Re: [asterisk-users] considering purchasing a t1 card, any recommendations?
I've been using the Digium TE110P for more than a year now without any hitches in a mission critical environment. It's in a Dell Poweredge 750. I had a problem compiling libpri at first and the digium help desk was more than helpful. I also had some echo issues, but those have mostly been resolved using the built in cancellation routines. -- Tom Hayden On 10/20/06, R.R. Libera [EMAIL PROTECTED] wrote: Wow, this is a completely neutral and very valuable review. Thanks a lot Zoa. I´m an * newbie; my little box will only needs 20 extensions to give termination to remote users and I´m about to buy a PRI interface; I decide to get Sangoma hardware.. a lot of people recommended it to me. In you review you said: The biggest choice you need to make is if you want onboard echo cancellation or not. How can I really know if I will need echo cancellation? I´m planning to get a single span card (which doesn´t include echo cancellation) but how can I know if I really need this feature?? Thanks in advance. R.R. Libera Zoa escribió: I think the recent Digium and Sangoma cards are quite similar. (and about the same price) I didn't try sangoma so far, never had any issues with the digium cards, I have no clue how the digium helpdesk is, i never needed to call them. (well not really correct i did call them once, years ago for a firmware problem with their first te410p revision, causing a crash once every few months they had the distributor send me replacement cards right away, before i returned the old ones, so that i could swap them without having to shut down the server for a week). Configuration and installation for the cards is pretty straightforward, all you need to do is compile the kernel modules for your kernel. I personally installed at least 20 digium pri cards, all on different hardware without problems related to the digium hardware. (sometimes i did have bad cables, bad pri's, oh and my embedded pc didn't provide enough power for FXO ports). You will probably find more people on the list with problems with digium than people with problems with sangoma. This might be because a lot more people seem to use the digium cards with asterisk than sangoma cards with asterisk. (Based on the people i speak to, i'd guess 1 to 5% use sangoma?). The biggest choice you need to make is if you want onboard echo cancellation or not, you might not need it and if you want it its going to cost you a lot more than without. (both for sangoma and digium hardware). - They both seem to use exactly the same Octasic echo cancellation module. If you need on board echo cancellation but don't need 4 ports, digium is the only choice with their 2 port card with Octasic echo cancellation module. (Afaik sangoma doesn't have such a 2 port board with on board E.C. but i could be wrong.) Btw, there are more options, dialogic has compatible cards and so does eicon. (you will need deeper pockets though, the eicon retails at +/- 12000 euro for a quad span i think - people who buy these for asterisk usually do so for hardware faxing or interconnection to different carriers at the same time.) Some people prefer digium over sangoma because they sponsor the asterisk development that way. I'm not one of them, i buy digium cards (or tell my customers to buy them) because i'm happy with their product. Dislaimer: I know some of the people within Digium quite well, so maybe i get exceptional support or they ship me handpicked gold plated, overclocked versions of their cards (not really since i just buy them from a reseller). Cheers, Zoa. Dovid B wrote: Can I now 5th it ? All this makes me wonder why Digium dosent work harder. I have mainly only seen others praise Sangoma over Digium. - Original Message - *From:* Tom Vile mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Wednesday, October 18, 2006 4:22 PM *Subject:* Re: [asterisk-users] considering purchasing a t1 card,any recommendations? I 4th it. On 10/18/06, *Matthew Thompson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 17 Oct 2006, at 22:09, Richard wrote: I would have to second the Sangoma buy. Their tech support is second to none and more then helpful. I've never had any problems with their products that wasn't my own fault. Thirded - I've just done another install with a Sangoma A102 - the setup guides you through all the way and takes no more than 30 minutes (Including recompiling zaptel, which it does for you) [EMAIL PROTECTED] :o) -- Matthew Thompson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http
Re: [Asterisk-Users] SE Michigan asterisk users group
Count me in, my office is in Livonia, but I currently reside in the D. Someone should set up a mailing list for this.--Tom HaydenOn 6/26/06, Michael George [EMAIL PROTECTED] wrote:Our main office is near Lansing, but we have a person who lives in the AA are that would like to attend such a group.On Thu, Jun 22, 2006 at 04:27:02PM -0400, BerkHolz, Steven wrote: I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley.---MThere are 10 kinds of people in this world:Those who can count in binary and those who cannot.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I am looking for a webphone on MY SITE
What would AJAX have anything to do with installing a softphone on your website? I think you need to be a bit more explicit? Are you looking for something that visitors to your website can use to call you?Kudos on throwing around the buzzword, though. --TomOn 4/26/06, Jim Houser [EMAIL PROTECTED] wrote: I need the same exact thing.Our site is almost all Perl with a little PHP.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of RonaldWiplingerSent: Wednesday, April 26, 2006 7:41 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] I am looking for a webphone on MY SITEI am looking for a way of not to install a softphone, preferable as a linkon a web site to a webphone on MY SITE !!!Has anybody an idea for that? AJAX? byeRonald Wiplinger___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersThis e-mail and any attachments may contain confidential and privileged information.If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal.Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on phones...
As far as I know, the Polycom's don't have any kind of echo cancellation for this type of thing, however there is a technology called a FICUS PLANT, which inhabits many offices and can solve your bare office problem :) --TomOn 1/11/06, Carlos Chavez [EMAIL PROTECTED] wrote: I am having a bit of a problem with several phones (Polycom 601 and Aastra 9133i). I have a new installation in a brand new office. The office is bare and there is a lot of echo. This causes all the phones on the office to have a very audible echo. I know it is not really a hardware problem, but is there a setting on the phones to handle this kind of echo? -- Carlos ChavezDirector de TecnologíaTelecomunicaciones Abiertas de México S.A. de C.V.Tel: +52-55-91169161 Ext 2001 -BEGIN PGP SIGNATURE-Version: GnuPG v1.4.1 (GNU/Linux)iD8DBQBDxUAhVhw7eWImqUMRAt9fAJ9xe/8L3PmvXwxMi4AloiO4rSEg/wCgqUYB46L37C91W4DP+cwGpATOktk==lceK-END PGP SIGNATURE- ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Altering Incoming CallerID
Why aren't you using the SetCallerID() cmd?
--
Tom
On 12/1/05, Hugh L. Johnson [EMAIL PROTECTED] wrote:
What do I need to do to alter incoming CallerID? The below isn't
working...
