Re: [Asterisk-Users] Dual PRI fail over
NFAS isn't needed for redundancy.. You can do this with 2 PRI's just have them in the same huntgroup at the CO.. All NFAS really does is free up a extra B Channel.. For 2 PRI's it makes no sense.. For anything over 2 you pickup 1 extra B channel per PRI.. We use NFAS on our RAS setup for the ISP.. We get a extra 6 lines on our 8 PRI group. - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 12, 2005 12:48 AM Subject: Re: [Asterisk-Users] Dual PRI fail over Tom wrote: I currently have a single PRI however we are getting a second PRI, and the provider (qwest) wants to know if our PBX supports GSAS (they say its a redundant d-channel technology but searching on google for GSAS reveals less than nothing). I've set something similar up before on a cisco 5350, where if one of the PRIs fails, all of the calls destined for either PRI will be routed down the one that didn't fail. Basically the 2 PRIs are bonded together, and act as one. During normal operation the calls come down each PRI in a load balanced fashion (IE if I've got 30 calls up, 15 will be on one PRI and 15 on the other). The term you are looking for is NFAS (Non-Facilities Associated Signaling), and it's fully supported in Asterisk. You can configure your two PRIs as a single trunk group with a primary and backup D-channel, and calls can be handled on both PRIs equally. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual PRI fail over
See this link: http://www.voip-info.org/tiki-index.php?page=NFAS - Original Message - From: Billy Huddleston [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, October 12, 2005 12:57 AM Subject: Re: [Asterisk-Users] Dual PRI fail over NFAS isn't needed for redundancy.. You can do this with 2 PRI's just have them in the same huntgroup at the CO.. All NFAS really does is free up a extra B Channel.. For 2 PRI's it makes no sense.. For anything over 2 you pickup 1 extra B channel per PRI.. We use NFAS on our RAS setup for the ISP.. We get a extra 6 lines on our 8 PRI group. - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 12, 2005 12:48 AM Subject: Re: [Asterisk-Users] Dual PRI fail over Tom wrote: I currently have a single PRI however we are getting a second PRI, and the provider (qwest) wants to know if our PBX supports GSAS (they say its a redundant d-channel technology but searching on google for GSAS reveals less than nothing). I've set something similar up before on a cisco 5350, where if one of the PRIs fails, all of the calls destined for either PRI will be routed down the one that didn't fail. Basically the 2 PRIs are bonded together, and act as one. During normal operation the calls come down each PRI in a load balanced fashion (IE if I've got 30 calls up, 15 will be on one PRI and 15 on the other). The term you are looking for is NFAS (Non-Facilities Associated Signaling), and it's fully supported in Asterisk. You can configure your two PRIs as a single trunk group with a primary and backup D-channel, and calls can be handled on both PRIs equally. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Over 10,000 lines. Will asterisk manage?
A SIP phone *could* normally send its media stream directly from phone to phone, if no transcoding is required, but when using Asterisk the media stream will always pass through the server, causing a pottential bottleneck. So, why not use SER to register all the SIP phones, as it doesn't handle the media-streams, just keeps track of the phones and does the 'handshake'. SER is supposed to be able to handle over 50.000 calls at a time, so one SER server would be enough. Then interface this with one (or more) Asterisk servers to connect to the local PSTN. But maybe I'm missing something fundamental, in which case I'm happy to learn. Um, Wrong, You can do re-invites and have the media go point-to-point, We do it all the time. ___ 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] Cisco 7960 + 7914 - not worked
7914 isn't supported with asterisk as of yet. - Original Message - From: Vasiliy Voropaev To: [EMAIL PROTECTED] Sent: Friday, October 15, 2004 6:32 PM Subject: [Asterisk-Users] Cisco 7960 + 7914 - not worked I have Cisco 7960 with 7914 operator console. 7960 successfully registered and working with chan_sccp2, but the buttons on the 7914 are all red. What may be wrong? sccp.conf: [SEP] description = VVG type = 7914 context = sip autologin = 821 speeddial = 11,Test1 speeddial = 12,Test2 Firmware version 3.1(MF.G2) "Expansion Module Stats" menu displays that Link State is "Not Supported". May be i need to upgrade firmware? Or add some extra option in SEP.cnf.xml ? Sorry for bad English. Best Regards, Vasiliy Voropaev. ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Polycom SIP 1.3.1 Reject Button
