Re: [asterisk-users] [OFF TOPIC] monit
ricky gutierrez wrote: someone on the list who is running successfully?, I am using asterisk 11.15 With CentOS 6.5 x64 I use monit, but I only watch the pid check process asterisk with pidfile /var/run/asterisk/asterisk.pid start program = /usr/sbin/service asterisk start stop program = /usr/sbin/service asterisk stop Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OFF TOPIC] monit
2014-12-29 4:51 GMT-06:00 Doug Lytle supp...@drdos.info: I use monit, but I only watch the pid check process asterisk with pidfile /var/run/asterisk/asterisk.pid start program = /usr/sbin/service asterisk start stop program = /usr/sbin/service asterisk stop Doug work fine my friend , thnk -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OFF TOPIC] monit
Hi list , I'm trying to run monit with asterisk, starting as simple # My PBX Asterisk check process asterisk with pidfile /var/run/asterisk/asterisk.pid start program = /etc/init.d/asterisk start with timeout 60 seconds stop program = /etc/init.d/asterisk stop with timeout 60 seconds if failed host 127.0.0.1 port 5038 then restart if 5 restarts within 5 cycles then timeout when I log in (monit interface) I see the status of asterisk is NOT MONITORED port 5038 is ready netstat -an | grep 5038 tcp0 0 127.0.0.1:5038 0.0.0.0:* LISTEN someone on the list who is running successfully?, I am using asterisk 11.15 With CentOS 6.5 x64 regards list. -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off Topic: Aastra BLF limit...
Hi Carlos, It's simply not possible due to a firmware limitation when general SIP and not Aastra proprietary mode (not enougth memory capacity). Don't lack your time by searching a non exisiting solution. Best Regards, Francois -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]de la part de Carlos Chavez Envoyé : lundi 28 décembre 2009 21:00 À : Asterisk Objet : [asterisk-users] Off Topic: Aastra BLF limit... Hi. Does anyone have a patch or workaround for the 50 BLF limit of Aastra phones? I have a couple 57i with the 560M console and only the first 50 BLF lines get registered. I am using the latest firmware from Aastra but I read that this limit was imposed because of a memory leak. Obviously my customer is complaining about these last 10 lines not showing their status. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off Topic: Aastra BLF limit...
Hi. Does anyone have a patch or workaround for the 50 BLF limit of Aastra phones? I have a couple 57i with the 560M console and only the first 50 BLF lines get registered. I am using the latest firmware from Aastra but I read that this limit was imposed because of a memory leak. Obviously my customer is complaining about these last 10 lines not showing their status. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off Topic
Please forgive this off-topic post... I've been on this list since 2005 (over 45k messages in my archive) and this is obviously really not something I normally do. If you have a minute and are feeling generous, please visit http://bailout.chipin.com/ and consider helping me out. Sorry if I've offended or wasted your time, but believe me that you don't feel as bad as I do. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OFF TOPIC] wich virtualization solution to use?
hi there are a lot of virtualization solution out there and every one is the best and has some pro and some cons... wich one do you recomend? the idea to isolate diferents servers asterisk apache ... it is a good idea? sorry for the off topic but here is a place where are a lot of linux gurus Thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OFF TOPIC] wich virtualization solution to use?
It's a good idea if it makes sense to you organisationally. There is no definitive answer on that; it is a methodological question. David fire wrote: hi there are a lot of virtualization solution out there and every one is the best and has some pro and some cons... wich one do you recomend? the idea to isolate diferents servers asterisk apache ... it is a good idea? sorry for the off topic but here is a place where are a lot of linux gurus Thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OFF TOPIC] wich virtualization solution to use?
On Sat, Apr 11, 2009 at 12:04 PM, David fire ddf...@gmail.com wrote: hi there are a lot of virtualization solution out there and every one is the best and has some pro and some cons... wich one do you recomend? the idea to isolate diferents servers asterisk apache ... it is a good idea? sorry for the off topic but here is a place where are a lot of linux gurus Thanks If all your virtual machines are linux, openvz is probably the easiest and provides the best performance. But all it does is linux. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off-topic: SIP DTMF most supported method
Hi, I know it is a bit off-topic, but I'd like to ask the community what is the current most supported way to deal with DTMF? I'm looking for an all-SIP system and I'm mostly interested in the end devices support of the different methods (DTMF in-band audio, DTMF RTP telephony events packets, SIP INFO, ...) Thanks in advance. Cesc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-topic: SIP DTMF most supported method
It's SIP in rfc (RFC2833) then SIP INFO and then if you can't do anything else inband audio (only G711) Martin On Mon, Apr 6, 2009 at 2:24 AM, Cesc Santa cesc.sa...@gmail.com wrote: Hi, I know it is a bit off-topic, but I'd like to ask the community what is the current most supported way to deal with DTMF? I'm looking for an all-SIP system and I'm mostly interested in the end devices support of the different methods (DTMF in-band audio, DTMF RTP telephony events packets, SIP INFO, ...) Thanks in advance. Cesc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] off topic - voip providers raided by FBI for unpaid telecom bills:
http://tech.slashdot.org/article.pl?sid=09/04/04/2013200 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] off topic - voip providers raided by FBI forunpaid telecom bills:
Lol how about someone raiding ATT exchanges for unpaid fees for low cost call terminations etc . Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of zoach...@securax.org Sent: Saturday, April 04, 2009 6:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] off topic - voip providers raided by FBI forunpaid telecom bills: http://tech.slashdot.org/article.pl?sid=09/04/04/2013200 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] off topic - voip providers raided by FBI forunpaid telecom bills:
Yes , scary shit, we lost our rights on PA, now since no one is watching those guys, they have all the power they need and want, while feeding the rest of us with crap e-drugs like hockey, stupid tv shows like biggest loser, and using marketing power words like terr0r and shit. While the citizens sleep , the wolves eat our livestock. Of course you can still vote, with the use of machines that are biased and uncontrolled . The system itself is broken , so whether you choose left or right , white or black , 1 or 0 , it's a lose-lose situation. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Collins Sent: April-04-09 6:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: a...@iotum.com Subject: Re: [asterisk-users] off topic - voip providers raided by FBI forunpaid telecom bills: Lol how about someone raiding ATT exchanges for unpaid fees for low cost call terminations etc . Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of zoach...@securax.org Sent: Saturday, April 04, 2009 6:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] off topic - voip providers raided by FBI forunpaid telecom bills: http://tech.slashdot.org/article.pl?sid=09/04/04/2013200 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off Topic: Disable Polycom Soundpoint DoNotDisturb Feature
Roi Stork wrote: However, the problem is that there is still no ringing sound so the user can't hear it. Is there a way to make the ringing tone audible? You can remap the DND key to do something else (or nothing). It may still be possible for the user to set DND status via the menus, though. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off Topic: Disable Polycom Soundpoint DoNotDisturb Feature