Running Asterisk 1.2 CVS HEAD
exten = NXXNXX,1,Wait(1)
exten = NXXNXX,2,Set(CALLERID(name) = Fred)
exten = NXXNXX,3,NoOp(${CALLERID(name)})
-- Executing Wait(IAX2/A-9, 1) in new stack
-- Executing Set(IAX2/A-9, CALLERID(name) = Fred) in new stack
-- Executing NoOp(IAX2/A-9, Hugh Johnson) in new stack
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Re: [Asterisk-Users] format
Youch. That's quite the switch! I'm surprised you couldn't HEAR the difference. :) -- Tom On 11/30/05, Steve Totaro [EMAIL PROTECTED] wrote: I think if you type show codecs in the CLI you can see what codecs are what by the number. It shows that you tried for g728 but got iLBC. -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 30, 2005 8:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] format Can anyone tell me what this line means? -- Accepting AUTHENTICATED call from 202.125.42.141, requested format = 256, actual format = 1024 does this mean a certain codec was requested but another one was delivered? Is there some configuration that I can make to improve the call quality? Currently my IAX2 Outbound trunk looks like; allow=ilbcg726ulaw auth=md5 context=from-pstn disallow=all host=202.125.42.252 qualify=3000 secret=82XXX type=friend username=0960XXX tia, Dean ___ --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 ___ --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] Best Communications Line for VoIP
If this is something mission-critical, I would *highly* recommend going with a voice T1 from your telco, especially for 20 lines, when a full T1 is 23b channels. I mean, you can do a hosted PBX type configuration, as well. So much depends on the specific situation. -- Tom On 11/22/05, Andy Kuo [EMAIL PROTECTED] wrote: If you need to make calls in and out to the PSTN, you need a T1/PRI. Unless you send the calls to other VoIP provider. Andy On 11/22/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: We are putting in an Asterisk VoIP solution and was wondering what the best communications medium would be for this implementation. We are going to need 20 telephone lines in/out of our business. We currently have a data T1. Could we put another data T1 to use for Asterisk, or would it be better to put in a Voice T1 or a PRI line? Also, when we do put this T1 or PRI line in, what would be the best equipment to use with the Asterisk box? Any other recommendations would be appreciated? Thank you, Jyran Glucky Advisory Programmer BlueWare, Inc. Strategic HealthWare Solutions 3060 W. 13th Street Cadillac, MI 49601 Phone: (231) 779-0224 ext. 111 Fax: 231-779-1002 mailto:[EMAIL PROTECTED] http://www.blueware.net DID YOU KNOW? BlueWare is the Grand Prize Winner of the 2005 IBM Beacon Award BEST DB2 (Document Management) Application Worldwide. BlueWare Market Share for Hospital Document Management Systems is in 25 states in the US. ___ --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 -- Tom ___ --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] Re: Anyone aware of a current Dell server model with 3PCI slots
Yeah, with our Dell Poweredge 750, we had all kinds of IRQ conflicts and whatnot. I booted up and in the BIOS I turned off all sorts of devices, including one of the ethernet cards, the USB, serial, etc. After that, things worked much better. -- Tom On 11/2/05, Matt [EMAIL PROTECTED] wrote: We have a poweredge 2850 that we use for our VoIP server and it has 3 PCI slots. I'm greeting to hear this. I have installed some Digium cards into this kind of servers. I get surprised when the slots pci gets shared IRQ with ethernet devices, raid controller or VGA card. Anybody knows how get unshare the IRQ of the slots pci ? (firmware, update, some special BIOS configuration,...) We answered Dell with no response. I can't say that I've had this problem with the 2850 we have. I also can't take the server down to look at it right now, however we just got another digium card which I need to put in at some point over the next few days, so I'll be taking it down sometimes soon. As far as sharing, make sure you have disabled everything you don't need USB, SERIAL, PARALLEL, etc. You can then set the PCI IRQ in the BIOS, I believe. CPU0 CPU1 0: 584222710 584230408IO-APIC-edge timer 1: 0 7IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 8: 0 1IO-APIC-edge rtc 14: 0 2IO-APIC-edge ide0 38:67120198751310 IO-APIC-level megaraid 48: 318573642 37 IO-APIC-level eth0 77: 1014625786 2080170691 IO-APIC-level t1xxp ___ --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 -- Tom ___ --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] RE: [Asterisk-biz] Asterisk as a Voice ConferenceServer
Can I ask what kind of trunking you are using for the calls? Zap/SIP/IAX? -- Tom On 11/2/05, Cullin J. Wible [EMAIL PROTECTED] wrote: We used to run a conference server on a PII 400Mhz with 512MB of RAM. We had 2 separate conference rooms with 15 users each (30 simultaneous) calls with no problem. We have since upgraded it to a P4 2.0Ghz with 1GB of RAM (just because it was getting old) and it still works just fine with even higher call volumes. No degradation of quality either that we can see. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Wednesday, November 02, 2005 2:51 PM To: Iain Barker Cc: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] RE: [Asterisk-biz] Asterisk as a Voice ConferenceServer Iain Barker Wrote: - Our experience with over 10 or more participants in a single Asterisk conference was that quality degraded quite rapidly. Is this really true as there were many in this list who had confirmed that they have used the conference bridge for a lot more connections than what you have Suggested as the upper limit. Logically the conference bridge should work at the same capacity as the number of calls Asterisk can handle in a given configuration. Though your solution looks impressive and probably is the best for upto 30 simultaneous calls, I am more interested in knowing what it takes for Asterisk to be able to handle the 100 channels I need to run Simultaneously. Seshu Kanuri -Original Message- From: Iain Barker [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 02, 2005 1:41 PM To: Kanuri, Seshu (Company IT) Subject: Re: [Asterisk-biz] Asterisk as a Voice Conference Server Seshu, Our experience with over 10 or more participants in a single Asterisk conference was that quality degraded quite rapidly. The solution was a dedicated hardware bridge for conference mixing http://www.aastra.com/enterpriseip/pro_238.asp Kanuri, Seshu (Company IT) wrote: I am working on a bid for a New York State requirement where we need to provide access to 100 Simulataneous Investors to get into a conference with the Pension Funds Officer for discussions. As you might have guessed it, I am presenting an Asterisk enabled Conference solution. One of the Bid requirement is to provide three verifiable references who have implemented a similar voice conference solution for more or less 100 simultaneous calls, with a possible recording of the entire call. If anyone has implemented this on a commercial scale, I am looking for referrals at this time, and a possible co-operation in future. I would appreciate if you can send me your name, contact Info, company name and a one para description of the solution and the name/type of client whom/where this solution is running at this time. A couple of minutes of your time is needed when the guys at Albany may like to speak to you for a confirmation that Asterisk is real and it can do the 100 people conference, what they are looking for. I do thousands of conferences a day using asterisk as the backend, most are in the 5-50 user range, but many are in the 150+ range. (but, I use app_conference, not app_meetme for them). I can give you my contact information off-list if you want it. -SteveK NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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 -- Tom ___ --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] I give up - Help with TE410P
Why don't you call their support? I've called and only had a good experience. Tech Support via email is always kind of weak no matter where you go. Call them, and go through their tech support department, they have some really intelligent and knowledgeable techs down there and I'm sure they'll be able to fix this problem. -- Tom On 10/29/05, Bart Fisher [EMAIL PROTECTED] wrote: Well, have you ever tried their support? They assume we are all dummies... A bunch of canned email messages to remind you to plug in the power cable. :) Ok, in a disparate act (and this might help someone body someday) I removed all the Digium card and emptied the zap*.conf files from the box and rebooted. I allowed Linux to remove the missing cards - this of course installs ztdummy. Next I shutdown and added all the cards at one time. - Booted and let Linux discover cards and allowed configuration. Copied back my zap*.conf files rebooted. This time it comes up 6 spans with green lights and 2 on first card with flashing red. I shutdown, and swap the two TE410P. Rebooted - all light green now. Since it's working, I'm done - but only go to show you these cards are flaky. Bart - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 3:35 PM Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 18:19, Bart Fisher wrote: Yep - that was easy part :) and these are T1 (D4, AMI, SF, and EM Wink) BTW Ok, well I'll go for the obvious question: have you contacted Digium technical assistance? You have paid for support within the price of the card. -A. ___ --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 -- Tom ___ --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] Middle Ground between POTS and T1?
I use a partial T1 as well (12B + 1D). Most CLECs offer them. -- Tom On 10/18/05, Goran Skular [EMAIL PROTECTED] wrote: Here (Croatia) is also possible to get partial E1 (PRI/PRA)... from one of our telcos (DT T-com) we can get PRA in 10 increments: 10B, 20B and 30B We have a partial T1 (5B + D, iirc) from Allstream - there may be a provider in your area that does something similar. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 17-Oct-05, at 12:48 PM, Matthew T. O'Connor wrote: I was wondering if there was a middle ground between POTS lines and a T1. I have a new office with a T1 line and while it's working well, it's a lot of money and we will never use anywhere near 23 lines at one time. Is it possible to get a few ISDN lines or something and bundle them together? Basically I would like to get the digital features of the T1 PRI (DID number, etc...) but smaller. Thanks, Matthew ___ --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 -- Tom ___ --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] what should i select ??????????
Well, Ishtiaq, to build on what Mark says... First, I hate to be a grammar nazi, but you should use better grammar in your emails. It looks very unprofessional using the word 'ur'. Moving on, if this is a new install, which it appears to be, I would do it *right* and put the solid investment into it the first time, so you're not replacing the hardware early or doing constant maintenance/upgrades. Alot of what you are asking depends on the circumstances. I mean, are you going to have PRI's between the different offices? How do you plan to terminate your calls to the PSTN? I personally, would put in SIP hardphones of the Cisco or Polycom variety and have each of your offices connect via IP to a central location for routing of PSTN traffic (whether you terminate calls to the PSTN or use a VoIP provider is another argument). However, it all depends on the amount of traffic you handle already, the type of connections you're looking at and a lot of other variables. Frankly, if you're deploying a network of this size, it's a mission critical assignment, and you don't have a whole lot of telecom/voip experience, I would look at hiring one of the consultants on this list to get you going in the right direction. It's worth the investment. -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 10/13/05, Mark Phillips [EMAIL PROTECTED] wrote: I've done exactly this recently. Frankly with hardphones being as cheap as they are I'd buy them. If you are messing about with analogue adapters etc you'll end up with all sorts of potentional echo problems not to mention the cost of the chanel banks etc. A hardphone will enable you many features that an anologue phone would not. Many hardphones have LCD displays and soft keys which you can assign funtions to such as company directories etc. This is possible in the Analogue world through the use of ADSI but an ADSI compliant phone is often more than a hardphone. There are of course network considerations. Can your network sustain the extra traffic the hardphones will add? Do you have the right physical layer at every station (usually CAT5)? Do you want to do Power Over Ethernet or run a small PSU to the wall for each phone? Only you can answer these questions. For my money, witha new install, I'd go all hardphone. I happen to like the Cisco 7960 which goes for about $260. However there are phones as cheap as $50 in quantity. Check out voipsupply.com for prices etc. Mark ishtiaq Ahmed wrote: hy all actually i want to have a setup of five offices having round about 200 extensions ( each office having 35 to 45 ) which will be connected through asterisk. now either i should go for voip phones( hard phones ). or use any interface card to asterisk server to which the analogue phones will be connected. - if i use analogue phones in the above case ( we have analogue phones already ) which card should i use.( plzz mention the name of card provided by digium ). i think using some interface cards( for analogue phones and one card for each office) will be cheaper than buying about 200 voip phones. what do u think i will be waiting for ur value able suggestion. i have searched alot for this noone has given me a clear suggestion ( mean to say answer at the max 20% of my question ). Yahoo! Music Unlimited - Access over 1 million songs. Try it free. http://pa.yahoo.com/*http://us.rd.yahoo.com/evt=36035/*http://music.yahoo.com/unlimited/ ___ --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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --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 -- Tom ___ --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] Compiling SpanDSP
I've been getting the same problem with the verbose issue. I just commented out the line, and it seemed to compile OK. -- Tom On 10/3/05, Doug Lytle [EMAIL PROTECTED] wrote: Dave Cotton wrote: On Mon, 2005-10-03 at 14:10 -0400, Doug Lytle wrote: Look at rxfax.c around line 88 there's an #if statement remove the references to callerid. This error has been around for a while. That took care of the callerid compile error, but not the verbose error: error: structure has no member named `verbose' Doug -- Ben Franklin quote: Those who give up essential liberties for temporary safety deserve neither liberty nor safety. ___ --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 -- Tom ___ --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] setting up asterisk as an sms central?