Sounds like you need to talk to polycom about a reduction in the capabilities of thier phone after the upgrade and have them move the menu option back.. - Original Message - From: Tor Setane [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 8:57 AM Subject: [Asterisk-Users] RE: Polycom SIP 1.3.1 Reject Button Brent D. Franks wrote: Hello, I recently upgraded to Sip 1.3.1 and noticed that the Reject Button is no longer appearent on the screen when a second incoming call comes in unless I press the hold button on the first call. Does anyone have a work around for this to reject a call while continuing to talk to the first party? I should also point out that I don't want it to be on *, as the situation varies from call to call. E.g. setting a count limit on a phone is not acceptable, as if the secretary is talking to someone from home, she can put them on hold and take the second call. Thanks, Brent D. Franks I can only answer for the IP600 - when I want to reject the second incoming call, I can do that with the Do Not Disturb button, or I can press blue down arrow and than use the reject soft key. The ongoing call is not interrupted and I don't have to put it on hold first. Regards, Tor Setane ___ 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] Cisco GW and DTMF problems
I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have dtmf-relay rtp-nte toggled in my dial peer.. Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [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] Cisco GW and DTMF problems
c2600-is5-mz.123-9 rfc2833 - Original Message - From: Tenorio, Leandro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, September 08, 2004 10:37 PM Subject: RE: [Asterisk-Users] Cisco GW and DTMF problems What version of IOS 're u using, and what's your dtmfmode in *? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Billy Huddleston Sent: Wednesday, September 08, 2004 6:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco GW and DTMF problems I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have dtmf-relay rtp-nte toggled in my dial peer.. Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [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 ___ 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] Asterisk GUIs at Astricon * REMINDER *
Dude, don't flam.. People don't use HTML capable E-Mail programs, or turn off html for reason.. Like spam and web-bugs, and/or using classic email programs like pine and mutt and linux. Geez.. - Original Message - From: Karl J. Vesterling To: [EMAIL PROTECTED] Sent: Sunday, August 01, 2004 10:17 PM Subject: RE: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER * Get an HTML capable E-Mail Program. They've been freely available for nearly 10 years now and on just about every platform that's got more than a 16 bit bus. PS: How do you manage/configure asterisk from punch card / paper tape? Just curious, I thought you might know... At 07:37 PM 8/1/2004, you wrote: 1. Don't post in html...I have to scroll three pages to the right just to read it. Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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] oh323 codec G7231A6K3
Asterisk doesn't support any form of G723 except with Pass through... You might try G726 or G729 Thanks, Billy - Original Message - From: Arnaud Pignard [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 23, 2004 12:45 PM Subject: [Asterisk-Users] oh323 codec G7231A6K3 Hi, I would like use codec G7231A6K3 with oh323, but seems asterisk don't undestood this codec. I can't use G7231, the remote gateway don't accept this version of G723. Thanks for help. -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ 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] BRI dead in USA?
TN is one of the BEST states for BRI's.. Bellsouth messed up and had to make some concessions to the PUC a long time ago.. You can get BRI anywhere, and it's a flat fee.. typically $80-$90 per month Biz rate, $35-45 Residential.. Thanks, Billy +--+ | Billy HuddlestonSenior System Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 5:27 PM Subject: RE: [Asterisk-Users] BRI dead in USA? On Tue, 2004-07-20 at 14:15, Scott Stingel wrote: Brian- Wow - that is high! I got quoted only $35/month for BRI (and a hefty installation) - not too bad. But the comment about no CLI scared me off. In Nashville, it is $90/month for a BRI without per minute charges for business. I remember paying $35/month for residential about 10 years or so ago. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian Sent: Tuesday, July 20, 2004 9:46 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] PRI dead in USA? Well they fail to realize that ISDN is used for more than data. I just wanna scream at them and say IT DOES VOICE TO YOU NINNY!.. Rates are far from reasonable. 167/mth here is what I would have to pay for ISDN-BRI. SBC is lame. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Scott Stingel Sent: Tuesday, July 20, 2004 10:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI dead in USA? Hi- Because a majority of my customers are in Europe, I've gotten quite used to working with ISDN (PRI) and BRI on a regular basis. Recently one of my customers asked me if I could terminate a few lines locally here in the USA (California), so I called up SBC to enquire as to how much it would cost to install a BRI here. Although the rates were reasonable (except the installation), I got the distinct impression that they really didn't want to install BRI's. Their comments were well, BRI is getting quite antiquated, and the like. They said with the advent of ADSL, there's not much of a market anymore, as most of past usage was modem related. I'm a little worried about the pricing going up, and availability going down in the near future. I don't have the volume yet to justify PRI. What are other's experience in the US with BRI? Also, they mentioned that I couldn't get caller ID with the BRI service, which I thought was a built- in feature. Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [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 ___ 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] T1 Hardware Echo Can