We have set the do not disturb feature on our polycom phone such that incoming calls will not be rejected and sent to 'Busy' status. The user can still toggle dnd on/off, but incoming calls will still get in, indicated by the blinking light and the screen status. We were able to do that by setting call.rejectBusyOnDnd=0 in sip.cfg. However, the problem is that there is still no ringing sound so the user can't hear it. Is there a way to make the ringing tone audible? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off Topic: 8x FXO Gateway
Hi, I'm seeking an 8 port FXO gateway. Please let me know if anyone can assist -- Regards, Sahil Gupta Corporate Advisor TigerCom Pte. Limited 296 River Valley Road Singapore 238337 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off Topic: 8x FXO Gateway
I'm seeking an 8 port FXO gateway. Please let me know if anyone can assist Use google and such, or at least specify the location (europe,usa,australia,whatever) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off topic...AOCN wanted
For those CLECs out there, if you know of a contract AOCN that you have personal experience with and would recommend, please reply. For those who don't know what an AOCN is, please delete this message. Bruce Komito WPTI Telecom (775) 236-5815 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off topic...AOCN wanted
On Monday 17 December 2007 16:19, Bruce Komito wrote: For those CLECs out there, if you know of a contract AOCN that you have personal experience with and would recommend, please reply. For those who don't know what an AOCN is, please delete this message. I know what an AOCN is, but please use the -biz list in the future for these types of queries. That is what it is there for. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: Avaya
Avaya makes 52% of it's revenue from professional services. In enterprises, you generally have 3 budgets: Captial, expense, professional services Avaya figured out that they could make more money tapping into professional services portion of the budget with charge by the hour union consultants than by selling equipment. Avaya is also the most pervasive vendor in the space when it come to calling dev products GA, so they can get their customers to pay them to beta test. Avaya's newest ploy is to get customers hooked on their systems and after 6 - 12 months of shear hell supporting the products, they kindly offer to outsource your voice infrastructure support using a system called SIG. SIG requires you to place a collector box on your network with an IPSEC VPN nailed up to Avaya corporate. This gives them full unchecked access to your network. Exciting huh? Introducing Avaya into a corporate network is about as smart as introducing syphalis into a high school. Sure, it was all fun and games at first, but eventually it catches up to you. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesse Molina Sent: Saturday, December 01, 2007 1:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Off-Topic: Avaya Salvatore Giudice wrote: They are cheap. You only have to pay for the box and the maintenance percentage. That is indeed the Avaya way. First you buy it, then you rent it. Stop paying their maintenance fees and their dial into your PBX and cripple the OS by removing customer maintenance command permissions. Hell, Avaya won't even give you root on any of their servers. You cant audit the box and you can't poll them unless you pay them money to join their partner program and get their SDK. If you already have Avaya, you should just buy Message Networking or a Mitel voicemail server if you want seamless voicemail with Avaya. However, you should know that using Avaya is probably a bad idea to begin with. Until February 07, the majority Avaya's soft switch products were running on Redhat 9, which was unsupported since 2003. Avaya was only managing a dozen packages and they've always left it up to the customer to know when they need an update, requiring the customer to request a field load. It has to be the worst update model in the industry when it comes to infrastructure monitoring and patching. By using Avaya, you are blindly trusting them to properly maintain a Linux appliance. This is something they are not capable of and you can't even audit them. Avaya is what happens to organizations when they have ignorant telecom infrastructure engineers deciding what products to buy. Avaya focuses sales on those engineers because they k now their products won't pass certification by network, systems, or security engineers. Telecom engineers only look for features and usually get their asses handed to them after they put Avaya VoIP into their infrastructure. Bravo. A well-deserved lambasting of this awful vendor. -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: Avaya
I manage a large Avaya implementation with three systems at different locations. I hate Avaya's manageability, lack of features, and extremely high cost. That's why I'm looking into alternatives to replace the whole thing in a year or two. I would appreciate any other opinions and findings regarding the integration with Avaya and switching from Avaya. Our IP phones are 4600 series as well. Also, I don't think SIP was even supported until CM v3.x, so you're SOL with anything earlier. Jim Houser wrote: This is both a hardware and software licensing issue. Avaya offers a SIP server separate from their main VoIP gateway. The core platform uses H.323. Either SIP or H.323 has a license cost per registered device. We have an Avaya S8300 Communications Manager providing H.323 and have this tied to an Asterisk deployment on a Sun Microsystems server. The connection between the two systems are handled by both T1, (PRI using Qsig), and H.323. The BIG issue we have is we cannot light the message waiting light on the Avaya 46XX phones registered to the Avaya server but using Asterisk voice mail. If anyone can help we would pay to solve this. Our Asterisk is 1.2.xx. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Cabrera Obed Sent: Friday, November 30, 2007 7:30 AM To: Asterisk Users Mailing List Subject: [asterisk-users] Off-Topic: Avaya Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP or H.323 ??? Anybody can't tell me this...so I'm here for thei reason. Thanks a lot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off-Topic: Avaya
Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP or H.323 ??? Anybody can't tell me this...so I'm here for thei reason. Thanks a lot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: Avaya
Salvatore Giudice wrote: They are cheap. You only have to pay for the box and the maintenance percentage. That is indeed the Avaya way. First you buy it, then you rent it. Stop paying their maintenance fees and their dial into your PBX and cripple the OS by removing customer maintenance command permissions. Hell, Avaya won't even give you root on any of their servers. You cant audit the box and you can't poll them unless you pay them money to join their partner program and get their SDK. If you already have Avaya, you should just buy Message Networking or a Mitel voicemail server if you want seamless voicemail with Avaya. However, you should know that using Avaya is probably a bad idea to begin with. Until February 07, the majority Avaya's soft switch products were running on Redhat 9, which was unsupported since 2003. Avaya was only managing a dozen packages and they've always left it up to the customer to know when they need an update, requiring the customer to request a field load. It has to be the worst update model in the industry when it comes to infrastructure monitoring and patching. By using Avaya, you are blindly trusting them to properly maintain a Linux appliance. This is something they are not capable of and you can't even audit them. Avaya is what happens to organizations when they have ignorant telecom infrastructure engineers deciding what products to buy. Avaya focuses sales on those engineers because they k now their products won't pass certification by network, systems, or security engineers. Telecom engineers only look for features and usually get their asses handed to them after they put Avaya VoIP into their infrastructure. Bravo. A well-deserved lambasting of this awful vendor. -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: Avaya