Well, it depends what country you're in and what kind of protocols you are using. Here in the US, I prefer to *not* use asterisk and use the perl module Net::SMPP to handle my SMS traffic between my gateway/aggregator and the carriers SMSC. It's somewhat easier to configure with special services, and database application. You could also take a look at a peice of software called Kannel, which handles SMS aggregation. However, if you are in Europe you should probably be able to get a fixed-line SMS service, and use the Asterisk cmd SMS: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Sms Cheers, -- Tom, On 9/29/05, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: yes On 28. sep. 2005, at 15.54, Tom Hayden wrote: You're going to need to explain a little more. When you say central are you talking about an SMSC? -- Tom On 9/28/05, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: hi is it possible to use asterisk as an sms central to send SMSes directly to clients on PSTN instead of just communicating with a central? the telco to which we're currently connected doesn't have a central roy ___ --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 -- Tom ___ --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 -- Tom ___ --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] Asterisk Echo problems, Urgent, please help,
What kind of POTS trunks/cards are you using? -- Tom On 9/29/05, Ian Bonham [EMAIL PROTECTED] wrote: Hi all, I hope someone can help, as I have an urgent problem. I've got a production Asterisk server thats been deployed, but we are seeing a strange voice echo problem. There is about a 250ms echo for the users in the office, and they are hearing their own voice back at them. I'm running the CVS Head code, on RH9.0. This is on a P4 box with 2gb of memory. The client SIP phones are Polycom Soundpoint IP600's, WiFi ZyXel 2000w handsets, and X-Lite (free) PC clients. All see the same problem. There is a bridge into the POTS (BT's SystemX) using a Voicetronix OpenSwitch12 card and the vpbhp driver. The echo occurs on both SIP-POTS calls, and SIP-SIP calls. I've tried a number of volume adjustments to correct the echo but it is always the same. If anyone has any ideas I'd really appriciate some help, as this is a major urgency, Many many thanks, Ian Bonham _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ --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 -- Tom ___ --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] Re: [Asterisk-biz] Problem with sending fax froma SIP extension
Well, I think what he means is that it's not VoIP, because you are using TDM on both ends. It looks like this: fax machine - TDM - * - TDM - PSTN If you had a SIP ATA attached to a fax machine, you would be using VoIP. That would look like this: fax machine - SIP/VoIP - * - TDM - PSTN I have found using faxes over any VoIP to be *extremely* unreliable. SpanDSP works pretty good, but you won't have much luck using a fax SIP extension. Cheers, -- Tom On 9/29/05, Jonathan k. Creasy [EMAIL PROTECTED] wrote: Why is what he is doing different than having the fax machine on a Sipura ATA? Just because both those ports are on the pci card that doesn't make them not Voice in betweenif I'm wrongeh...oh well -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, September 28, 2005 9:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: [Asterisk-biz] Problem with sending fax froma SIP extension Before responding please read my message again, in my message I worte: 'faxing over VoIP' what you are doing is NOT over VoIP. On 9/28/05, u [EMAIL PROTECTED] wrote: On 9/27/05, C F [EMAIL PROTECTED] wrote: 1. Search the archives 2. Search again 3. Now search the internet 4. The fact is that faxing over VoIP without T.38 doesnt really work. Because it works 60% of the time it doesn't mean that it works, until it works 99% of the time. Since Asterisk does NOT support T.38 it doens't support faxing over VoIP other solutions are available, search the wiki. On 9/26/05, Mark Armstrong [EMAIL PROTECTED] wrote: I am having exactly the same problem. Any information would be appreciated. Faxing is a commercial proposition that my team is working on, any limitations of Asterisk or E1 cards needs to be highlighted. Regards Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, 27 September 2005 9:42 AM To: [EMAIL PROTECTED]; Commercial and Business-Oriented Asterisk Discussion Subject: Re: [Asterisk-biz] Problem with sending fax from a SIP extension why don't you post this to asterisk-users? On 9/22/05, Andy Kuo [EMAIL PROTECTED] wrote: Hi All, I'm having problem sending fax from SIP extensions (Linksys PAP2) through Asterisk Zap channels (ISDN PRI). The SIP extensions can receive fax without problems, but sending fax fails most of the time. Does anyone have this problem? Please advice. Thank you. AK I have a fax connected to a zaptel pci card (FXS). Fax are sent right into an FXO card connected to my PSTN. I don't experience problems but then again I am a light user. Just out of curiosity, is the problem related to sending FAX via IAX or SIP to another asterisk server? if asterisk will support T.38 passthru to say Hylax Fax, does that mean I need a Hylax fax daemon in every asterisk server? ___ --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 -- Tom ___ --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] Auto Answer Fax
You can use SpanDSP. http://www.voip-info.org/tiki-index.php?page=spandsp -- Tom On 9/29/05, Rene Nelson [EMAIL PROTECTED] wrote: Can anyone point me to a good howto or example on how to get * to recognize inbound faxes and adjust accordingly? Ideally I would like it to grab the fax and email it to me, but I dont know if that is really possible yet or not. Thanks Neri ___ --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 -- Tom ___ --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] setting up asterisk as an sms central?
You're going to need to explain a little more. When you say central are you talking about an SMSC? -- Tom On 9/28/05, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: hi is it possible to use asterisk as an sms central to send SMSes directly to clients on PSTN instead of just communicating with a central? the telco to which we're currently connected doesn't have a central roy ___ --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 -- Tom ___ --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] asterisk SMS and sprintpcs
Well, you need to be a bit more specific. How are you trying to send it? Are you using an SMSC? What kind of lines do you have? -- Tom On 9/26/05, Jerry Geis [EMAIL PROTECTED] wrote: Does anyone know about sending SMS messages to a sprint pcs phone. Can you give me a few details. Thanks, Jerry ___ --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 -- Tom ___ --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] AGI Script to interact with ACCESS Databse and Set CID info on the fly.
If you used a perl or PHP agi script, you could probably use some kind of ODBC drivers to communicate between the two. -- Tom Hayden On 9/22/05, Tim King [EMAIL PROTECTED] wrote: Well guys here comes the fun part. I have a Microsoft access (VBA) application that interfaces with my SQL database. This app pulls of info from the SQL record and than picks up the phone and dials that locations number. I have purchased a few hundred NpaNxx's for my own use. I want get into too much detail there but no worries this is legal. I need to change my CID info on the fly. So I am thinking it should be easy to make an AGI script that just sets the CID info on a particular line using two variables being passed to it $Line_No to tell it what line to set and than $CID to be the number to set on that extension for that call. It also should be relatively simple to have the access app take a look at the area code and phone number for the location being called and pull a phone number from the NUMBERS table which has all of my numbers in it and pass that over. The real question is how do we get Access to speak to an AGI script. Has anyone done anything like this? Thanks a lot for reading but this will be a fun one. ___ --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 -- Tom ___ --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] Polycom IP500 Quickstart page or files?