Do these work with PRI's as well? What's a ball park price on these? Thanks, Billy - Original Message - From: Robb Meredeth [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 09, 2004 8:53 PM Subject: Re: [Asterisk-Users] T1 Hardware Echo Can Well, first off we're not using them directly off the asterisk. Our Asterisk runs into our Alcatel OmniPCX 4400 and then to the PRI. We have about 70 or so IP phones off the 4400 and just a few off the Asterisk box (so far) and of those probably 10 to 15 (alcatel) users complain consistently about the echo. We've only put it on one of our spans to start with and I directed the most complaining users to it. So far the feedback has been very good. I use an IP phone every day and I rarely notice it, but I may have just learned to tune it out. :) But having said that I have noticed that even that occasional echoing call that I was getting seems to have gone away. Robb - Original Message - From: Brent Franks [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 09, 2004 7:41 PM Subject: RE: [Asterisk-Users] T1 Hardware Echo Can Ditech Communications ( http://www.ditechcom.com/ ) has a 2 slot and a 4 slot chassis and you can populate them with just 1 echo cancelling card if you like. They have RJ45 jacks and you have to use a T1 Cross cable on both sides. We Hi Robb, Thanks for your reply. Have you found that the Echo Can's from Ditech take care of your echo? Do you experience echo on any calls (e.g. 1 out of 300 calls, 1 out of 500..?) Thanks a lot for your advice on this. I will give them a call. Additionally, what is a ball park range on pricing? Thanks in advance! - Brent ___ 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] asterisk grandstream aleatory error
I've been working with Mark today on fixing this very bug.. The patch ProgramerTED did may have fixed it, but, I don't think it was the right fix. We should have something done later today on this problem. Thanks, Billy aka Connor - Original Message - From: Alberto Fernandez [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 10:37 AM Subject: Re: [Asterisk-Users] asterisk grandstream aleatory error There is an open BUG for this problem, ProgramerTED sent also a fix for this issue. But it still not in the CVS, Im waiting with a verry OLD version because of it. Roll back to before 6/15/02 and you will be fine. Hopefully soon someone can include that fix into the cvs. On Tue, 2004-07-06 at 20:55, interopen wrote: Hello all, It just start happening a week ago on a handytone 286, sip extension, every time i call to this extension it rings one time and it hungs, in the asterisk console this error repeats: Jul 7 00:19:32 WARNING[98311]: chan_sip.c:471 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) I change the ip of the handytone and work for a couple of days but start again. Any solution or trick. Thanks in advanced, Ivan ___ 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] Echo cancellation, when software doesn't cut it. Whats next?
I've got the same problem NEAR end echo (We hear the echo on OUR side, person on the PSTN never hears it..) We're tyring to get our PRI carrier to run us through a echo can, or re-write it through a switch they have which has built in echo cans... Ugg.. Thanks, Billy - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 01, 2004 10:56 AM Subject: Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next? On Thursday 01 July 2004 08:39, Robinson Tim-W10277 wrote: All voip providers will use digital 4-wire interconnect to Asterisk or similar, so echo problems are much reduced, as there are only 'echo points' at the far end and your handset. And on my PRI that is specifically where my echo is coming from... the far end. VOIP calls through nufone have no echo MOST PSTN calls through the PRI have no echo SOME PSTN calls (usually to local numbers NOT terminated at my local CO) have significant echo... I too have been unsuccessful in getting this zapped. My connection: Norstart MICS -- Adit600 --- T100P -- IAX2 -- TE405P -- Bell Canada PRI *1 = Xeon/2.4 with HT with T100P *2 = Xeon/2.4 with HT with TE405P *2 also does the NuFone IAX2 connection (it is always in the loop, as *1 is on a private network) Strange stuff, I am going to look at T1 echo cancellation hardware if I cant' get this solved. Tried: - echotraining=800 on *1 and *2 - echocancel=32,64,128 on both Eventually the MICS will have a digital connection to *1 instead of going through the Adit600 but we haven't got there yet :-) 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 ___ 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] Echo cancellation, when software doesn't cut it. Whats next?