This is both a hardware and software licensing issue. Avaya offers a SIP server separate from their main VoIP gateway. The core platform uses H.323. Either SIP or H.323 has a license cost per registered device. We have an Avaya S8300 Communications Manager providing H.323 and have this tied to an Asterisk deployment on a Sun Microsystems server. The connection between the two systems are handled by both T1, (PRI using Qsig), and H.323. The BIG issue we have is we cannot light the message waiting light on the Avaya 46XX phones registered to the Avaya server but using Asterisk voice mail. If anyone can help we would pay to solve this. Our Asterisk is 1.2.xx. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Cabrera Obed Sent: Friday, November 30, 2007 7:30 AM To: Asterisk Users Mailing List Subject: [asterisk-users] Off-Topic: Avaya Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP or H.323 ??? Anybody can't tell me this...so I'm here for thei reason. Thanks a lot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: Avaya
If you desire SIP in Avaya, you have to add a SES (SIP Enablement Server) to your Avaya setup. They are cheap. You only have to pay for the box and the maintenance percentage. You don't need to buy user ports or any of that garbage as long as you setup your extensions using Optum, which is a free Avaya feature. The SES maintains a registry and a dial plan. SIP phones attached to SES send media directly to medpros and the SES does a protocol conversion between SIP and H.323 to bridge a connection between the SIP phone and the CLAN cards. The voicemail issue you describe with the MWI is because Avaya's systems use qsig trunks to connect to voicemail servers. Asterisk is not connected int hat manner, so of course you won't be able to support Avaya MWI's. However, you can deposit a script on your asterisk that would send the standard notifies to the Avaya phones to manipulate the MWI's directly. However, you will need to statically address the phones and keep track of them because you cannot poll an SES server for their SIP URI's. Hell, Avaya won't even give you root on any of their servers. You cant audit the box and you can't poll them unless you pay them money to join their partner program and get their SDK. If you already have Avaya, you should just buy Message Networking or a Mitel voicemail server if you want seamless voicemail with Avaya. However, you should know that using Avaya is probably a bad idea to begin with. Until February 07, the majority Avaya's soft switch products were running on Redhat 9, which was unsupported since 2003. Avaya was only managing a dozen packages and they've always left it up to the customer to know when they need an update, requiring the customer to request a field load. It has to be the worst update model in the industry when it comes to infrastructure monitoring and patching. By using Avaya, you are blindly trusting them to properly maintain a Linux appliance. This is something they are not capable of and you can't even audit them. Avaya is what happens to organizations when they have ignorant telecom infrastructure engineers deciding what products to buy. Avaya focuses sales on those engineers because they k now their products won't pass certification by network, systems, or security engineers. Telecom engineers only look for features and usually get their asses handed to them after they put Avaya VoIP into their infrastructure. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Houser Sent: Friday, November 30, 2007 9:54 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Off-Topic: Avaya This is both a hardware and software licensing issue. Avaya offers a SIP server separate from their main VoIP gateway. The core platform uses H.323. Either SIP or H.323 has a license cost per registered device. We have an Avaya S8300 Communications Manager providing H.323 and have this tied to an Asterisk deployment on a Sun Microsystems server. The connection between the two systems are handled by both T1, (PRI using Qsig), and H.323. The BIG issue we have is we cannot light the message waiting light on the Avaya 46XX phones registered to the Avaya server but using Asterisk voice mail. If anyone can help we would pay to solve this. Our Asterisk is 1.2.xx. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Cabrera Obed Sent: Friday, November 30, 2007 7:30 AM To: Asterisk Users Mailing List Subject: [asterisk-users] Off-Topic: Avaya Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP or H.323 ??? Anybody can't tell me this...so I'm here for thei reason. Thanks a lot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: add GSM codec to X-Lite
Latest version of X-lite does not have GSM codec . Downgrde your versiona dn you will get gsm codec . I read on their forums that next version will again be including GSM codec . On 03/11/2007, Julio Tejera [EMAIL PROTECTED] wrote: Latest version of X-Lite does not support GSM codecs any more It could be a good idea that you post on the rigth place not here :o) jat - Original Message - From: Alejandro Cabrera Obed [EMAIL PROTECTED] To: asterisk Users Mailing List asterisk-users@lists.digium.com Sent: Friday, November 02, 2007 2:05 PM Subject: Re: [asterisk-users] Off-Topic: add GSM codec to X-Lite SIP wrote: Alejandro Cabrera Obed wrote: Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in allow=gsm line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF manual of X-Lite 3.0 say it has GSM builtin support. Do you know what's the matter with X-Lite and GSM ??? Can I add it ??? Really thanks Alejandro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It lists GSM on my audio codec settings. Perhaps there's something wrong with your install? Try disabling the Zero Touch bandwidth detection. It has, in the past, interfered with my selection of codecs. N. Thanks for your support...I've uninstalled my X-Lite 3.0 softphone and after that I've downloaded the X-Lite 3.0 again from the official web site. But when I go to audio codecs settings, the GSM codec is not present. I disable the zero touch bandwith detection and restart the softphone, but the GSM codec is not present at all. Any idea ??? Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Internal Virus Database is out-of-date. Checked by AVG. Version: 7.5.446 / Virus Database: 269.3.0/758 - Release Date: 4/12/2007 11:52 AM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: add GSM codec to X-Lite
Alejandro Cabrera Obed wrote: Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in allow=gsm line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF manual of X-Lite 3.0 say it has GSM builtin support. Do you know what's the matter with X-Lite and GSM ??? Can I add it ??? Really thanks Alejandro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It lists GSM on my audio codec settings. Perhaps there's something wrong with your install? Try disabling the Zero Touch bandwidth detection. It has, in the past, interfered with my selection of codecs. N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off-Topic: add GSM codec to X-Lite
Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in allow=gsm line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF manual of X-Lite 3.0 say it has GSM builtin support. Do you know what's the matter with X-Lite and GSM ??? Can I add it ??? Really thanks Alejandro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: add GSM codec to X-Lite
SIP wrote: Alejandro Cabrera Obed wrote: Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in allow=gsm line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF manual of X-Lite 3.0 say it has GSM builtin support. Do you know what's the matter with X-Lite and GSM ??? Can I add it ??? Really thanks Alejandro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It lists GSM on my audio codec settings. Perhaps there's something wrong with your install? Try disabling the Zero Touch bandwidth detection. It has, in the past, interfered with my selection of codecs. N. Thanks for your support...I've uninstalled my X-Lite 3.0 softphone and after that I've downloaded the X-Lite 3.0 again from the official web site. But when I go to audio codecs settings, the GSM codec is not present. I disable the zero touch bandwith detection and restart the softphone, but the GSM codec is not present at all. Any idea ??? Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: add GSM codec to X-Lite