Huh? You can easily configure an IP500 via a web browser. Just point the URL to the IP addr of the telephone. -- Tom Hayden On 9/22/05, Wilson Pickett [EMAIL PROTECTED] wrote: Hi, I just got my ip500 back after months of waiting. Is there an easy way to get it hooked up to asterisk without [t]ftp servers and all that or is there a quickstart page somewhere? tia r ___ --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 -- Tom ___ --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] Fax-Email for Hosted PBX
I would agree wholeheartedly with everything Colin just said. I've had extensive experience with SpanDSP and with routing thru to an ATA. Both are touchy and work OK, but not well enough to make the users stop calling me :) Currently, we use SpanDSP and it works OK - although sometimes pages get mangled. I would *highly* recommend just getting a couple traditional POTS lines for faxes - it's still the most reliable way to go and what users are familiar with. You could hook a hylafax box up to those lines and still use a fax-email gateway. -- Tom Hayden On 9/15/05, Colin Anderson [EMAIL PROTECTED] wrote: We use SpanDSP to recieve faxes on our PRI, a couple hundred a day with a failure rate of ~5% which is pretty good I think but enough to tick people off. Always the same fax numbers fail. What I did is have an exception list context that is run just before RxFax. If the caller ID matches a bad machine, the call is rerouted to a regular fax not off of a TDM or ATA but forwarded to a POTS line (actually another channel on the same PRI - love that Adtran!) In my tests, routing thru VoIP or out thru a TDM or ATA just plain doesn't work, or doesn't work enough for it to be usable. You wold serve your customer best by having Asterisk with SpanDSP at the telco demarc rx the fax directly, and avoid shunting the call to an ATA or TDM. -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: Thursday, September 15, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Fax-Email for Hosted PBX Best scenario does not route faxes over the IP network as a VoIP call. You can either use spandsp as a fax on the Asterisk box, (has problems, but the delveloper is behind solving them) You can route the calls to a fax server located in the same colo via tdm. (you can use HylaFax on Linix of any other solution. I have used Fax over VoIP for about 2-3 years. Some machines can't handle it. Others can. Receiving has a higher success rate than sending -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter Sent: Thursday, September 15, 2005 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Fax-Email for Hosted PBX I'm proposing to install an Asterisk PBX at a collocation facility for a remote customer. Each of the customer locations will have an SPA-3000 with the FXO port connecting a POTS circuit and the FXS port connecting a fax machine or red phone. In addition to voice traffic, the customer has a high volume of incoming and outgoing faxes. Would it be possible, using g711 between the SPA-3000 and server, to have spandsp/rxfax receive a fax from the POTS circuit via the SPA-3000? From the locally attached fax machine? (I realize that packet loss will have a adverse effect on fax transmissions.) Would I be better-off attaching the fax machines to a Mediatrix 2102? Any help is appreciated. ___ --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 -- Tom ___ --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] translate letters into digits
Why do you need to write an application for this? Why don't you just make joe extension 563?? -- Tom On 9/13/05, Armin Schindler [EMAIL PROTECTED] wrote: Hi, I was wondering if there is already an application or a simple mechanism to convert the dialed extension into digits if letters were used. I don't know if there is a name for that, I mean the letters on the phone keypad: ABC=2, DEF=3, ... So when I call e.g. JOE, the extension 563 shall be used. Do I need to write my own little application to accomplish this? Thanks, Armin ___ --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 -- Tom ___ --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] Echo cancellation again ...
I have experienced pretty nasty echo on my PRI w/TE110P. The echo was only coming from other POTS lines, because cell phones already have echo cancellation, and other PBX's had the same. I resolved the problem by turning on the AGGRESSIVE option and it works fine now, and we haven't noticed a severe degradation in sound quality - most of my operators were just happy the echo was gone :) -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 8/17/05, Doug Lytle [EMAIL PROTECTED] wrote: Alan Bunch wrote: I have been reading with great interest the posts on trouble shooting echo cancellation with *. Is it just coincidence that all of this discussion has been with analog lines. Are PRI's susceptible to echo problem like POTS lines. Alan, I have experienced echo on our PRI with EC turned off. Granted, it was Asterisk server 1 connecting via IAX to server 2, connecting via a PRI to call my cell phone. Turning on EC removed this echo. Doug ___ 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 -- Tom ___ 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] snom hint
It's in the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom About halfway down the page where it says: SNOM SUBSCRIBE/NOTIFY support for monitoring extension states -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 8/17/05, Gerd Mueller [EMAIL PROTECTED] wrote: Hi list, anybody any example how to use it? I did not find any hint in the wiki nor in the mailinglist archive :-(. I want to use one button showing my agents the actual state (logged in or logged off) Thank you Gerd ___ 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 -- Tom ___ 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] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
Well, it's unlikely you're going to find a PCI card that can handle twenty analog lines, however I suggest you look at purchasing a call bank such as the adit 600. You then can link up your * server with the call bank using a T1 card and control and route calls using that method. -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 8/11/05, Sean Rima [EMAIL PROTECTED] wrote: I have a brief from a local hotel to build a PBX using Asterisk but they want to use their exisiting telephones and wiring from an old PBX that no longer works. Basically, I can build the system but an looking for a card that will allow for upto 20 extensions to be wired into the back of the PC. Doeas anyone know of a solution to this Sean-- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie ___ 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 -- Tom ___ 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] Help with calling Perl AGI interface
I'll second that. Make sure your script is in /var/lib/asterisk/agi-bin and you have the right permissions on it. I really just wanted to reply to your post though to congraduate you, Dan Marino, on your recent induction into the Pro Football Hall of Fame ;) -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 8/11/05, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Dan Marino wrote: I have installed the Perl library from http://asterisk.gnuinter.net/asterisk-perl and am wondering how I reference agi-test.agi from extensions.conf I have added exten = s,1,AGI,agi-test.agi but that doesn't seem to do it. Is there a certain directory .agi files should be, is that the problem? Depending on your asterisk install, the agi-bin directory can be somewhere like /var/lib/asterisk/agi-bin or /usr/share/asterisk/agi-bin locate agi-bin is your friend :) Cheers, Jean-Michel. ___ 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 -- Tom ___ 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] No audio when calling between internal phones
I encountered a similar problem with CVS-HEAD and sip2sip calls between our Polycom IP500s. I attempted to diagnose the problem and there are a few patches on mantis, but none of them worked for me. I flipped back to stable and have had no problems since. Anyone got any ideas? -- Tom On 8/10/05, Gareth Blades [EMAIL PROTECTED] wrote: I am running the latest CVS version of Asterisk. Calls between an IAX client and SIP phones (Grandstream SP2000 and Sipura SPA-841) works fine and so do external call over the Internet from the SIP desk phones. However when I call from either the Grandstream/Sipura phones to another one I get no audio. I have the G711 ulaw codec defined as the preferred on on all phones. Any idea what is going wrong? I am guessing it is something to do with native transfers which is performed in this situation. ___ 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 -- Tom ___ 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] No audio when calling between internal phones
Then perhaps you have a NAT problem or some other issue. -- Tom On 8/10/05, Gareth Blades [EMAIL PROTECTED] wrote: I did try installing the 1.0.9 version but I have the same problem with that release aswell. On Wed, 2005-08-10 at 14:14, Tom Hayden wrote: I encountered a similar problem with CVS-HEAD and sip2sip calls between our Polycom IP500s. I attempted to diagnose the problem and there are a few patches on mantis, but none of them worked for me. I flipped back to stable and have had no problems since. Anyone got any ideas? -- Tom On 8/10/05, Gareth Blades [EMAIL PROTECTED] wrote: I am running the latest CVS version of Asterisk. Calls between an IAX client and SIP phones (Grandstream SP2000 and Sipura SPA-841) works fine and so do external call over the Internet from the SIP desk phones. However when I call from either the Grandstream/Sipura phones to another one I get no audio. I have the G711 ulaw codec defined as the preferred on on all phones. Any idea what is going wrong? I am guessing it is something to do with native transfers which is performed in this situation. ___ 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 -- Tom ___ 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] First PRI
They let you chose your protocol? Nice guys, I've never been asked - just told. I don't know any major advantages between the different signalling formats, though, I don't think there really are any major differences. I've had no problems with ni1 and ni2 with Asterisk. -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 8/9/05, Wiley Siler [EMAIL PROTECTED] wrote: Hello All, I am getting my first PRI installed in a couple of weeks and I wanted to ask for a little advice. I have a single span Digium card I will be using for the install. Id there a benefit to which protocol I use? When asked, I told them to set it up as NI2. The PRI is through MCI and will be used for local and long distance with DIDs and features like CallerID, etc. Any advice would be appreciated. Thanks! Wiley ___ 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 -- Tom ___ 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] FCC to require wiretaps from VoIP providers
