2[iaxtel]/1 ( s1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 1476626 1 )Ring Dial IAX2/XXX:[EMAIL PROTECTED]/[EMAIL PROTECTED] But I never hear a ringing on the actual phone, and it seems to stay in this state (i.e. never gets to bridge mode) for a long time..to a point that ijust hang up. Thanks, Mark ___ 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
[Asterisk-Users] iaxtel and d-link router
I've been playing around with asterisk for the last few weeks and now I have the system up and running but whenever I make a call using iaxtel all is good for the first call. After I hang up the call the d-link router looses it's mind and must be rebooted. Nothing IP will work through the router (to the internet) after the call. Has anyone else seen this happen? I know what the solution is... new router Chris ___ 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] iaxtel and d-link router
check for a firmware update first. i had problems with a d-link until i did a firmware update and that fixed it. - Original Message - From: Christopher C. Howard [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, April 20, 2004 2:21 PM Subject: [Asterisk-Users] iaxtel and d-link router I've been playing around with asterisk for the last few weeks and now I have the system up and running but whenever I make a call using iaxtel all is good for the first call. After I hang up the call the d-link router looses it's mind and must be rebooted. Nothing IP will work through the router (to the internet) after the call. Has anyone else seen this happen? I know what the solution is... new router Chris ___ 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] IAXTel toll-free gateway
Title: IAXTel toll-free gateway Is anyone else having trouble placing toll-free calls though IAXTel lately? Mine just stopped working yesterday, yet I seem to be able to make 1-700 calls. -brian 1-700-676-3830
Re: [Asterisk-Users] IAXTel toll-free gateway
Is anyone else having trouble placing toll-free calls though IAXTel lately? Mine just stopped working yesterday, yet I seem to be able to make 1-700 calls. It's up/down/etc rather frequently, so no surprise. Good thing it's not a paid service or we'd all have an issue. Consider it as a temporary testing facility, not a production resource. ___ 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] IAXTel multiple registers?
With entries in sip.conf, I can route incoming SIP calls with an extension specified in the register command: register = user:[EMAIL PROTECTED]/123 The register command in iax.conf does not support specifying the extension. If I want to register multiple IAXTel accounts, how can I make them branch to different extensions or contexts when a calls arrives? ___ 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] IAXTel multiple registers?
You do this with contexts attached to the [provider] section in the iax.conf file and you provide coresponding contexts and extensions in your extensions.conf file. John Barton Hodges wrote: With entries in sip.conf, I can route incoming SIP calls with an extension specified in the register command: register = user:[EMAIL PROTECTED]/123 The register command in iax.conf does not support specifying the extension. If I want to register multiple IAXTel accounts, how can I make them branch to different extensions or contexts when a calls arrives? ___ 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] IAXTel multiple registers?
Both register commands register with the iaxtel provider. No matter which number is dialed to reach Asterisk, it takes you to the same [provider] section, and thus the same context. I need for 2 register commands, registering to the same provider, to branch to different contexts or extensions. [EMAIL PROTECTED] wrote: You do this with contexts attached to the [provider] section in the iax.conf file and you provide coresponding contexts and extensions in your extensions.conf file. John Barton Hodges wrote: With entries in sip.conf, I can route incoming SIP calls with an extension specified in the register command: register = user:[EMAIL PROTECTED]/123 The register command in iax.conf does not support specifying the extension. If I want to register multiple IAXTel accounts, how can I make them branch to different extensions or contexts when a calls arrives? ___ 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