Latest version of X-Lite does not support GSM codecs any more It could be a good idea that you post on the rigth place not here :o) jat - Original Message - From: Alejandro Cabrera Obed [EMAIL PROTECTED] To: asterisk Users Mailing List asterisk-users@lists.digium.com Sent: Friday, November 02, 2007 2:05 PM Subject: Re: [asterisk-users] Off-Topic: add GSM codec to X-Lite SIP wrote: Alejandro Cabrera Obed wrote: Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in allow=gsm line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF manual of X-Lite 3.0 say it has GSM builtin support. Do you know what's the matter with X-Lite and GSM ??? Can I add it ??? Really thanks Alejandro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It lists GSM on my audio codec settings. Perhaps there's something wrong with your install? Try disabling the Zero Touch bandwidth detection. It has, in the past, interfered with my selection of codecs. N. Thanks for your support...I've uninstalled my X-Lite 3.0 softphone and after that I've downloaded the X-Lite 3.0 again from the official web site. But when I go to audio codecs settings, the GSM codec is not present. I disable the zero touch bandwith detection and restart the softphone, but the GSM codec is not present at all. Any idea ??? Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Internal Virus Database is out-of-date. Checked by AVG. Version: 7.5.446 / Virus Database: 269.3.0/758 - Release Date: 4/12/2007 11:52 AM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] off-topic: Avaya 46xx, release 032207 ... help
Hi, I am trying to use an Avaya 4602 phone, which I just updated from a very old SIP software to the latest I could find on avaya's site (032207). The upgrade went fine and it gets registered on the Asterisk server. Now, a couple of glitches, though. - The phone's web server is not working ... so I have no easy way to configure it. It used to work with the old release of the software. I get on the firefox browser a connection has been reset error message. - Avaya admin guide keeps mentioning all the commands you can enter via the keyboard on the phone ... but they don't work for me ... (the MUTE + numbers combination). Any ideas? the web browser problem is the most annoying one. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] off-topic: Avaya 46xx, release 032207 ... help
Cesc Santa wrote: Hi, I am trying to use an Avaya 4602 phone, which I just updated from a very old SIP software to the latest I could find on avaya's site (032207). The upgrade went fine and it gets registered on the Asterisk server. Now, a couple of glitches, though. - The phone's web server is not working ... so I have no easy way to configure it. It used to work with the old release of the software. I get on the firefox browser a connection has been reset error message. - Avaya admin guide keeps mentioning all the commands you can enter via the keyboard on the phone ... but they don't work for me ... (the MUTE + numbers combination). Any ideas? the web browser problem is the most annoying one. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users don't know about the web interface, but hold then RESET # resets the phone, HOLD ADDR # allows you to set the ip address etc try those Regards Robb ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off Topic: Open Source USB Softphone
You could use an rebranded (OEM) idefisk - does sip and IAX and uses XML for the config files, not the registry - making it possible to use it on a usb stick. More info : http://www.asteriskguru.com/idefisk/oem/ (But its not open source, nor free). Joachim Mike Lynchfield wrote: sip would be the required one as iax..well.. also openwengo wont work.. to much overhead .. broswrer needed.. ie component + flash + css+js etc.. not viable.. so im also asking anyone have one ? since ihave a supply of around 2000 of the vonage usb stick OEM.. On 3/30/07, *Michael Van Donselaar* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works with the Yealink phone. On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone)
Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call handling, DSP or anything really number crunching, no telephony terminal or other services. The lowest-performance device that plugs into the USB, with its own Linux instance. In OEM quantity, under $50? Under $100? On Sun, 2007-04-01 at 02:51 -0700, [EMAIL PROTECTED] wrote: Date: Sat, 31 Mar 2007 16:02:06 -0500 From: Mike Lynchfield [EMAIL PROTECTED] Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 sip would be the required one as iax..well.. also openwengo wont work.. to much overhead .. broswrer needed.. ie component + flash + css+js etc.. not viable.. so im also asking anyone have one ? since ihave a supply of around 2000 of the vonage usb stick OEM.. On 3/30/07, Michael Van Donselaar [EMAIL PROTECTED] wrote: Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works with the Yealink phone. On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED] wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s ___ --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 -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -- (C) Matthew Rubenstein ___ --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: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone)
Probably not exactly what you're looking for but Astlinux runs on Gumstix. Would be suitable for prototyping. Michael On Sun, 01 Apr 2007 09:08:17 -0400, Matthew Rubenstein wrote: Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call handling, DSP or anything really number crunching, no telephony terminal or other services. The lowest-performance device that plugs into the USB, with its own Linux instance. In OEM quantity, under $50? Under $100? On Sun, 2007-04-01 at 02:51 -0700, [EMAIL PROTECTED] wrote: Date: Sat, 31 Mar 2007 16:02:06 -0500 From: Mike Lynchfield [EMAIL PROTECTED] Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 sip would be the required one as iax..well.. also openwengo wont work.. to much overhead .. broswrer needed.. ie component + flash + css+js etc.. not viable.. so im also asking anyone have one ? since ihave a supply of around 2000 of the vonage usb stick OEM.. On 3/30/07, Michael Van Donselaar [EMAIL PROTECTED] wrote: Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works with the Yealink phone. On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED] wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s ___ --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 -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone)
Try installing Monte Vista http://www.mvista.com/ on the usb stick. It will be a lot cleaner than taking a standard server distribution of linux and stripping out all the unwanted kernel modules. Monte Vista is an embedded linux that should be able to boot your server off a 128mb usb stick with Asterisk installed. You should probably strip asterisk down to the bare essentials for your project as well. You should be aware that flash memory is generally not the best medium to store data when you have a high number of read/writes. Flash memory will fail much more quickly under these conditions. You might want to consider using a usb microdrive instead of a flash stick. Pick a microdrive that generates as little heat as possible. BTW, what exactly is the motivation for running linux off of a usb stick? If you would like cdr's, you could likely do so with ngrep and a perl script. Good luck, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Sunday, April 01, 2007 9:08 AM To: Asterisk-Users Subject: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone) Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call handling, DSP or anything really number crunching, no telephony terminal or other services. The lowest-performance device that plugs into the USB, with its own Linux instance. In OEM quantity, under $50? Under $100? On Sun, 2007-04-01 at 02:51 -0700, [EMAIL PROTECTED] wrote: Date: Sat, 31 Mar 2007 16:02:06 -0500 From: Mike Lynchfield [EMAIL PROTECTED] Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 sip would be the required one as iax..well.. also openwengo wont work.. to much overhead .. broswrer needed.. ie component + flash + css+js etc.. not viable.. so im also asking anyone have one ? since ihave a supply of around 2000 of the vonage usb stick OEM.. On 3/30/07, Michael Van Donselaar [EMAIL PROTECTED] wrote: Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works with the Yealink phone. On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED] wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s ___ --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 -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users]Off Topic: Open Source USB Softphone)
Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call handling, DSP or anything really number crunching, no telephony terminal or other services. The lowest-performance device that plugs into the USB, with its own Linux instance. In OEM quantity, under $50? Under $100? When you say devices do you mean an off the self device or a module you can use to build a custom device ? In the first case there are a lot of fisrt generation routers coming into the market at very low prices for example http://www.wirelesslan.gr/product_info.php?cPath=127products_id=866 http://www.wirelesslan.gr/product_info.php?products_id=670 If you are looking for a SoC type device there are several although, the 100$ range looks more realistic There are several devices that could be used. DimmPC comes in my mind - http://www.amctechcorp.com/dimmpci/index.html Digi's Connectcore http://www.digi.com/products/embeddedsolutions/connectcore9u.jsp Check Linux devices for a larger list http://www.linuxdevices.com/articles/AT8498487406.html Hope it helps Stelios Stelios S. Koroneos Digital OPSiS - Embedded Inteligence http://www.digital-opsis.com ___ --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] Off Topic: Open Source USB Softphone
sip would be the required one as iax..well.. also openwengo wont work.. to much overhead .. broswrer needed.. ie component + flash + css+js etc.. not viable.. so im also asking anyone have one ? since ihave a supply of around 2000 of the vonage usb stick OEM.. On 3/30/07, Michael Van Donselaar [EMAIL PROTECTED] wrote: Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works with the Yealink phone. On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED] wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s ___ --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 -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --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] Off Topic: Open Source USB Softphone
On Fri, 30 Mar 2007, laurent schweizer wrote: openwengo.org Looks good - however question/answer 1 in their FAQ: http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsNg#a1 Can WengoPhone 2.0 be used with any SIP provider? Not right now. However, it is the item with the highest priority on our todo list, apart from having a 2.0 release. So expect to see this feature implement right after the first NG release. But it is open source, so should be easy to hack something in :) Gordon Laurent 2007/3/29, Gordon Henderson [EMAIL PROTECTED]: On Thu, 29 Mar 2007, Luis Claudio Santos wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? Maybe not quite what you want, but I used a Yealink USB phone with Linux and there was a driver for it that would let you read the keyboard, and program the display - it wasn't a bit-mapped display though. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off Topic: Open Source USB Softphone
Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works with the Yealink phone. On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED] wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off Topic: Open Source USB Softphone
I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s -- Abraços Luis Claudio Mobile + 55 21 9215 2888 Mobile +55 15 9141 8402 Office +55 15 2102 5859 ___ --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] Off Topic: Open Source USB Softphone
On Thu, 29 Mar 2007, Luis Claudio Santos wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? Maybe not quite what you want, but I used a Yealink USB phone with Linux and there was a driver for it that would let you read the keyboard, and program the display - it wasn't a bit-mapped display though. Gordon ___ --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] Off Topic: Open Source USB Softphone
openwengo.org Laurent 2007/3/29, Gordon Henderson [EMAIL PROTECTED]: On Thu, 29 Mar 2007, Luis Claudio Santos wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? Maybe not quite what you want, but I used a Yealink USB phone with Linux and there was a driver for it that would let you read the keyboard, and program the display - it wasn't a bit-mapped display though. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (Off-topic) Voip number tracert
Hi, I was having a conversation with some friends and one of them brought up the question about find out information about one sip number. Is there any way to find out that a number, for instance 1(646)- is actually [EMAIL PROTECTED] In other words, I will know that the number belongs to the provider abc voip. Please, Does anyboy know if that existis or am I tripin that much? Thank you _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ --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] (Off-topic) Voip number tracert
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Daniel Cyt wrote: Hi, I was having a conversation with some friends and one of them brought up the question about find out information about one sip number. Is there any way to find out that a number, for instance 1(646)- is actually [EMAIL PROTECTED] In other words, I will know that the number belongs to the provider abc voip. Please, Does anyboy know if that existis or am I tripin that much? Have a look for E164 and ENUM. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFCcsCS6d5vy0jeVcRAhJOAJ9LeQi4vMLezZSbC4UZLLwyD+CluQCeOtIb X72lxUdrHcXIvRIk8cGvK5M= =uQ4I -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off Topic: Hardware Required
Hi, Apologies for the off-topic post, is there anybody in NYC with a bunch of video cards lying around that I might be able to get picked up this evening or early tomorrow ? Regards, Sahil Gupta VoiceValley ___ --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] Off Topic: Hardware Required
Yes and with some free pizza and coke as well. Located at the Bowery subway station, under the stairs. On 8/31/06, Sahil Gupta [EMAIL PROTECTED] wrote: Hi, Apologies for the off-topic post, is there anybody in NYC with a bunch of video cards lying around that I might be able to get picked up this evening or early tomorrow ? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OFF TOPIC: Core router upgrade for a voip colocation center
Hello, hope this isn't too far offtopic here but being a troller for a long time here I've realized there is a great knowledge base so I wanted to at least see if i could get some tips. I help run a small colocation company in California and I am in the middle of recommending a new 'core router' platform for our network. We offer mainly colo and dedicated servers, and several of our clients use our space for VOIP services so quality even under high peak usage is a must. We are not huge, but as we have had near 200% growth in the past 12 months and need to expand our network asap to keep up. Simply put, I'd love to hear feedback and/or suggestions from any of you guys who have gone through this already. Our network map is real simple: [Carrier 7609] -- 100 mbit -- Our cisco 7206 -- 100 mbit -- racks [the racks on our end are a series of switches, mainly 2948gl3's] We push about 60 mbit to/from our (1) carrier at peak right now, and the router keeps up fine [its a cisco 7206 npe 150 btw, very low end on the 7206 line], and at peak we have under 50,000 packets per second, and our 7206 File router-choices.txt not changed so no update needed. [EMAIL PROTECTED] Hardware]$ cat router-choices.txt #1 - http://list.linux-vserver.org/archive/vserver/ #2 - webhostingtalk.com: jharington68/adam123 http://www.webhostingtalk.com/forumdisplay.php?f=44 #3 - asterisk mail list http://lists.digium.com/mailman/listinfo/asterisk-users #4 - cisco mail list? HOTMAIL [EMAIL PROTECTED] pw/adam123 Hello, hope this isn't too far offtopic here but being a troller for a long time here I've realized there is a great knowledge base so I wanted