Really nothing new. They've done this with wired carriers for years. -- Tom On 8/8/05, Adam Megacz [EMAIL PROTECTED] wrote: Scary. http://www.eff.org/news/archives/2005_08.php#003876 -- PGP/GPG: 5C9F F366 C9CF 2145 E770 B1B8 EFB1 462D A146 C380 ___ 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 -- Tom ___ 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] Switchboards
Well Don, it depends on how you get things setup from your telco. You could get a T1/PRI (or E1) and just trunk all your calls. Then you could have DIDs for each employee along with a primary number, which could receive multiple calls at a time (it's just a DID). If you go with the POTS solution, then you'll need to get a channel bank and do things that way. Frankly, it would be a hell-of-a-lot-easier and cheaper if you got trunked calls. Why buy 23 POTS channels when you can probably get an equivalent number of channels + DIDs for much less money? -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 8/5/05, Don Brearley [EMAIL PROTECTED] wrote: Hello, I am still researching my dive into Asterisk at my workplace, and I was wondering about how switchboard activities are handled.. Right now, a call comes into our switchboard, and the operator forwards them to the appropriate line, thus freeing up the primary number and allowing more calls in. Everyone on campus has a direct-dial line as it is right now. I want to eliminate most of those lines, and switch everyone to extensions instead. If I understand correctly, with Asterisk, i'll need to figure out how many lines are in use at any time (i'll say 20% to be safe) -- so I'll need to have roughly 25-30 POTS lines on standby for inbound calls? The way I see it in my head is, a call comes in on the primary number, and the operator will forward them to the correct extension, and Asterisk will route the call to another circuit, freeing up the primary line. Is this correct? (Sorry for my lack of correct terminology.. still getting familiar!) - Don ___ 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 -- Tom ___ 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] Polycom Phones
Uhmm. Well, he may just be using the typical asterisk configuration of just editing the .conf files rather than using AMP or [EMAIL PROTECTED] For the first issue, that kind of sounds like a problem with the polycom configuration. I don't have my pdf of the polycom config in it with me right now, but I'll bet there is a setting you can toggle to fix that on the boot server files (or maybe even on the phones config). What version are you using? Second, it sounds like you may be having problems with SIP or NAT or registration somehow. I'm curious to see the CLI output on that before I could diagnose it. Hope you like the Polycoms! I love them! I suggest you check out the wiki page on them - it's EXTREMELY helpful at getting things setup correctly with all the great features they built into those phones. -- Tom Polycom Fanboy Hayden Astoria Telecom, LLC www.astoriatelecom.net On 8/5/05, Ariel Batista [EMAIL PROTECTED] wrote: Chris Gamble wrote: Just got in a bunch of polycom phones for use on my shiny new asterisk box, but found 2 small issues I was wandering if someone could help me with. Are you using AMP or Asterisk @ Home? First, though the phones support 2 call appearances, if I am on a call, the second call does not ring through -- it goes to voicemail instead of letting me put the first on hold to talk to the second. Is there a way to fix this? If you are then you need to turn call waiting on * 70 The second is: a lot of my phones will not ring for internal extensions. They show up on the screen as a call ringing in, but the phone itself wont ring. About 50% however do ring. What could cause this? Are the phone registered correctly? What are the settings you have on them. As usual, thank you all for your kind support in getting this far! ___ 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 -- Tom ___ 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] strange dial problem with polycom 501
I've noticed the same issue with my IP500s. When dialing an extension like 106, it will do the 10 okay, then bounce the cursor back to the beginnig, and i'll end up dialing 610. I'm still trying to figure this one out. -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 7/28/05, Michael George [EMAIL PROTECTED] wrote: I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above, if just typed into the phone, might end up: 9507291234. Other times the cursor might jump right back to the beginning of the number. This doesn't happen when they enter the number and the press dial, so it seems to be a digitmap problem. However, the digitmap is nearly the same as what I've used on IP-500s in the past. It is: [0]|*8|*2xx|#5|#7x|[278]xx|[9]11|9411|9x.T [Actually it was [0]|*8|*2xx|#5|#7x|[278]xx|[9]11|9411|9x.T -- I don't know where that space came from, but I'll take it out and test again today.] Are there any obvious problems with that digitmap? Anything else that I should take a look at? 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 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 -- Tom ___ 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] strange dial problem with polycom 501
Michael, I think I got it solved. In my sip.cfg on the boot server make sure that dialplan.impossibleMatchHandling=2 Here's what the SoundPoint manual says about this var If set to 0, the digits entered up to and including the point where an impossible match occurred are sent to the server immediately. If set to 1, give reorder tone. If set to 2, allow user to accumulate digits and dispatch call manually with the Send soft key. Cheers, Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 7/28/05, Bruno De Luca [EMAIL PROTECTED] wrote: try to see if u have set at sip.conf *dtmfmode=rfc2833* Michael George wrote: I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above, if just typed into the phone, might end up: 9507291234. Other times the cursor might jump right back to the beginning of the number. This doesn't happen when they enter the number and the press dial, so it seems to be a digitmap problem. However, the digitmap is nearly the same as what I've used on IP-500s in the past. It is: [0]|*8|*2xx|#5|#7x|[278]xx|[9]11|9411|9x.T [Actually it was [0]|*8|*2xx|#5|#7x|[278]xx|[9]11|9411|9x.T -- I don't know where that space came from, but I'll take it out and test again today.] Are there any obvious problems with that digitmap? Anything else that I should take a look at? Thank you. -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.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 -- Tom ___ 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] Can you caculate with me?
Give 'em a break. You just posted this message (and I presume your support request) seven hours ago. They usually get back to me very quickly. -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 7/28/05, Ronald Wiplinger [EMAIL PROTECTED] wrote: Bob Goddard wrote: On Thursday 28 Jul 2005 13:07, Ronald Wiplinger wrote: before I accuse somebody to overbill I would like you to calculate with me: Rate: 0.0189 for calling Taiwan via NuFone Duration: 930 seconds Lets vote for the answers:0.7269 or 0.2929 ??? Assuming it is per minute; 930 * 0.0189 / 60 = 0.29295 Thanks for your help. NuFone invoiced for me 0.7269 Asking for an answer from them, ... Guess what??? NO ANSWER at all. bye Ronald Wiplinger ___ 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 -- Tom ___ 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] ClueCon in 2 Weeks!
Mine had no problem sending me. I can't wait! -- Tom On 7/24/05, Brian West [EMAIL PROTECTED] wrote: I'll talk to your boss if he has a problem! ;) /b On Jul 23, 2005, at 11:03 PM, Terry Moore-Read wrote: Mine did. [EMAIL PROTECTED] 7/21/2005 2:54 PM Brian West wrote: ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! snip Thanks, Brian West Asterlink.com snip Anyone else think that was a joke at first impression? Good luck convincing the boss to pay for your way to ClueCon ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by Lukins Annis, P.S. NOTICE: This email may contain confidential or privileged material, and is intended solely for use by the above referenced recipient. Any review, copying, printing, disclosure, distri- bution, or any other use, is strictly prohibited. If you are not the recipient, and believe that you have received this in error, please notify the sender and delete the copy you received. Thank You! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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 -- Tom ___ 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] Dell Hardware
I'm using a Dell Rackmount Poweredge 750 with a TE110P and have not had any problems with IRQs or whatever. There is an option in the bios to assign IRQs to PCI cards so it wasn't a problem. I had a few echo issues (I suspect due to the e1000 eth card), but AGGRESSIVE echo cancelation killed them. -- Tom On 7/22/05, Bruno De Luca [EMAIL PROTECTED] wrote: I know. but u can't disable the USB controller always. If u have an server w/ others functions... Bruno De Luca Graziosi DELL computers ussualy has got IRQ conflicts with the USB and slots PCI. If you disable the USB controller from BIOS you get a perfect server. I have tried several PowerEdge 2850 like Asterisk dedicated server and it's running perfectly. I have tried IBM xServer 226 and 346 and the IRQ conflicts (network with slots PCI and with video card) make noises, echos and cuts off . :( Elio Rojano == Avanzada7 -VoIP Departure- http://www.avanzada7.com/ We are using this combination. we are thinking about change the DELL computers! Bruno De Luca Graziosi Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? ___ 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 BRUNO DE LUCA GRAZIOSI Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com CONFIDENTIALITY NOTICE This message and its attachments are addressed solely to the persons above and may contain confidential information. If you have received the message in error, be informed that any use of the content hereof is prohibited. Please return it immediately to the sender and delete the message. Thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ 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] Extension Lights Patch
I've been using the extension lights on my polycoms before that patch, so I'm not sure what it fixed, but I've only seen the lights work on Polycoms and Snoms. Try using the hint priority and see if it works for your gxp2000, be sure to post your results! -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 7/20/05, Anton Krall [EMAIL PROTECTED] wrote: Guys I just read on the wiki: 2005-07-19 - long awaited extension lights (hint priority) and call pickup on various phones work with newly released asterisk patch digium bugtracker - feel free to test and report findings to the bugtracker to have this commited to cvs. How does this work? And will it work only on certain phones or can it work with the gxp2000? ___ 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 -- Tom ___ 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] new to Asterisk, is it possible to call two external lines and connect them using two channels
I'm afraid that you're going to have to incur the cost of the call if you are calling both parties then bridging the call. Otherwise, there would be no-one to bill. Besides, your box is going to be the one doing the bridging. -- Tom On 7/20/05, Robert Bachan [EMAIL PROTECTED] wrote: Hi All, I am just looking at using Asterisk now and the first thing I need to do is via pass two external numbers to asterisk and call out connecting the calls togther. These will be through our physical PBX connected to the asterisk server. We are essentially trying to connect two external numbers through asterisk with us incurring the cost of the calls. I've been reading on call out files and the manager API. I can see how to call an external number and bridge it to an extension. Do I have to run an application on that extension to then dial the second external number and then they both connect? Many Thanks in advance guys Rob __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 -- Tom ___ 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] Extension Lights Patch
I made a prior post about this: http://lists.digium.com/pipermail/asterisk-users/2005-July/115285.html -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 7/20/05, Eric Rees [EMAIL PROTECTED] wrote: Could you pass along the information you used to get the Polycom lights to work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Hayden Sent: Wednesday, July 20, 2005 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Lights Patch I've been using the extension lights on my polycoms before that patch, so I'm not sure what it fixed, but I've only seen the lights work on Polycoms and Snoms. Try using the hint priority and see if it works for your gxp2000, be sure to post your results! -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 7/20/05, Anton Krall [EMAIL PROTECTED] wrote: Guys I just read on the wiki: 2005-07-19 - long awaited extension lights (hint priority) and call pickup on various phones work with newly released asterisk patch digium bugtracker - feel free to test and report findings to the bugtracker to have this commited to cvs. How does this work? And will it work only on certain phones or can it work with the gxp2000? ___ 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 -- Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ 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] Asterisk/Hylafax = Receive/Send faxes
I have hylafax attached to a Sipura ATA, and just have Asterisk route calls to the fax DID to that ATA. I wouldn't recommend doing this as it is highly unreliable and only about 50-60% of the faxes actually finish. As far as I know, you can't route faxes to hylafax within the same box, however Asterisk does have some fax functions, with spandsp which works similar to hylafax's faxgetty. -- Tom On 7/18/05, Jian Hong GUAN [EMAIL PROTECTED] wrote: Hi, Can you tell me how to configure Hylafax + Asterisk in order to be able to receive/send faxes. Best regards, Guan ___ 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 -- Tom ___ 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] configuring trunks
Your question isn't very clear, but there are a vareity of trunks available. You could get a trunk that uses the zaptel card, such as a PRI/T1/E1. Or you could get a 'virtual' trunk that uses IP such as IAX or SIP. There are numerous providers of IAX and SIP trunks out there, so look around. Check out: http://www.voip-info.org/tiki-index.php?page=VOIP%20Service%20Providers If you want an analog trunk or T1/E1 you need the zaptel card. -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 7/18/05, cciecert [EMAIL PROTECTED] wrote: hi i am new to asterisk and i want to configure trunk with a voice gateway as i read i must have a zaptel card installed in order to do so. but i want to configure the trunk without any cards installed in the server is there anyworkaround to do this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: configuring trunks
Well, I'm not familiar with the Welltech Hardware, but you'll probably just want to configure your * box to register with your SIP server, provided that your Welltech device acts as a SIP server. -- Tom Hayden Astoria Telecommunications www.astoriatelecom.net On 7/18/05, cciecert [EMAIL PROTECTED] wrote: - Original Message - From: cciecert To: asterisk-users@lists.digium.com Sent: Monday, July 18, 2005 1:29 PM Subject: configuring trunks hi i am new to asterisk and i want to configure trunk with a voice gateway as i read i must have a zaptel card installed in order to do so. but i want to configure the trunk without any cards installed in the server is there anyworkaround to do this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ 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] No channels after starting asterisk
What kind of output do you get with ztcfg -vv ?? -- Tom On 7/13/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jul 13, 2005 at 05:19:08PM +0200, Kib Eki wrote: Hi, i am running * 1.0.9 with a newer version of the TE405P. Modprobe wct4xxp and ztcfg are OK. zap show channels only shows me the following. my zapata.conf: [pstn] Shouldn't that be [channels] ? Why can't i see or use my channels? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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 -- Tom ___ 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] Equipment for small office setup
Well, my confs are really long, so I can't go and copy and paste them.
It uses the exact same technique that is used for the snom phones,
and there is no documentation that I could find. This feature works
great on both IP500 and IP600s.
For my IP500
The sip firmware is 1.5.2.0054
My phone in sip.conf is
[p103]
type=friend
username=p103
secret=**
host=dynamic
context=outbound
nat=no
mailbox=103
disallow=all
allow=ulaw
dtmfmode=rfc8233
subscribecontext=toll-trunks
callerid=Tom 103
The relevant portion of my extensions.conf is (in context toll-trunks):
exten = 100,hint,SIP/p100
exten = 100,1,Macro(line,${P100})
exten = 101,hint,SIP/p101
exten = 101,1,Macro(line,${P101})
exten = 102,hint,SIP/p102
exten = 102,1,Macro(line,${P102})
exten = 103,hint,SIP/p103
.. and so on, until all my extensions are used.
The configuration on the phones is far to long/complex to copy and
paste. I just used the standard configs from the wiki, but played
around a little and set the bw variable to 1 in the directory for
buddy watch. Basically, a directory entry looks like this (in the
macaddr-directory.xml file):
item
lnJones/ln
fnMike/fn
ct104/ct
sd1/sd
rt3/rt
dc/
ad0/ad
ar0/ar
bw1/bw
bb0/bb
/item
I'm too lazy to walk across the office and look at the IP600's
firmware, but it's the newest that i could find. The configuration is
pretty much the same, but your directory can display more.
Oh, also, I'm pretty sure you *HAVE* to use the boot server for the
buddy watch feature to work correctly. I could never get it to display
the little 'person' icons using the web or phone interface, but using
the directory file solved that problem. Now, when someone is on the
line, the red light lights up on the IP600 and on the IP500, the icon
is replaced.
If you can get this to work too, you should make an entry in the wiki.
Cheers,
--
Tom
On 7/5/05, Adam Goryachev [EMAIL PROTECTED] wrote:
On Wed, 2005-06-29 at 16:04 -0400, Tom Hayden wrote:
For a small office setting, i've found the Polycom IP600 works great.
Our receptionist usually transfers 90% of her calls to about 5
employees, and the status (buddy watch) feature works great with *
(using hint). She can see who is active at any given time.
Obviously, it's only limited to 5 max status lines, although the Buddy
List feature can show more.
Do you have details on how this works?? I've only ever seen people
reporting success with snom phones?
What sip firmware on the phones?
What in the asterisk sip.cfg?
what in the asterisk extensions.conf?
what in the phones phone.cfg
what in the phones sip.cfg ?
I would love to be able to support/use this.
Thanks for any info you can provide.
Regards,
Adam
--
--
Adam Goryachev
Website Managers
Ph: +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au
--
Tom
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[Asterisk-Users] Fwd: Hangupcode == 44
Just to follow up on my previous issue, Hangupcode 44 indicated that my telco hadn't actually turned ON the circuit (hence no channels). I had multiple techs tell me that the circuit was on and ready-to-go and failed to mention that I needed to call their national activation center. A quick call to the center solved this problem. -- Tom -- Forwarded message -- From: Tom Hayden [EMAIL PROTECTED] Date: Jun 6, 2005 1:16 PM Subject: Hangupcode == 44 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sorry to be bombing the list today, you guys have been more than helpful and I'm having nightmares googling for things :) Anyway, I have a PRI through an Adit 600 and it seems to be setup properly now but when I make a call out I get resultcode 44 (AST_CAUSE_REQUESTED_CHAN_UNAVAIL). Is this a problem with my equipment or a problem with the telco's switch or the adit 600? Additionally, when I try to call a DID inward, I get an All Circuits are Busy now message. Any ideas? -- Tom -- Tom ___ 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] PRI got event: HDLC Abort (6) on Primary D-channel of span 1
What does zttool say? Do you have any IRQ issues or anything? -- Tom On 6/29/05, Michael Blood [EMAIL PROTECTED] wrote: I receive this error on the asterisk console and it is pretty much ALWAYS coming up. Sometimes there will be a break where it does not display. We had our PRI provider test the lines and they claim that there is no signalling problem. It doesn't matter if there are no calls or if there are 10 calls in progress the error is still displayed. I also get an annoying popping or clicking sound but that doesn't always correspond with this error coming up so it is likely a separate issue. I have loaded all modules by hand like below as someone suggested in a search for HDLC errors on the list. insmod zaptel insmod wct1xxp Unfortunately it did not help Has anyone run into this in the past? Michael ;zapata.conf switchtype=national context=incoming_eli_pri_1 signalling=pri_cpe group=1 channel = 1-11 bchan=1-11 dchan=24 ;zaptel.conf span=1,1,0,esf,b8zs bchan=1-11 dchan=24 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394
Re: [Asterisk-Users] Equipment for small office setup
For a small office setting, i've found the Polycom IP600 works great. Our receptionist usually transfers 90% of her calls to about 5 employees, and the status (buddy watch) feature works great with * (using hint). She can see who is active at any given time. Obviously, it's only limited to 5 max status lines, although the Buddy List feature can show more. -- Tom On 6/29/05, Wilson Pickett [EMAIL PROTECTED] wrote: 1 Master phone for a receptionist. Is there an easy way at the moment for one of these bigger phones (cisco or whatever) to view the status of the various lines etc? Some phone with an expansion board maybe? Steve, Flash Operators Panel is a very good tool for a receptionist if they have a PC screen at their desk. I think it would be easier to see who's doing what on the phones with that than any SIP hardphone I've seen. ___ 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 -- Tom ___ 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] PRI Lines not being answered (No User Responding)
Hello! Continuing my PRI saga - I have a PRI setup and appears to be answering calls OK, but my carrier is cutting all the calls after 15 seconds. For example, when I call from my cell phone, it goes straight to a busy signal - however the CLI shows the call coming in and being answered. Additionally, when I call from another ground line, it will ring once or twice, again show as answered, but then give a busy tone. I have a TE110P. I called my carrier and they told me that it gave their trace a No User Responding - however I show my box is answering! Any ideas? I verified that my D-chan should be 24 and the switch is of ni1. Is there some other method or signalling I ought to be using to indicate to the carrier's switch that I have answered? Should I have the carrier switch something for me? Thanks in advance guys - i'm going to put my entire experiences on this one in the wiki :) ** zaptel.conf defaultzone=us span=1,1,0,esf,b8zs bchan=1-12 dchan=24 loadzone=us ** zapata.conf ( cut for brevity) signalling = pri_cpe switchtype = ni1 context = DIDincoming group = 2 channel = 1-12 ** extensions.conf (cut for brevity) [DIDincoming] exten = 0980,1,Answer exten = 0980,2,Dial(SIP/p100,30) exten = 0980,3,Voicemail(100) Here's my CLI output on a cut call (i added the ) -- Accepting call from '734432' to '0980' on channel 0/1, span 1 -- Executing Answer(Zap/1-1, ) in new stack == Spawn extension (DIDincoming, 0980, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Tom ___ 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] PRI Lines not being answered (No User Responding)
Ah. Problem Solved. I need to upgrade my zaptel, libpri, and asterisk from 1.0.7 to the stable cvs release and it solved my problems. Thanks! -- Tom On 6/7/05, Francois Lambert [EMAIL PROTECTED] wrote: Hi Tom, Have you tried replacing your Answer command with Ringing, some PRI requires the Ringing signal before? It worked for us Francois Lambert COO Atelka/Aheeva Inc. Tel.: 514-448-4905 #2200 Cel.: 514-570-4797 ___ 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 -- Tom ___ 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] IRQ Problems
Hello I just installed a TE110P in a Dell Poweredge 750 (rackmount), which is connected via crossover T1 cable to a adit 600. Anyway, I've encountered an array of errors, which I believe I have narrowed down to the 22 IRQ misses I encounter on zttool. I've noticed that the te110p and the usb device share an IRQ. Could that be the source of my woes? Would disabling the usb device in the bios solve my problems or could it be my configuration? -- Errors generated by asterisk chan_zap.c:7970 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. ---zaptel.conf defaultzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us -- zapata.conf (relevant portions) signalling = pri_cpe switchtype = national group = 2 context = fxoout cidsignalling = dtmf cidstart = polarity channel = 1-23 -- proc/interrupts CPU0 0:6334299IO-APIC-edge timer 1: 14390IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 3IO-APIC-edge rtc 14: 2IO-APIC-edge ide0 15: 505440IO-APIC-edge ide1 21: 150019 IO-APIC-level eth1 22: 63326503 IO-APIC-level t1xxp -- lspci -bv | grep IRQ Flags: bus master, medium devsel, latency 0, IRQ 11 Flags: bus master, medium devsel, latency 0, IRQ 10 (USB) Flags: bus master, fast devsel, latency 0, IRQ 255 Flags: bus master, medium devsel, latency 0, IRQ 7 Flags: medium devsel, IRQ 5 Flags: bus master, 66Mhz, medium devsel, latency 0, IRQ 3 Flags: bus master, 66Mhz, medium devsel, latency 32, IRQ 11 Flags: bus master, medium devsel, latency 32, IRQ 10 (**TE110P**) Flags: bus master, VGA palette snoop, stepping, medium devsel, latency 32, IRQ 255 Thanks in Advance, -- Tom ___ 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] IRQ Problems
I Changed it to pri_net and get the following error now: Jun 6 08:52:17 WARNING[3438]: chan_zap.c:7970 pri_dchannel: PRI Error: We think we're the network, but they think they're the network, too. Any ideas? -- Tom On 6/6/05, Gustavo Alvarez [EMAIL PROTECTED] wrote: The problem is that both adit600 and Asterisk box are configured as clock slaves (pri_cpe). Change pri_cpe to pri_net in your zapata.conf, and your asterisk box would be the clock source. It should work. Good Luck. Argentilinux. Tom Hayden escribió: Hello I just installed a TE110P in a Dell Poweredge 750 (rackmount), which is connected via crossover T1 cable to a adit 600. Anyway, I've encountered an array of errors, which I believe I have narrowed down to the 22 IRQ misses I encounter on zttool. I've noticed that the te110p and the usb device share an IRQ. Could that be the source of my woes? Would disabling the usb device in the bios solve my problems or could it be my configuration? -- Errors generated by asterisk chan_zap.c:7970 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. ---zaptel.conf defaultzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us -- zapata.conf (relevant portions) signalling = pri_cpe switchtype = national group = 2 context = fxoout cidsignalling = dtmf cidstart = polarity channel = 1-23 -- proc/interrupts CPU0 0:6334299IO-APIC-edge timer 1: 14390IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 3IO-APIC-edge rtc 14: 2IO-APIC-edge ide0 15: 505440IO-APIC-edge ide1 21: 150019 IO-APIC-level eth1 22: 63326503 IO-APIC-level t1xxp -- lspci -bv | grep IRQ Flags: bus master, medium devsel, latency 0, IRQ 11 Flags: bus master, medium devsel, latency 0, IRQ 10 (USB) Flags: bus master, fast devsel, latency 0, IRQ 255 Flags: bus master, medium devsel, latency 0, IRQ 7 Flags: medium devsel, IRQ 5 Flags: bus master, 66Mhz, medium devsel, latency 0, IRQ 3 Flags: bus master, 66Mhz, medium devsel, latency 32, IRQ 11 Flags: bus master, medium devsel, latency 32, IRQ 10 (**TE110P**) Flags: bus master, VGA palette snoop, stepping, medium devsel, latency 32, IRQ 255 Thanks in Advance, -- Tom ___ 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] IRQ Problems
Is there some kind of guide available or information about connecting to the adit or using it's CLI console? All I have is a null-modem cable. Will that work? Thanks -- Tom On 6/6/05, Zoa [EMAIL PROTECTED] wrote: Hmm, looks like i sent that email a little too fast. ( i scanned through the email but didn't read it all for some reason... sorry) You seem to have two problems, the error is not related to a shared irq but to a setting on either the adit 600 or in the zapata.conf From your last email it looks like you need to change something on the adit 600. (i dont have an adit 600 myself so i dont know what exactly). Cheers, Zoa. Zoa wrote: The sharing of an IRQ is indeed a problem Read this tutorial to know exactly what to do: http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html Greetings, zoa, Tom Hayden wrote: Hello I just installed a TE110P in a Dell Poweredge 750 (rackmount), which is connected via crossover T1 cable to a adit 600. Anyway, I've encountered an array of errors, which I believe I have narrowed down to the 22 IRQ misses I encounter on zttool. I've noticed that the te110p and the usb device share an IRQ. Could that be the source of my woes? Would disabling the usb device in the bios solve my problems or could it be my configuration? -- Errors generated by asterisk chan_zap.c:7970 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. ---zaptel.conf defaultzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us -- zapata.conf (relevant portions) signalling = pri_cpe switchtype = national group = 2 context = fxoout cidsignalling = dtmf cidstart = polarity channel = 1-23 -- proc/interrupts CPU0 0:6334299IO-APIC-edge timer 1: 14390IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 3IO-APIC-edge rtc 14: 2IO-APIC-edge ide0 15: 505440IO-APIC-edge ide1 21: 150019 IO-APIC-level eth1 22: 63326503 IO-APIC-level t1xxp -- lspci -bv | grep IRQ Flags: bus master, medium devsel, latency 0, IRQ 11 Flags: bus master, medium devsel, latency 0, IRQ 10 (USB) Flags: bus master, fast devsel, latency 0, IRQ 255 Flags: bus master, medium devsel, latency 0, IRQ 7 Flags: medium devsel, IRQ 5 Flags: bus master, 66Mhz, medium devsel, latency 0, IRQ 3 Flags: bus master, 66Mhz, medium devsel, latency 32, IRQ 11 Flags: bus master, medium devsel, latency 32, IRQ 10 (**TE110P**) Flags: bus master, VGA palette snoop, stepping, medium devsel, latency 32, IRQ 255 Thanks in Advance, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ 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] Hangupcode == 44