I failed to mention I'm using a Cisco 2600 with Sip Re-invites.. and YES, I do have the echo can on the Cisco turned on, The echo is S bad, it's not even touching it... When we place 1-800 calls or call LD via our offnet provider, everything works fine, it's just with local calls on the PRI... We found out that the 1-800 #'s go out our carriers Sonus switch (a VoIP switch) which has 128ms Echo Can in it... Hmmm... Thanks, Billy - Original Message - From: Brent Franks [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 01, 2004 12:16 PM Subject: Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next? On Thu, 1 Jul 2004, Mike Benoit wrote: Obviously the less I spend the better. But if we have to, a few thousand more I guess. The problem I have is that this setup is more of a trial run. Once it works, I'm going to be cloning slightly smaller setups to 9 other cities. But they are pretty small, 1 or 2 lines and 2-4 phones in each location. I totally understand this. My users complain frequently about echo, and I am still unable to determine why sometimes it works great, other's it does not. The CPU and Memory are powerful enough to handle it, and we rarely ever see any load on the box. I too feel this is the major caveat to Asterisk right now. I am curious how anyone is achieving a near echo free system. We are shooting for 1 out of every 300 calls to have echo, which I think can be a realistic goal. Given the nature of open source, and the mix-and-match of components that come up, I can see where Digium is in a hard place to nail down the cause of every occurance. I will only be using POTS lines in each location. The current setup works great besides the echo, and some of the information I've read point to the Telco being the issue. If thats the case, I should in theory be able to get them to fix the problem. (though I could be dreaming) I think ultimately, if a Mediatrix box, or Cisco box can accomplish echo cancellation, Asterisk should be able to do it with as much success. Being that I am not an experienced Programmer, I try not to complain to loudly. With my level of involvement, I typically make the business case to customers and spec out ROI, etc. I do have a technical background, and am getting better at trouble shooting Asterisk and working on the source code. In fact, subscribing to the CVS list has taken me leap years ahead of understanding the changes and why they are being committed. I don't know how much more putting a DSP to handle echo can on the cards would cost, but if it were 400 - 500 more I would certainly pay it without a second thought, provided it worked. Echo, I think, is the largest draw back to VoIP, and will be the limit to entry into many businesses. I know my client, if they were to do it all over again, would choose a regular TDM (nortel, avaya) solution over the echo they are experiencing. I think asterisk is definitly headed in the right direction though, and nothing good comes over night. So everyone who has worked on it deserves to be commended. Without their insight and dedication, we wouldn't even be talking about this, or have alternatives to turn to. Regards, - Brent ___ 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] anyone use mailboxexists?
Yes, I use it. Here's a sample extension of how to use it. exten = 1234,1,Answer() exten = 1234,2,MailboxExists(1234) exten = 1234,3,Dial(SIP/1234,20) ; Try to ring for 20 seconds, no answer goto voicemail exten = 1234,4,Voicemail(b1234) ; send to voicemail if busy exten = 1234,103,Dial(SIP/1234) ; Try to ring till answered exten = 1234,104,Busy() ; Give busy tone if busy. exten = 1234,204,Voicemail(u1234) ; send to voicemail if no answer - Original Message - From: Michael George [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 15, 2004 5:36 PM Subject: [Asterisk-Users] anyone use mailboxexists? I replied to a post of mine a few days ago asking of anyone uses mailboxexists(). I haven't received any replies. Perhaps few use it or perhaps the reply was overlooked. I thought I'd post the question one last time before giving up on it for now... Thanks! -Michael ___ 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] e164.org
'local' target? What's that?
- Original Message -
From: Matthew Asham [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 22, 2004 9:55 PM
Subject: Re: [Asterisk-Users] e164.org
You know, sleep deprivation cause people to do dumb things. The example
I pasted was hastily pasted and renumbered,
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
are actually:
exten = _1NX,103,Playback(enum-lookup-failed)
exten = _1NX,104,Hangup
Duane wrote up some more detailed examples at
http://www.e164.org/config.php.
Sorry for not proofing that when I posted it. I'll go sleep now.
On Sat, 2004-05-22 at 18:46, Tony Hoyle wrote:
Matthew Asham wrote:
; north america enum
exten = _1NX,1,Playback(doing-enum-lookup)
exten = _1NX,2,EnumLookup(${EXTEN})
exten = _1NX,3,BackGround(enum-lookup-successful)
exten = _1NX,4,Dial(${ENUM},30,tr)
exten = _1NX,5,Hangup
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
Interesting.. how does it know to go to '6', or does it just jump +4
on failure?
That reminds me I seriously need to restructure my extensions.conf...
there's
no way currently I could add anything like that without major surgery
(only
discovered the 'local' target this afternoon so I have everything
copied/pasted).
Tony
___
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] *** Asterisk sunday news: Read the sampleconfigs, Luke!
Mark, Would you please re-config or use a different mail client as to not send your replies back as attachments?? It's VERY kludgy, and, I'm just going to stop reading them.. along with all the other folks.. Thanks, Billy - Original Message - From: Mark Elkins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 8:41 AM Subject: Re: [Asterisk-Users] *** Asterisk sunday news: Read the sampleconfigs, Luke! ___ 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] Cisco 7940 Phones as paging system?
That won't work.. That'll DIAL multiple phones/extensions, but will only
bridge 1 of them when it auto-answers..
What we need is a way to have something like meetme call multiple extensions
and bridge them to a meetme confrence (all of them muted but the admin of
course, as it's a one way page) and then we would have a true paging
system..
- Original Message -
From: Vic Cross [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 08, 2004 10:11 AM
Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system?
On Fri, 7 May 2004, Ian A. Underwood wrote:
Joe Antkowiak wrote:
exten = 5101,1,Dial(SIP/5101,10,tA(intercom-tone))
exten = 5101,2,Congestion
That's not too bad, but how do you page a group of phones...like a real
intercom? That's what I'm dying to know!
in extensions.conf:
[globals]
INTERCOMLINES=SIP/Alice6SIP/Bob6SIP/Chuck6...
Then the extension is as per Joe's example, but replacing SIP/5101 with
${INTERCOMLINES}.
Extending this, you could set up various intercom numbers for different
parts of the office...
[globals]
SALESINTERCOM=SIP/Sales1-6SIP/Sales2-6...
MKTGINTERCOM=SIP/Marketing1-6...
...
[yourcontext]
exten = 5101,1,Dial(${SALESINTERCOM},10,tA(tone))
exten = 5102,1,Dial(${MKTGINTERCOM},10,tA(tone))
...
exten = 5110,1,Dial(${SALESINTERCOM}${MKTGINTERCOM}${...},10,tA(tone))
Cheers,
Vic Cross
___
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] 729 licence on scsi
SO, do you have a IDE CDROM? - Original Message - From: Mark Elkins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 07, 2004 4:13 PM Subject: [Asterisk-Users] 729 licence on scsi ___ 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] Cisco 7940 Phones as paging system?
hey, can you send me the tone? - Original Message - From: Joe Antkowiak To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Friday, May 07, 2004 4:30 PM Subject: RE: [Asterisk-Users] Cisco 7940 Phones as paging system? This is what we have for this customer. They have five phones right now. Their normal extensions are 610x, but for intercom its 510x: exten = 5101,1,Dial(SIP/5101,10,tA(intercom-tone)) exten = 5101,2,Congestion If you want the wav file, let me know. If you make your own, be sure to put a 1-2 second pause in the beginning, because when the cisco answers it takes a second or to before it will send any audio to the speaker. -Original Message-From: mitchel [mailto:[EMAIL PROTECTED] Sent: Friday, May 07, 2004 4:16 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Cisco 7940 Phones as paging system? Hey Joe, Could I get a sample config for playing some intro tones on the intercom? I have the same thing but nobody is using it now because they are afraid of having someone call in and "listen in" so we need some way to announce the incoming intercom call. Thanks, MitchelJoe Antkowiak wrote: I am currently using 7960's with *, and line 6 is set to auto answer. Worksgreat, customer is happy. As far as an intro-tone, you can set the dialcommand to play a sound (using the announce option) before the call isconnected. I grabbed a simple tone wav file, and made it play that. Now,when the intercom ext is called, it plays the tone on the destination phone,and wa-la, intercomSo it works. Let me know if you need sample configs.-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of Philipp vonKlitzingSent: Friday, May 07, 2004 12:57 PMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system?Hi! able to support intercom/paging. Having searched the archives, it appears that this question was asked about 6 months ago, and the answer was that the Cisco phones support this using SCCP and having one line set to auto-answer, but at the time this was not supported in the SIP image. Is this still the case?Dunno about Cisco, but wanted to let you know that the recent Grandstream firmware (.55 and later) now also has an auto-answer option. Still I guess I should mention that the microphone of the GS phones in speakerphone mode is far from a brilliant implementation (- echo for the remote speaker talker, and too thin sound from the person in the room).Cheers, Philipp___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Win a $20,000 Career Makeover at Yahoo! HotJobs
Re: [Asterisk-Users] Fax Over VoIP
g711ulaw - Original Message - From: Michael Shuler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 14, 2004 7:01 PM Subject: [Asterisk-Users] Fax Over VoIP Anyone know what protocols support a fax machine i.e. g.729, g.711, etc? Michael Shuler ___ 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] Bug with 'r' in dial
The lastest CVS's versions (both stable and head), the 'r' option in app_dial doesn't work with SIP and Re-invites. I've heard reports that it's not working with IAX2 either.. I'm using Cisco gateway and cisco ATA's and I am doing re-invites, and it's worked up till this point.. What's going on? Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [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] Several H323 bugfixes - working SIP - H.323 translator
I just tried this, and it's not working for me.. I can't call a 2600 or a
CCM... What version of OpenH323 and PWLIB did you all use?
- Original Message -
From: Marian Durkovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 10:35 AM
Subject: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323
translator
Hi all,
in an effort to create a SIP - H.323 translator we've found and fixed
several problems in H.323 channel. These inlcude:
for SIP-H.323 calls
- no ringback tone
- ringback not related to H.323 events
- one-way audio with Cisco CallManager
- incorrect Caller ID
for H.323-SIP calls
- not able to establish call with Cisco IOS 12.3(4)T
- ringback not related to SIP events
- no support for 183 Call Progress
- incorrect Caller ID
Please find the patches against aterisk 0.7.2 release below.
M.
--
Marian Durkovic network manager
Slovak Technical University Tel: +421 2 524 51 301
Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351
812 43 Bratislava, Slovak RepublicE-mail/sip: [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
[Asterisk-Users] Codec negotation with re-invites..
I'm about over this.. okay,, here is what I got.. [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = inbound ; Default for incoming calls tos=lowdelay tos=184 disallow=all; Disallow all codecs allow=ulaw [gateway] type=friend host=1.1.6.9 canreinvite=yes qualify=yes dtmfmode=rfc2833 context=default disallow=all allow=ulaw allow=g729 [sipphoneg729] type=friend secret=password nat=yes host=dynamic canreinvite=yes qualify=200 context=longdistance-g729 dtmfmode=rfc2833 mailbox=2199 disallow=all allow=g729 [sipphoneulaw] type=friend secret=password nat=yes host=dynamic canreinvite=yes qualify=200 context=longdistance dtmfmode=rfc2833 mailbox=2199 disallow=all allow=ulaw okay, when I place a call from sipphoneulaw to the outside world via gateway, everything works fine.. If I place a call from sipphoneg729, it doesn't work.. One leg to the gateway will be ulaw, the leg to the phone will be g729, and, I have 1 way audio.. The sip phone can hear anything from the gateway, but, the gateway can't hear the phone. I've even went as far as to setup a seperate context for the g729 phone and do this.. ,SetVar,SIP_CODEC=g729 which, says it sets it to g729, but it's still a ulaw call.. Guys, this is a real problem... We're going be doing mixed configs.. and if a gateway says it can do both, and phone says it can only do one... then we should be using the compatable codec... PLEASE help.. This is going to cause problems in our rollout. Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [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] Codec negotation with re-invites..
Okay, I add allow=g729 into the [general] section of sip.conf... I can now place calls via ulaw or g729 without any problems.. simply by setting the allow= in the phone's sip entry.. However, INBOUND is a whole nother problem... I get a really strange buzz sound on inbound calls.. and... here is a snippit of show sip channels while the call is in progress.. 1.1.1.24 8659342199 505b634c5cc 00103/0 0ms ms ULAW 1.1.1.29 8656914260 B392830B-17 00102/00102 0ms ms G729A .24 is the sip phone, .29 is the gateway I'm totally lost on this. - Original Message - From: Alex Volkov [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 12, 2004 3:23 PM Subject: Re: [Asterisk-Users] Codec negotation with re-invites.. Sounds to me that your asterisk first negotiates g729 with your phone, then negotiates ulaw with the gateway (since it *is* the preferred codec in your config), and on a re-invite the logic breaks up either in the phone or in the gateway (or perhaps in the asterisk itself, I am not absolutely clear on the details of re-invites). Try changing the order of codec preference for the gateway and see if that fixes your g729 phone and breaks the ulaw phone at the same time. Alex. - Original Message - From: Billy Huddleston [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 12, 2004 2:02 PM Subject: [Asterisk-Users] Codec negotation with re-invites.. I'm about over this.. okay,, here is what I got.. [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = inbound ; Default for incoming calls tos=lowdelay tos=184 disallow=all; Disallow all codecs allow=ulaw [gateway] type=friend host=1.1.6.9 canreinvite=yes qualify=yes dtmfmode=rfc2833 context=default disallow=all allow=ulaw allow=g729 [sipphoneg729] type=friend secret=password nat=yes host=dynamic canreinvite=yes qualify=200 context=longdistance-g729 dtmfmode=rfc2833 mailbox=2199 disallow=all allow=g729 [sipphoneulaw] type=friend secret=password nat=yes host=dynamic canreinvite=yes qualify=200 context=longdistance dtmfmode=rfc2833 mailbox=2199 disallow=all allow=ulaw okay, when I place a call from sipphoneulaw to the outside world via gateway, everything works fine.. If I place a call from sipphoneg729, it doesn't work.. One leg to the gateway will be ulaw, the leg to the phone will be g729, and, I have 1 way audio.. The sip phone can hear anything from the gateway, but, the gateway can't hear the phone. I've even went as far as to setup a seperate context for the g729 phone and do this.. ,SetVar,SIP_CODEC=g729 which, says it sets it to g729, but it's still a ulaw call.. Guys, this is a real problem... We're going be doing mixed configs.. and if a gateway says it can do both, and phone says it can only do one... then we should be using the compatable codec... PLEASE help.. This is going to cause problems in our rollout. Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [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 ___ 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] Stange notices and Warnings..
I've been getting a few strange things latly.. Mar 1 14:44:07 NOTICE[1142106560]: chan_sip.c:5585 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '63.169.60.253' -- Registered SIP '2767069017' at 63.169.60.253 port 5060 expires 120 This is a Cisco ATA running 3.1, I've got several others, but this one is the only one that does it... Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [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] Problems with ATA's locking up..
That's just it, I'm not doing anything.. Just normal use.. as far as I can tell, they end up locking up with or without anyone using them as far as I can tell.. Thanks, Billy - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 12:08 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. Hi, Citeren Billy Huddleston [EMAIL PROTECTED]: Using them with * and a Cisco GW, they're on PUBLIC ip's, No Firewall or anything, I am using re-invites. Pretty standard setup. When they lockup, you can't ping them, or get to the http interface, and I even think the IVR stops responding when you push the button. Yes, but is anything specific happening when they hang ? (What are you doing that seems to cause the hang ?) I have pretty similar setups, so I could try to recreate your scenario ? Florian ___ 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] Problems with ATA's locking up..
http://www.nxs.net/cisco_ata_186.htm - Original Message - From: CW_ASN [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 12:40 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. Could you share your 3.0.0 config? - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 2:10 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. Hi, Citeren CW_ASN [EMAIL PROTECTED]: 3.0.0 have some problems. Sometimes, ata answers to invite with Not found or Busy here. This is a strange behavior. I'm using now 2.16.2 Hm ? I have not seen this happening yet. 2.16 has alternative behaviour regarding flash transfers... Florian ___ 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] Problems with ATA's locking up..
Anyone had any problems with ATA's running 3.0 software locking up? Thanks, Billy +--+ | Billy HuddlestonSenior System Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [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] voip phones
and with the HT-286 you get a Chinaman in a box! :) - Original Message - From: Michael Koehler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 5:45 PM Subject: Re: [Asterisk-Users] voip phones I prefer SIP Phones, Grandstream BT-100 IP-Phone or the HT-286 Analogue telephone adapter Why? - brilliant user interface, with or with out a web browser - cristal clear voice even with low band codecs - PPP over ethernet (PPPoE) aware - continual firmware improvement - plenty of tweak options - economically priced - protocol conform - made in china - fast shipping Retail from $39 to $245 .. google is your friend. Tim Sailer wrote: What is the best inexpensive voip phone out there? I want to try a few with *, but don't want to go broke while I'm just playing around... Tim ___ 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] canreinvite and codec negotations...
Okay, now on to my problem.. I have people who will be using ulaw, and I have people who will be using g729.. I want to set it up so that canreinivte will work.. I have a single cisco gateway.. Asterisks isn't handling the negotation between the 2 devices very well.. For example.. [gateway] type=friend host=1.2.3.4 canreinvite=yes qualify=200 dtmfmode=rfc2833 context=default disallow=all allow=ulaw allow=g729 [123] type=friend secret=abc nat=yes host=dynamic canreinvite=yes qualify=200 context=default dtmfmode=rfc2833 mailbox=2199 callerid=Joe Blow 123-456-7890 disallow=all allow=g729 [321] type=friend secret=abc nat=yes host=dynamic canreinvite=yes qualify=200 context=default dtmfmode=rfc2833 mailbox=321 callerid=Joe Blow 321-456-7890 disallow=all allow=ulaw Okay, in this configs, gateway would be my cisco 26xx gateway.. ext 123 would be a g729 customer.. and 321 would be a ulaw customer. When someone calls ext 123, the cisco send the call to asterisk, it *WILL* use ulaw... then it will initiate a call to g729, well... Now we have a codec mismatch, and canreinvite won't work... EVEN though gateway can do g729.. ext 321 won't have any problems.. It'll work fine for them.. What can we do to get this to work like it should? Thank, Billy +--+ | Billy HuddlestonSenior System Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [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] G.729 Licenses from Digium
IDE/SCSI interfaces, SCSI only installed, WITH IDE CDROM installed with CDROM in drive. - g729 WILL WORK. I'm running a system right now with 24 licences.. Tested it with a single license before purchasing the other 23. You MUST have a CDROM in the drive when you run the install program.. and MUST have it in the drive when you bring Asterisk's up. Thanks, Billy - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 5:28 PM Subject: RE: [Asterisk-Users] G.729 Licenses from Digium zoa wrote: This is absolutely not true. I have 3 (raid) scsi asterisk machines in production. Joachim. At 11:32 21/01/2004 -0500, you wrote: In my view at least one IDE drive must be installed in order for * g729 license to work. To simplyfy, here is the matrix (This is how I think it is please confirm) IDE Disk Install - g729 coder work. IDE/SCCI interfaces. Only a SCSI disk installed - g729 will not work. IDE/SCSI Interfaces. At lease one IDE disk installed - g729 will work. SATA Serial ATA Disk I have no clue how it works. Is SATA considered a IDE disk or a SCSI disk ? This is an issue that VoiceAge need to address soon. - SamW -Original Message- From: Amaury Jacquot [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 4:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 Licenses from Digium Terence Parker wrote: OK - but what counts as a SCSI system? These days there are lots of pseudo-SCSI systems around - such as our server which runs a serial-ATA RAID but the driver is loaded as a SCSI device. Is that still IDE? Or SCSI? technically, it uses the SCSI command set over a serial link, so, it's SCSI Terence I know one thing for sure... G729 WILL NOT WORK after installation *(it never realy installs but does the segmentation faults), * will not start, and you will need to prevent g729 module from Starting in order for * to start. So do not buy if your box is SCSI in any part. Ta SJ ___ 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 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 Can you please clarify which part are you referring as not being true? ___ 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] Handytone 286 - calling out
I've not had ANY problems using info OR rfc2833.. I did have problems using inband. Try switching to it and see how it works.. I NEVER had a problem with double digits, and, I believe that the reference to GS phones having that problem with * was retracted. Thanks, Billy - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 26, 2003 4:14 AM Subject: RE: [Asterisk-Users] Handytone 286 - calling out Billy Huddleston wrote: change dtmf to info on both * and in the handytone. - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 25, 2003 8:01 PM Subject: [Asterisk-Users] Handytone 286 - calling out Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just hangs in there. ATA is behind NAT, registers to an * with public IP with no problems and it uses 1.0.4.17 firmware. Web config screen has detected firewall/NAT type is open Internet as network firewall. Here is my sip.conf: [2202] callerid=HandyTone 2202 username=2202 context=intern qualify=500 type=friend secret=XX host=dynamic dtmfmode=inband canreinvite=no reinvite=no disallow=all allow=ulaw allow=alaw Any suggestions/pointers will be appreciated. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users My understanding from this months GS related posts is that info is not sending the digits properly. Is that the case with you? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Handytone 286 - calling out
change dtmf to info on both * and in the handytone. - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 25, 2003 8:01 PM Subject: [Asterisk-Users] Handytone 286 - calling out Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just hangs in there. ATA is behind NAT, registers to an * with public IP with no problems and it uses 1.0.4.17 firmware. Web config screen has detected firewall/NAT type is open Internet as network firewall. Here is my sip.conf: [2202] callerid=HandyTone 2202 username=2202 context=intern qualify=500 type=friend secret=XX host=dynamic dtmfmode=inband canreinvite=no reinvite=no disallow=all allow=ulaw allow=alaw Any suggestions/pointers will be appreciated. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] strange SIP authentication/authorization behaviour
loose username=ipphone9 Not needed.. the [109] is really the username - Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 24, 2003 11:42 AM Subject: [Asterisk-Users] strange SIP authentication/authorization behaviour When I have an ip hardphone username setup in my sip.conf : [109] type=friend username=ipphone9 secret=bla-la host=dynamic dtmfmode=rfc2833; Choices are inband, rfc2833, or info defaultip=172.20.0.139 mailbox=109 ; Mailbox for message waiting indicator callerid=ipphone9 109 callgroup=1 pickupgroup=1 and this user has a wrong password then calls are denied, but when I just change the userID on the phone to a nonexistant for example 110, the calls go through ! though I see on the console messages about wrong authentication. I`m running a CVS version from friday. Thanks, -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
Use CIPE, It's a UDP based VPN solution. - Original Message - From: Alastair Maw [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 20, 2003 1:37 PM Subject: Re: [Asterisk-Users] tunnel iax via gnophone with ssh? On 20/11/03 15:44, Chris Hirsch wrote: Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT. Anybody have any ideas? Yes, IIRC SSH only tunnels TCP. IAX is UDP based. You'll need to find something that will tunnel UDP over TCP, so you can tunnel that over SSH (!). Good luck. :) Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snatching calls
how could you do this with sip and VOIP? - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 04, 2003 5:40 PM Subject: Re: [Asterisk-Users] snatching calls On Tue, 2003-11-04 at 15:28, Shoval Tomer wrote: Hi, Our current PBX (Panasonic) has a setting that enables users to snatch calls ringing at other extensions. I'm not sure snatching is a correct term for this so let me elaborate. Let's say you sit in a room with five other people. Each one has it's own extension. One person goes out. As soon as he leaves the room his phone starts ringing. The other guys in the room want to answer his phone for him and take a message, but they won't get up for it. I just pick up our extension, hit *40, and the call is automatically transferred to my extension. Is this doable with Asterisk? Is it possible to divide extensions into groups like a group per room so I won't snatch a call from another room in the building, that I wasn't even aware was ringing You want to look into call groups and pickup groups. To pickup the call you use *8#. from /usr/src/asterisk/configs/zapata.conf.sample ; ; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing ; and it is a member of a group which is one of your pickup groups, then ; you can answer it by picking up and dialing *8#. For simple offices, just ; make these both the same ; callgroup=1 pickupgroup=1 -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