to at least see if i could get some tips. I help run a small colocation company in California and I am in the middle of recommending a new 'core router' platform for our network. We offer mainly colo and dedicated servers, and several of our clients use our space for VOIP services so quality even under high peak usage is a must. We are not huge, but as we have had near 200% growth in the past 12 months and need to expand our network asap to keep up. Simply put, I'd love to hear feedback and/or suggestions from any of you guys who have gone through this already. Our network map is real simple: [Carrier 7609] -- 100 mbit -- Our cisco 7206 -- 100 mbit -- racks [the racks on our end are a series of switches, mainly 2948gl3's] We push about 60 mbit to/from our (1) carrier at peak right now, and the router keeps up fine [its a cisco 7206 npe 150 btw, very low end on the 7206 line], and at peak we have under 50,000 packets per second, and our 7206 has little/no features enabled [just static routes and passing all traffic between 2 Ethernet 100 mbit interfaces]. To date we have had 2 problems, both were DOS attacks launched FROM one of our customer's servers flooding a full 100 mbit wire with more packets per second than the router could handle (the 2948gl3's spiked to about 50% cpu load during the attack but the 7200 literally just died for 3 minutes as the interface(s) all rebooted]. So our main goal to grow is something that can handle a lot more in this arena against a DOS, and handle our future growth. In then next 12 months we plan to add a 2nd carrier, at t3, 100mbit, or possibly oc3 speed, and possibly upgrade our main carrier to a GigE connection. Probably maxing combined in the 300 mbit range, more likely closer to half that in 12 months. Problems/Requirements - Budget is in the $5k to $20k range ($20k if its going to outlast me even past my 12 month projections) - must not 'collapse' under simple packet flow DOS attack - must handle BGP4 from 2 carriers with full route tables - We plan to buy used, prices below are based on USED, 30 day warranty ebay postings = Choices/Options that we have looked at: Option #1: Cisco VXR 7206 [$4k to $12k] Option #2: Cisco 12008 [$7k to $14k] Option #3: Cisco 6509 [$10k to $15k] Here are the 3 main options, broken down a bit more in depth. [I have not ruled out juniper all together, but not enough experience with them and lots of experience with cisco, makes cisco our better option i think, especially since its easier to find used cisco gear than it is to find used juniper gear at a decent price]. [option #1 - Cisco 7206 VXR] Estimated: $4,000 [$6,000 with 400 mhz, $12,000 with the 1 ghz cpu upgrade] 1 Cisco 7206 VXR NPE 300 mhz w/max ram 2 AC Power 2 Fast Ethernet Adapters (1 included on the NPE) + lots of experience on this unit + lots of spare cards (most compatible) + can keep old 7200 as a hot standby, minimizing long term downtime - END OF LIFE/sale/support on most of the 7200 product line over 5 years ago! The VXR model is darn close to end of life i suspect - minimal horse power here for the money, prone to death by packet attack [option #2 - Cisco GSR (12008)] Estimated: $7,000 to $14,000 [varies if I start with GigE or just
Re: [Asterisk-Users] OFF TOPIC: Core router upgrade for a voip colocation center
Josh,You are in a position that many of our customers have found themselves in. For ISPs/Colo operations starting out Cisco 720x, Foundry BigIron, RiverStone, and Extreme switches all present an aggressive price point for the performance. However once you pass this point Foundry, RiverStone and Extreme all start to become exponentially expensive due to the lack of parts on the resale market. I would advise you against the 7206/12008 upgrade for a couple of different reasons. 1) Like you said they arenearing their EOL date,2) Processor performance is limited based on today's standards (unless you want to shell out for the NPE-G1 or the PRP), 3) They have a limited number of fixed interface ports, and additional line cards are expensive. With the exception of sites running Sonet the 6500 platform is the only way to go. We have sites running the SUP 720 3BXL cards with over 20 full BGP sessions pushing Gigs worth of traffic through them. When you look at processor/memory utilization you wouldn't even know the switch was being used. For your configuration I would recommend a 6503/6504 with a Sup 32 (WS-SUP32-GE-3B)supervisor module... 1) The Sup 32 comes with enough processor/memory to handle BGP in real world situations (256MB standard, upgradable to 1GB),2) It has 8 SFP ports on the supervisor module which is enough for most mid tier applications, 3) 32GB shared bus,4) 15 million packets per second, and my favorite reason, 5) it runs IOS My company is based in Los Angeles, give me a call and I will be more than happy to go over all of this with you. Best, MaxMax ClarkCreative Thought, Inc.(866)231-7371 x 3874(213)784-3874 Direct(866)369-0953 24/7 SupportIT should facilitate business, we can help.On 1/23/06, josh harrington [EMAIL PROTECTED] wrote: Hello, hope this isn't too far offtopic here but being a troller for a long time here I've realized there is a great knowledge base so I wanted to at least see if i could get some tips.I help run a small colocation company in California and I am in the middle of recommending a new 'core router' platform for our network.We offer mainly colo and dedicated servers, and several of our clients use our space for VOIP services so quality even under high peak usage is a must.We are not huge, but as we have had near 200% growth in the past 12 months and need to expand our network asap to keep up. Simply put, I'd love to hear feedback and/or suggestions from any of you guys who have gone through this already. Our network map is real simple: [Carrier 7609] -- 100 mbit -- Our cisco 7206 -- 100 mbit -- racks [the racks on our end are a series of switches, mainly 2948gl3's] We push about 60 mbit to/from our (1) carrier at peak right now, and the router keeps up fine [its a cisco 7206 npe 150 btw, very low end on the 7206 line], and at peak we have under 50,000 packets per second, and our 7206 File router-choices.txt not changed so no update needed. [EMAIL PROTECTED] Hardware]$ cat router-choices.txt #1 - http://list.linux-vserver.org/archive/vserver/ #2 - webhostingtalk.com: jharington68/adam123 http://www.webhostingtalk.com/forumdisplay.php?f=44 #3 - asterisk mail list http://lists.digium.com/mailman/listinfo/asterisk-users #4 - cisco mail list? HOTMAIL [EMAIL PROTECTED]pw/adam123 Hello, hope this isn't too far offtopic here but being a troller for a long time here I've realized there is a great knowledge base so I wanted to at least see if i could get some tips.I help run a small colocation company in California and I am in the middle of recommending a new 'core router' platform for our network.We offer mainly colo and dedicated servers, and several of our clients use our space for VOIP services so quality even under high peak usage is a must.We are not huge, but as we have had near 200% growth in the past 12 months and need to expand our network asap to keep up. Simply put, I'd love to hear feedback and/or suggestions from any of you guys who have gone through this already. Our network map is real simple: [Carrier 7609] -- 100 mbit -- Our cisco 7206 -- 100 mbit -- racks [the racks on our end are a series of switches, mainly 2948gl3's] We push about 60 mbit to/from our (1) carrier at peak right now, and the router keeps up fine [its a cisco 7206 npe 150 btw, very low end on the 7206 line], and at peak we have under 50,000 packets per second, and our 7206 has little/no features enabled [just static routes and passing all traffic between 2 Ethernet 100 mbit interfaces]. To date we have had 2 problems, both were DOS attacks launched FROM one of our customer's servers flooding a full 100 mbit wire with more packets per second than the router could handle (the 2948gl3's spiked to about 50% cpu load during the attack but the 7200 literally just died for 3 minutes as the interface(s) all rebooted].So our main goal to grow is something that can handle a lot more in this arena against a DOS, and handle our future growth. In then next 12 months we plan
[Asterisk-Users] Off Topic - Employment Opportunity - PERL, Melbourne, AU.