Sorry to be bombing the list today, you guys have been more than helpful and I'm having nightmares googling for things :) Anyway, I have a PRI through an Adit 600 and it seems to be setup properly now but when I make a call out I get resultcode 44 (AST_CAUSE_REQUESTED_CHAN_UNAVAIL). Is this a problem with my equipment or a problem with the telco's switch or the adit 600? Additionally, when I try to call a DID inward, I get an All Circuits are Busy now message. Any ideas? -- Tom ___ 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] IRQ Problems
Well, it seems to be working better now, but everything gets rejected with cause code 44. I probably shouldn't, but I started a new thread about that. I can't call in or out - all cause code 44. I suspect it's coming from the adit - Calling my telco/adit installer. -- Tom On 6/6/05, Rich Adamson [EMAIL PROTECTED] wrote: Not strange at all. One of the two devices is in loopback mode, and my first guess would be the 600. Fix that first, then muck with the parameters in zapata.conf. Thats really strange, one side has to be pri_net and the other pri_cpe always, no matter what devices you use (cisco, asterisk, nortel, ericsson, etc.) If you set asterisk as pri_cpe, and it says the other side is also configured as pri_cpe, change it to pri_net. If it also says that the other side is configured for pri_net, then there must be something wrong with your adit600. Try checking the configuration of the adit600. Argentilinux Tom Hayden escribió: I Changed it to pri_net and get the following error now: Jun 6 08:52:17 WARNING[3438]: chan_zap.c:7970 pri_dchannel: PRI Error: We think we're the network, but they think they're the network, too. Any ideas? -- Tom On 6/6/05, Gustavo Alvarez [EMAIL PROTECTED] wrote: The problem is that both adit600 and Asterisk box are configured as clock slaves (pri_cpe). Change pri_cpe to pri_net in your zapata.conf, and your asterisk box would be the clock source. It should work. Good Luck. Argentilinux. Tom Hayden escribió: Hello I just installed a TE110P in a Dell Poweredge 750 (rackmount), which is connected via crossover T1 cable to a adit 600. Anyway, I've encountered an array of errors, which I believe I have narrowed down to the 22 IRQ misses I encounter on zttool. I've noticed that the te110p and the usb device share an IRQ. Could that be the source of my woes? Would disabling the usb device in the bios solve my problems or could it be my configuration? -- Errors generated by asterisk chan_zap.c:7970 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. ---zaptel.conf defaultzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us -- zapata.conf (relevant portions) signalling = pri_cpe switchtype = national group = 2 context = fxoout cidsignalling = dtmf cidstart = polarity channel = 1-23 -- proc/interrupts CPU0 0:6334299IO-APIC-edge timer 1: 14390IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 3IO-APIC-edge rtc 14: 2IO-APIC-edge ide0 15: 505440IO-APIC-edge ide1 21: 150019 IO-APIC-level eth1 22: 63326503 IO-APIC-level t1xxp -- lspci -bv | grep IRQ Flags: bus master, medium devsel, latency 0, IRQ 11 Flags: bus master, medium devsel, latency 0, IRQ 10 (USB) Flags: bus master, fast devsel, latency 0, IRQ 255 Flags: bus master, medium devsel, latency 0, IRQ 7 Flags: medium devsel, IRQ 5 Flags: bus master, 66Mhz, medium devsel, latency 0, IRQ 3 Flags: bus master, 66Mhz, medium devsel, latency 32, IRQ 11 Flags: bus master, medium devsel, latency 32, IRQ 10 (**TE110P**) Flags: bus master, VGA palette snoop, stepping, medium devsel, latency 32, IRQ 255 Thanks in Advance, ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ 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] VoiPSupply Dot Com
VoIPSupply has given me some of the best customer service I've ever received ANYWHERE. period. I called their 800 number and spoke with Dave (David?) and he advised me on purchases and hooked me up with everything I wanted/needed! Thanks VoIPSupply! -- Tom Hayden On 5/25/05, Karl J. Vesterling [EMAIL PROTECTED] wrote: VoipSupply tech support is awesome... I'll second that observation. At 06:51 PM 5/25/2005, you wrote: I have followed this thread but didn't say anything until I read this one, since I think that the problems he had with VoipSupply.com were the writers fault and not voipsupply.coms. So here I start, I have purchased from voipsupply.com over 120 phones (Polycom, Cisco, Sipura, Uniden) the last few weeks. I have never had a technical or other sales problem with them, not even shipping, of course when I order large quantities I don't expect next day delivery, it's more like 10 days. But whenever I ordered smaller quantities they were shipped the day they were ordered. I even tried by ordering direct from their site without calling the sale rep I usually use, so that I don't get special attention, and viola it was shipped that same day. As for technical support, I can only say that I couldn't ask for something better, they are very good in supporting as much as they have to go (new firmwares, deployment examples, and so on). On 5/25/05, Karl J. Vesterling [EMAIL PROTECTED] wrote: We just bought (a couple weeks ago) Qty 6 Cisco Phones from them. There was no licenses with the phones... Go figure... So, evidently they're supposed to be new, and one would expect they come with the license, but alas no license. Really, since when does it come with a license? http://www.voipsupply.com/product_info.php?manufacturers_id=10products_id=356 They don't even say that it comes with the license. Anyhow why do you want the license? It's only needed for call manager. Try dealing with Cisco and they will tell you $300 + without the license. I'm coming from there. We discovered this during our initial purchase of a few Sipura Units and (2) Cisco 7960 Phones. So if you went ahead and purchased 6 more, you obviously knew whats going on. When we purchased the (6) additional Cisco 7960's it would seem that no only was there no license (as expected), as your ignorance expected. But the phone itself was configured with an unknown password and they weren't consistent with the Firmware versions installed. Needless to say upgrading these to the latest SIP Firmware was quite the difficult task. Cisco ships them with SCCP, so why to you expect voipsupply.com to change it? Nowhere is it mentioned that it will be preloaded with sip. http://www.voipsupply.com/product_info.php?manufacturers_id=10products_id=356 Difficult task? really? so is eating and sleeping, if you haven't got the time (I haven't, at least not always). Also worthy of note is that I flew to the location where we were deploying these units (Buffalo NY, which oddly enough is the home town of Voip-Supply) and they had shipped the purchased items to the Bill-To address instead of the Ship-To address. This makes me think that you didn't either: Ask them to ship to another place, or the Ship-To address wasn't registered with your bank, before you check out the web site clearly states, that if your shipping address is not on file with your bank it will be shipped to your bill to address. Add to that the packages were a day late, and since the Bill-To address is residential they didn't arrive until 4:55PM on a Friday. So why didn't you call UPS or Fedex to fight about this, nothing to do with Voipsupply.com, unless you ordered it way early in the day, and they shipped it the day afterwards. Residential address? I don't see how this is voipsupply.coms fault. Looks more like your fault to me. I had arrived, and anticipated their arrival Thursday AM. No ones fault what you thought, anticipated, or ate that morning. In short, I wound up sitting on my hands with my thumb stuck in an extremely uncomfortable place for 16 hours eagerly anticipating the arrival of the purchased items. I really want to know where that thumb was stuck, any chase of you revealing that? Please?? Here's the kicker. Since we purchased two MediaTrix 1204's I was expecting to be able to get in touch with their technical support if I had difficulties. Alas, Mediatrix business hours are 9:00AM - 5:00PM Mon - Fri Eastern. No dice there... Can you say Show Stopper ??? I thought this was a voipsupply dot com thread. So, not only did I sit on my hands for 16 hours, but this caused me TONS of OK, this gives me some clue at where you thumb was, oh my god for 16 hours? that really hurts. Can you still type with 10 fingers
Re: [Asterisk-Users] Polycom takes long time for reboot to access web page
I have the same problem with the IP300, 500, and 600s. I think it's just because the phone takes a while to start the web services back up after it reboots. -- Tom Hayden On 5/20/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote: When I change a setting via the web interface on a polycom 500, it takes minutes to allow access through the web interface again. Any idea why it is so slow? Chris Mason ___ 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 -- Tom ___ 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] Two TDM04 with Poweredge
Has anyone on this list succesfully managed to get two (or more) TDM04 (with four FXO each) working on a Dell PowerEdge server? If so, which model? Was it a hassle? I'm doing a seven-line installation and a callbank seems like overkill, I just don't want to get suck with a PowerEdge that gets into an IRQ mess. Thanks in Advance, Tom Hayden ___ 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] Getting a good deal on a PRI
I find the prices for a PRI/T1s vary widely. In some locations it's still feasable to have 10 POTS lines with a callbank over a PRI. In some cases, your telco will provide the callbank if you buy special bundles or contracts. You can then connect up the callbank into the PBX with TE110P card. Try calling up some of your local CLECs - they'll probably give you a good deal on either. -- Tom On Apr 8, 2005 9:49 AM, Damon Estep [EMAIL PROTECTED] wrote: Call XO www.xo.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of snacktime Sent: Thursday, April 07, 2005 5:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Getting a good deal on a PRI We have 10 incoming POTS lines to our offices, and a nortel norstar pbx. I've been looking at replacing it with * at some point in the future, and the point that looks most cost effective is when we move to PRI. Problem is, I'm not really sure how to go about getting a good deal, or what questions to ask. 90% of calls will be inbound. I called up Qwest and they quoted me $800 month. I haven't called up any CLEC's yet to see what they can do. Any suggestions? We are in Seattle, Washington. Chris ___ 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 -- Tom -- Tom ___ 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