Sorry if this is off-topic, but I know there's a quite a few smart people who frequent these groups, and I was thinking that it'd be a good place to ask. We have an opening for an experienced PERL programmer. If you (or anyone you know) is interested, please feel free to email me for more details. -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ 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] Off Topic - Employment Opportunity - PERL, Melbourne, AU.
Rod Bacon wrote: Sorry if this is off-topic, but I know there's a quite a few smart people who frequent these groups, and I was thinking that it'd be a good place to ask. We have an opening for an experienced PERL programmer. If you (or anyone you know) is interested, please feel free to email me for more details. Hi, I have 5 years Perl experience, numerous Perl modules on CPAN, and some free time on my hands... - I have written quite a few CPAN modules - I am currently writing AGI scripts I do - Object oriented perl - Like to have proper makefiles, test, and pod that makes things maintainable - use strict; - use warnings; Currently I live in Reunion Island but we could start working immediately on a per-project basis. Regards, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ 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] Off Topic - Employment Opportunity - PERL, Melbourne, AU.
Oops, sorry for the list reply :/ -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ 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] Off Topic - Employment Opportunity - PERL, Melbourne, AU.
Sorry, I should have stated that the position is a FULL-TIME position, based in our Melbourne office. - Original Message - From: Jean-Michel Hiver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 08, 2005 2:55 PM Subject: Re: [Asterisk-Users] Off Topic - Employment Opportunity - PERL,Melbourne, AU. Rod Bacon wrote: Sorry if this is off-topic, but I know there's a quite a few smart people who frequent these groups, and I was thinking that it'd be a good place to ask. We have an opening for an experienced PERL programmer. If you (or anyone you know) is interested, please feel free to email me for more details. Hi, I have 5 years Perl experience, numerous Perl modules on CPAN, and some free time on my hands... - I have written quite a few CPAN modules - I am currently writing AGI scripts I do - Object oriented perl - Like to have proper makefiles, test, and pod that makes things maintainable - use strict; - use warnings; Currently I live in Reunion Island but we could start working immediately on a per-project basis. Regards, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] off topic - DECT phones with FSK VMWI in the UK
Off topic but I am after a DECT phone to connect to my sipura 3000 that has a FSK VMWI light or flashing envelope on the LCD screen. Any ideas 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
Re: [Asterisk-Users] off topic - DECT phones with FSK VMWI in the UK
On 26 Jan 2005, at 13:11, Chris Stenton wrote: Off topic but I am after a DECT phone to connect to my sipura 3000 that has a FSK VMWI light or flashing envelope on the LCD screen. Any ideas Can you post the relevant extension from sip.conf and the contents of voicemail.conf. Also, check your caller ID is set to the UK standard under Regional on your Sipura. Phil. -- Phil Quinney IT Consultant - Any-Ideas ___ 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] off topic - DECT phones with FSK VMWI in the UK
-Original Message- From: Chris Stenton [mailto:[EMAIL PROTECTED] Sent: 26 January 2005 13:12 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] off topic - DECT phones with FSK VMWI in the UK Off topic but I am after a DECT phone to connect to my sipura 3000 that has a FSK VMWI light or flashing envelope on the LCD screen. Any ideas Chris Possibley helpful but possibley not. These two phones work a treat with mediatrix (I would expect the same would be true with Sipura is the 3000 does support VMI), tho to access Vmail you need to manually enter a quick dial or I just haven't found the setting yet: BT Freestyle 2100 BT Calypso 1100 Colour However after buying two to test we found that the cheaper none colour version was much better audio quality so we went with the Freestyles instead. HTH alex This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ 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] [OFF TOPIC] Voip phone sellers in India
check with webtel,Mob:32333033 On Sun, 2005-01-09 at 19:33 +0100, Vikram Rangnekar wrote: I am looking for some in India to buy VOIP phones from. Please get in touch with me off the list on [EMAIL PROTECTED] Sorry for the off topic mail I am just having such a hard time finding any voip phones in India that I got desperate and didnt know which list to post this on. -- Sandeep A.S [EMAIL PROTECTED] Netcontinuum Pvt Ltd ___ 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] [OFF TOPIC] Voip phone sellers in India
I am looking for some in India to buy VOIP phones from. Please get in touch with me off the list on [EMAIL PROTECTED] Sorry for the off topic mail I am just having such a hard time finding any voip phones in India that I got desperate and didnt know which list to post this on. -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] off topic - SSH configuration for Digium Support
I've an issue with my TDM4000P card and I will be calling Digium later to ask for their help. Could anyone help me with a basic configuration so they can SSH to me? Thanks John ___ 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] off topic - SSH configuration for Digium Support
On Fri, 7 Jan 2005 10:36:50 +, John Middleton wrote: I've an issue with my TDM4000P card and I will be calling Digium later to ask for their help. Could anyone help me with a basic configuration so they can SSH to me? On your router you'll need to port forward port 22 to your Asterisk server. Persuming that you already have sshd running on your server that's about it. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Off Topic] humour, XMAS, ground loop - good business strategy
hi, I received this e-mail which contains a ballad, at first I thought it was junk mail, but then I read through it, for the EE members of this list, it may be quite humorous. I don't know if the ballad is original, but at least it's the XMAS season, so it's something to lighten up your day, eh? -samudra How the Ground Loop Stole Christmas Every dude down in dude-ville Liked Christmas a lot... But the Ground Loop, Who lived under Dude-ville, Did NOT! The Ground Loop hated Christmas! And Santa and Miss Claus Now please don't ask why, No one knows the cause. Maybe his connector wasn't screwed on just right. Could be, perhaps, that his crimps were too tight. But I think the most likely cause of it all Might be his cable was one conductor too small. But Whatever the reason, the crimps or the screws, He lurked there on Christmas Eve hating the dudes. They're shopping and bopping, they're trying and dipping, they're chirping and burping and buying and sipping They're hanging their stockings! He growled with a frown Tomorrow is Christmas! I must take them down! The Ground loop was angry, his face full of fury Had the dudes known, They'd have started to worry From deep in his mind, a plan did now hatch So evil and wicked, a way he could catch Dudes and dudettes in the midst of their shopping Action and motion all suddenly stopping His plan was so vile, his scheme was so clever The dudes would ponder and remember forever The dancing and singing and cash registers ringing Would grind to a halt and now become screaming The wicked intent of Ground Loop's foul mind Searched dudeville all over, hoping to find A careless young dude, oblivious to worry Who strung up his wires in too much of a hurry Probing and looking and peering and viewing The eyes of Old Ground Loop were roving and moving At last he did find on late Christmas Eve A witty and gritty technician named Steve So nifty and thrifty and clever was Steve Economy and elegance you wouldn't believe A master he was of superior design Quality, six sigma and kaizen combined Solid and robust and savvy his plan Earthquakes, hailstorms and rain to withstand A world class design, I give you no jive Except for a small problem with 485 Sarah in purchasing had called to contest A thousand dollar reduction was her request For Sarah, young Steve's heart did so yearn Shave pennies he would, her favor to earn In blueprints and drawings and plans he looked The requested amount of savings was booked Her hand at the dance he then requested With hope in her heart, his invitation accepted The music played and dancing ensued A beautiful evening with romance imbued And as she decided his company she liked Ol' Ground Loop with fury did finally strike His target of terror, the signals in town In bedlam and confusion the traffic would drown And just as Steve turned to take her back All the stoplights of dude-ville went Black Chaos and confusion did quickly arise The Ground Loop now claimed his deadly prize Only for want of optical isolation The town of dudeville suffered desolation Riots and shouting and crashes were heard Gridlock and damage and cursing incurred As Ground Loop witnessed this loss of control A tumult of joy filled his dark soul Shamefaced and panicked the couple sat glued A strenuous discussion quickly ensued You asked for cost savings Steven accused You skipped isolation Sarah diffused Upright they bolted as both realized This disaster would get them both downsized They became a team, no longer a faction They opened the door and flew into action Through gridlocked streets the couple dashed To save Dude-ville from Ground Loop's hash Back to the office they breathlessly ran So Christmas in Dude-ville could resume again Raiding his toolbox Steve rummaged with fury Never before had he worked with such hurry Widgets and gidgets and gadgets did fly Until the optical isolator caught his eye With sweat on his brow replaced the foul node And threw the failed unit in the commode The power switch flipped in a flash of commotion Steve prayed for stoplights to begin their motion Despite his best hope, the darkness remained The Christmas disaster his job record stained Down to his knees he fell in despair Until he heard sleigh bells filling the air Up he looked, and to his surprise Eight reindeer and sleigh greeted his eyes Can it really be? Can Santa be real? I thought Santa was a a mythical deal! This santa was young, and he had no fear He had the tools of a data engineer Steve looked at his nametage, now he could see This was Mike Fahrion from BB! Saving your bacon today is my job Restoring dude-ville, yessiree Bob I have now come because one thing you forgot Without changing bias, terminate you must not! He lectured to Steve all about termination Bias and cost cuts and infatuation And then with a wave of his hand he did turn The stoplights of dude-ville to once again burn All around dude-ville the happiness spread The cars and the trucks
Re: [Asterisk-Users] Off topic question
G'day David, On Thu, 26 Feb 2004, David J Carter wrote: Q) Is ADSL a standard? and will his router/modem work in AU? There are a few standards for xDSL. ADSL down here uses the G.Lite standard (maximum speed 1M5/256kbps), so as long as the modem supports that it'll be fine technically (although legally you cannot connect any equipment that doesn't have AU approval, including a foreign version of a device that does have AU approval). Prices for modems offered by ISPs at signup have become so cheap here that it may not be worth your friend bringing the modem with him ;-) Check out www.whirlpool.net.au for info on xDSL ISPs here. Check out OZtell also -- to bring this back on-topic a bit -- an ISP offering xDSL service at good rates and introducing VoIP facilities as well. Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Off topic question
Hi, Sorry for the of topic question, but where else do you get so many telco guys in one place. I have a customer who is moving to Australia and was on ADSL here in the UK. Q) Is ADSL a standard? and will his router/modem work in AU? I have told him a tentative yes but would page the oracles for clarification. Regards Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Off topic: suggestion for call center ?
Hi guys, I am in the process of establishing a Call Center. I need some suggestions from those who have already worked on such setups using Asterisk. My scenerio is: US T1 --- Asterisk gw 1 -[gsm compression]-- Asterisk gw 2 [with TDM10B] --- Sip Phones [Xlite] Calls from US are landing perfectly on the Xlite and tdm10b, and vice versa. However I am having some trouble as mentioned below: 1. Call quality on Tdm10b is good, but sometimes on Xlite, sound becomes choppy (like the voicemissed for few mili secs). I am using jitterbuffer=no is IAX.conf (if set to yes, sounds becomes more choppy). 2. There is lot of echo during the conferencing [on Xlite], when in a room more than 4 people talk at the same time. 3. I am using GSM compression on the Asterisk, and my Xlite also uses GSM, but when I switch from GSM to iLBC in the server (Xlite still using GSM), sound qulaity degrades. Plz recommend your suggestions, and is Xlite good for Call Center ?? Further the echo that I am getting, can be caused due to normal headphones ??? can some guide me which headphone is best in terms of echo cancelling and noise reductions. Did anyone tried the Plantronics PLA-H161N (http://www.thephonesource.com/PLA-H161N.htm) ?? TIA Azher Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing
