RE: [Asterisk-Users] IP Phone Recommendation
You can use the speeddial buttons. They are configurable Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristof Hardy Sent: den 13 december 2005 09:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IP Phone Recommendation Anders Svensson wrote: We use Grandstream GPX2000 for this. It works ok. Support 11 lines in basic. Anders I also use this phone, have read about the 11 lines, but how does one 'manage' these lines? The first 4 are easy, you have buttons for that, but how can you use the 'others' ? (incoming/outgoing) ___ --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] IP Phone Recommendation
We use Grandstream GPX2000 for this. It works ok. Support 11 lines in basic. Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duracom ISP Lists Sent: den 12 december 2005 23:36 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] IP Phone Recommendation We are going to replace our existing PBX system with an Asterisks box. I have 7 phone lines that come in and I need to get a phone that would support that many lines at minimum. Do you guys recommend any phones that you have used that work well. Kris ___ --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] ip phones
The only one I can think of to decent price level is the Grandstream GXP 2000. Also have headset jack¨ Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Fraser Sent: den 28 november 2005 17:27 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ip phones Hi all, Does anybody have any info on a decent quality sip hard phone that is headset compatible? Thank you John ___ --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] Looking for Windows based Asterisk
http://ipswitchboard.thorben.dk/index.php?option=com_contenttask=viewid=26 Itemid=46 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: den 24 november 2005 20:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Looking for Windows based Asterisk I use putty.exe it works wonders. available here: http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html You need ssh running on linux for it to work. On 11/24/05, Stefan-Michael. Guenther (in-put GbR) [EMAIL PROTECTED] wrote: Hi, Does anyone know of a Asterisk Manager Interface client application that can run from a Windows XP machine to manage Asterisk installed on a Linux Machine. if you consider the IE to be a client application, you could use the Asterisk PBX Manager from Thirdlane (www.thirdlane.com). Bye, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which is Better!
We have tried both but given up hope about them. So now we only use Quintum DX series. Amazing machine Anders Svensson Bobas Communication From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Goran Donev Sent: den 22 november 2005 16:41 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which is Better! Which FXO gateway is better and has better sound quality. AudioCodes? Or Mediatrix. Thanks for your input ___ --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 1.2 Aastra/Sayson 480i DTMF Problem
We changed to newest fw released yesterday and out came a new phone. Solved a LOT of problems. Perhaps yours to. Anders Svensson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Pajari Sent: den 22 november 2005 23:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk 1.2 Aastra/Sayson 480i DTMF Problem We are experiencing problems with DTMF when using Asterisk 1.2 and the Aastra/Sayson 480i running 1.2.1.1002 firmware -- callers cannot navigate voicemail or other menus. Of course, we have the sip.conf set to RFC2283 (and nothing changed in our config files between 1.0.9 and 1.2 when things stopped working). Anyone else noticed this? We have a problem report into Sayson but are going to back out from 1.2 and revert to 1.0.9 in about 12 hours because of this and other problems with 1.2. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --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
[Asterisk-Users] E1 Gateway
Hi all! Someone who can recommend a good E1 gateway for terminating VoIP traffic. H323 or Sip Regards Anders Svensson ___ --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] E1 Gateway
I dont think its a good idea to put an * in Bosnia when we are in Sweden. Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: den 21 november 2005 10:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] E1 Gateway Anders Svensson wrote: Someone who can recommend a good E1 gateway for terminating VoIP traffic. H323 or Sip Asterisk! /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DSL router with QOS
If you mean a just a router without modem we sell Draytek in Europe for about 170 usd. Built in QS, 32 VPN tunnels etc. Works great. They also have a bigger model 3300 with up to 8 fxs ports built in. That model also comes with 4 WAN so you can have redundant lines, loadbalance and so on. I am sure it can be bought all over the world Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Keith Schmidt Sent: den 10 november 2005 21:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] DSL router with QOS Any recommendations on an ADSL router with QOS for VOIP built in? Anything sub $500 would be great. Thank you Keith Schmidt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys PAP2: supported codecs
I don't think they want to solve it. It's the same with the Sipura boxes. Only SPA 2100 supports 2 G729 sessions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard James Blundell II Sent: den 10 november 2005 18:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys PAP2: supported codecs The latest pap2 firmware supports G729 on one channel at a time. I think they did this to lower cost on the g729 licensing fees. On Thu, 2005-11-10 at 20:19 +0100, Jose Limeres wrote: Hi, Can someone just confirm whether codecs G.729 and G.723 are operational with the Linksys PAP2 (reseller version, no Vonage version)? I know they are supported but there was a posting from February this year saying that G.729 was operational only in one channel at the time. Apparently they were planning to solve this issue with version 2 in the firmware. Thanks ___ --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 -- Thank-you, Richard James Blundell II Internet Engineer Telepacket, Inc. 27455 Tierra Alta Way, Suite A. Temecula, CA 92590 Direct:(714)263-9090 Mobile:(951)757-5899 Fax:(714)263-9001 E-Mail: [EMAIL PROTECTED] Please note that: This email message may contain confidential and privileged information and is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution, or use of the content of this message is prohibited. If you have received this message in error, please reply by email and delete the material from your computer. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Planet Network - VIP-153
We have the PBX 2000 running as a testbox. Works very good. I am pretty sure that is Asterisk inside. Have not tested the phone. And Planet is a companu that buy most products OEM from other manufacturers so quality can be different on a different product Anders From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: den 10 november 2005 20:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Planet Network - VIP-153 Anyone used a sip from from Planet Network? VIP-153 http://www.planetnw.com/ http://www.planetstoresite.com/Merchant2/merchant.mvc?Screen=PRODStore_Code=PNIProduct_Code=VIP-152TCategory_Code=VOIP ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip provider problem or?
Hi! We are running an * with 3 sip providers. Provider 1 works perfect, provider 2 also. But the 3:rd one is a problem. All seems normal until we try to make a call. The phone rings by the called party and picks is up and hear only silence. The caller (local extension on the *) still gets ring tone as of no one answer the call. The providers ssw treats the call as answered and get no errors Any hints where to start looking? Regards Anders Svensson ___ --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] Sip provider problem or?
Sorry. Forgot to say that if I connect an ip phone directly to the provider it works without problwm Anders From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anders Svensson Sent: den 8 november 2005 11:09 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sip provider problem or? Hi! We are running an * with 3 sip providers. Provider 1 works perfect, provider 2 also. But the 3:rd one is a problem. All seems normal until we try to make a call. The phone rings by the called party and picks is up and hear only silence. The caller (local extension on the *) still gets ring tone as of no one answer the call. The providers ssw treats the call as answered and get no errors Any hints where to start looking? Regards Anders Svensson ___ --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] ATA-488 FXO
Yes you can connect the fxo to a asterisk using sip I have cut out a piece of the manual. It works for m 5.2.7 VoIP-to-PSTN Calls To make a VoIP-to-PSTN call, users need to dial the FXO SIP account phone number first. A ring tone is played once followed by a dial tone. At this time, users can dial a PSTN telephone number or a mobile telephone number then # (or wait for 4 seconds). The call will be established afterwards. If no PSTN number is entered after the dial tone, HandyTone-488 will hang up automatically in 10 seconds. In the web configuration page, if the Route to PSTN field is configured, the second stage dialing is eliminated. That is, after users dial the FXO SIP account number, the PSTN number will be called automatically. 5.2.8 PSTN-to-VoIP Calls To make a PSTN-to-VoIP call, PSTN callers need to originate a call to the FXO port telephone number first. If no one answers the FXS phone after 4 (default value, can be configured) ring tones, a dial tone is played. At this time, users can dial a VoIP telephone number then # (or wait for 4 seconds). The call will be established afterwards. If no VoIP number is entered after the dial tone, HandyTone-488 will hang up automatically in 10 seconds. In the web configuration page, if the Route to VoIp field is configured, the second stage dialing is eliminated. That is, after users dial the FXO port telephone number, the VoIP number will be called automatically. Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Michaelson Sent: den 8 november 2005 19:40 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ATA-488 FXO Is anyone using a Grandstream ATA-488 FXO port to connect a PSTN trunk to their Asterisk box (via SIP, of course)? Is it possible to have such a beast operate reasonably? If so, is it also possible to use the FXS port concurrently and independently? ___ --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
[Asterisk-Users] Uninstall AMP
Hi! How do I uninstall AMP and FOP from my Asterisk? Regards Anders Svensson ___ --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] Uninstall AMP
I agree. It is like we newbie's on Asterisk is just trouble for the list members. Pity there is no newbie list. But all were newbie's in the beginning and not so pompous as some on this list Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Ferrell Sent: den 4 november 2005 19:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Uninstall AMP Claudio Canseco wrote: Really is that the way to uninstall FOP and AMP?, thank you i've been looking for an answer about it. Regards Claudio. No Claudio, That will wipe your system. He's being a smartass. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Uninstall AMP
That doesn't solve much. What I want to do is to stop using AMP and FOP. Best way perhaps is to alter start and stop script but I can't find any info about how to do that Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: den 4 november 2005 19:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Uninstall AMP I want to give the benefit of doubt to the suggestion as I think there is a misunderstanding of the suggested method of removal of AMP. I guess that he was suggesting to remove it from your linux installation by using the rm -rf command as under cd /var/www rm -rf * which will efectively remove all the web pages associated with the AMP instalation. -Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anders Svensson Sent: Friday, November 04, 2005 1:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Uninstall AMP I agree. It is like we newbie's on Asterisk is just trouble for the list members. Pity there is no newbie list. But all were newbie's in the beginning and not so pompous as some on this list Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Ferrell Sent: den 4 november 2005 19:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Uninstall AMP Claudio Canseco wrote: Really is that the way to uninstall FOP and AMP?, thank you i've been looking for an answer about it. Regards Claudio. No Claudio, That will wipe your system. He's being a smartass. ___ --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 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
RE: [Asterisk-Users] Satellite WAN
We have a few satellite trunks for VoIP in Africa and have some experience. Please mail me off list and we can discuss it [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: den 2 november 2005 18:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Satellite WAN We have built an Asterisk network using an MPLS-based IP VPN. We have one location in New Brunswick Canada that consistently gives us major quality problems, whereas the others are flawless. Quality problems take the form of static, poor voice tonality, popping clicking, drops, sporadic echo, you name it. The latency of a QoS prioritized packet between the Canada site and our hub in Atlanta is 85ms (ping). I have been searching for an alternative network provider, but I'm told that they would all take the same route from the US into Canada, as there is simply no major backbone running into NB east of Toronto. So now I'm thinking about satellite. I have no idea if a) this would even be economically feasible, and b) if the latency would be any better. If anyone out there has had any such satellite network experience with VoIP, I like to hear from you. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Satellite WAN
Price is high that is correct but latency is not correct. We have a number of Satellite VoIP Trunks in Africa and no location has more then 500 ms latency. In all locations we have 2 Mbit dedicated lines using C-band and the hub is in the US. But price is HIGH. 6000 usd per month Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Janczuk Sent: den 2 november 2005 20:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Satellite WAN Sattellite links aren't cheap, and, the worst of all, you have in a idel condition, 1.4 seconds latency. Hope this help... Juan. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Adam Robins Enviado el: Miércoles, 02 de Noviembre de 2005 02:01 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] Satellite WAN We have built an Asterisk network using an MPLS-based IP VPN. We have one location in New Brunswick Canada that consistently gives us major quality problems, whereas the others are flawless. Quality problems take the form of static, poor voice tonality, popping clicking, drops, sporadic echo, you name it. The latency of a QoS prioritized packet between the Canada site and our hub in Atlanta is 85ms (ping). I have been searching for an alternative network provider, but I'm told that they would all take the same route from the US into Canada, as there is simply no major backbone running into NB east of Toronto. So now I'm thinking about satellite. I have no idea if a) this would even be economically feasible, and b) if the latency would be any better. If anyone out there has had any such satellite network experience with VoIP, I like to hear from you. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/156 - Release Date: 02/11/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/156 - Release Date: 02/11/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA3000 as trunk - no caller ID
Have you read this? http://voipspeak.net/index.php?option=c . d= Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Higley Sent: den 30 oktober 2005 16:12 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID I am trying to do the SIPURA-3000 as an incoming pstn trunk. I found the how-to on the geekgazette as well, however, my sipura-3000 only just sits and rings and rings and rings. I have set up the peer and the user values, as per the configuration, and when I look at the web status info page of the spa3000 it just says ringing ringing ringing. If I turn on ring-thru-to-line-1 and set it to yes, then the phone rings, but i cannot for the life of me get it to go into the extension that i have defined on the asterisk system. Could someone assist me with this? Thanks. Kerry Garrison wrote: A phone plugged into it will grab the CID on about the second ring and I have adjusted the SPA3000 out to 5 rings with no difference. What gets passed to asterisk is whatever is set in the 3000's Display Name field. If the Display Name field is blank, then nothing comes across and the phones display 'Unknown'. I have been wondering if there is a variable you can put into the display field. There are some fields that use variables like $PROXY and $NOTIFY, of course simply trying $CID uses '$CID' as the caller ID. You don't need any clever manipulation tricks with the current firmware. Have you got PSTN CID for VOIP CID set to yes ? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA3000 as trunk - no caller ID
http://voipspeak.net/index.php?option=com_contenttask=viewid=24Itemid=27 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Higley Sent: den 30 oktober 2005 18:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SPA3000 as trunk - no caller ID That link is not found Have you read this? http://voipspeak.net/index.php?option=c . d= Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Higley Sent: den 30 oktober 2005 16:12 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID I am trying to do the SIPURA-3000 as an incoming pstn trunk. I found the how-to on the geekgazette as well, however, my sipura-3000 only just sits and rings and rings and rings. I have set up the peer and the user values, as per the configuration, and when I look at the web status info page of the spa3000 it just says ringing ringing ringing. If I turn on ring-thru-to-line-1 and set it to yes, then the phone rings, but i cannot for the life of me get it to go into the extension that i have defined on the asterisk system. Could someone assist me with this? Thanks. Kerry Garrison wrote: A phone plugged into it will grab the CID on about the second ring and I have adjusted the SPA3000 out to 5 rings with no difference. What gets passed to asterisk is whatever is set in the 3000's Display Name field. If the Display Name field is blank, then nothing comes across and the phones display 'Unknown'. I have been wondering if there is a variable you can put into the display field. There are some fields that use variables like $PROXY and $NOTIFY, of course simply trying $CID uses '$CID' as the caller ID. You don't need any clever manipulation tricks with the current firmware. Have you got PSTN CID for VOIP CID set to yes ? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GSM cards / mobile phone cards for Asterisk?
Only the pricing is not that fantastic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev Sent: den 28 oktober 2005 15:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] GSM cards / mobile phone cards for Asterisk? Get VoiceBlue VoIP GSM gateway. It works very well with asterisk. I have been using it for the last 4 month and its fantastic! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomasz Chmielewski Sent: Friday, October 28, 2005 10:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] GSM cards / mobile phone cards for Asterisk? I was wondering if there is something like that on this Earth: Some of our users are mobile users - they are rarely in one place for longer than 15 minutes. They use mobile phones a lot. From our mobile operator we have an offer which allows us to call for ___ --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
[Asterisk-Users] Whats wrong with incoming
Hi! Can anyone help a newbie with this problem. Everything works except my incoming SIP. Get this in the log Oct 21 14:04:25 VERBOSE[2696]: 10 headers, 0 lines Oct 21 14:04:25 DEBUG[2696]: # Testing 83.140.41.62 with 192.168.1.0 Oct 21 14:04:25 DEBUG[2696]: Target address 83.140.41.62 is not local, substituting externip Oct 21 14:04:25 VERBOSE[2696]: Sending to 83.140.41.62 : 5060 (non-NAT) Oct 21 14:04:25 VERBOSE[2696]: Transmitting (no NAT): SIP/2.0 481 Call Leg Does Not Exist Via: SIP/2.0/UDP 83.140.41.62:5060;branch=z9hG4bK4b2f50a2 From: 0703171306 ;tag=as59af657d To: ;tag=as0f1899e3 Call-ID: [EMAIL PROTECTED] CSeq: 103 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extension dialing out
Newbie warning Hi! Can I setup an extension that dial out directly to the phone number I have with my sip provider. Like dial exten 110 and it connects to my sip phone number Regards Anders Svensson ___ --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] SIP to SIP sadness
Look in rtp.conf. You must have the same udp-ports open as the settings in rtp.conf Anders From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Furdyk Sent: den 17 oktober 2005 21:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP to SIP sadness Okay so it seems like it was the firewall, someone just suggested that we disable it (On Redhat server) and it's working fine... so does anyone know clearly what ports (other than 5060) SIP uses for these calls? -- Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Furdyk Sent: October 17, 2005 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP to SIP sadness Wow, after getting the O'Reilly book delivered last week along with two Digium TDM400P's,I'm really getting the hang of this. But the SIP to SIP issue is still a problem... and it seems silly because everything else (should have been?) so much harder but is working pretty flawlessly. Basically I get no audio either way, and it tries to do a native bridge (handoff?) So when I dial another SIP extension, I get: --- -- SIP/324-ab4d answered SIP/322-7e8d We're at 192.168.1.195 port 16874 Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (NAT) to 192.168.1.24:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.24:5060;branch=z9hG4bK8E845F95F34044ACA77C54EF28288C32;received=192.168.1.24;rport=5060 From: Michael Furdyk sip:[EMAIL PROTECTED];tag=411158625 To: sip:[EMAIL PROTECTED];tag=as6606adb1 Call-ID: [EMAIL PROTECTED] CSeq: 30931 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 239 v=0 o=root 3348 3348 IN IP4 192.168.1.195 s=session c=IN IP4 192.168.1.195 t=0 0 m=audio 16874 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/322-7e8d and SIP/324-ab4d -- SIP read from 192.168.1.24:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKB644BADE71EF4422878597A96BE8D613 From: Michael Furdyk sip:[EMAIL PROTECTED];tag=411158625 To: sip:[EMAIL PROTECTED];tag=as6606adb1 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 30931 ACK Max-Forwards: 70 Content-Length: 0 Here is my default in SIP.conf. Each SIP config has canreinvite=no [general] disallow=all allow=gsm allow=ulaw nat=no canreinvite=no externip=(real external IP is here) localnet=192.168.1.195/255.255.255.0 srvlookup=yes sipdebug=yes I have tried nat=no and nat=yes ___ --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] 2 POTS to
Hi! I would recommend a 8 port fxs MOSA 3708 from Vodtel. Works perfect with Asterisk and a reasonable priced compared to Quintum Tenor. Can be bought on www.bobascom.com. The webshop is in Europe but sell equipment to customers all over the world Anders From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: den 14 oktober 2005 22:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 2 POTS to I dont think the Quintum hardware supports SIP devices (just SIP trunks). -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claudio Canseco Sent: Friday, October 14, 2005 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 2 POTS to Hi all, Im trying to build an small home system. I have 2 pots lines, and i need to make 8 extensions and be able to use my old analog phones. What would you recommend to use asthe 8FXS switch? I saw some equipment from quintum, they have a Tenor AS that offer 4 FXS ports. But i don't know if it is the best solution. Does anyone have a better solution to build this system? If an analog switch for 2 incoming POTS to 8 POTS is a better solution, i would appreciate ifyou could point me to posibles solutions. But I would prefer not to lose the IP option, so later i could ad some ip phones, or softphones, and be able to make calls to FWD numbers, etc, through my internet connection. Regards, tia Claudio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX ATA
Hi! Has anyone tested this IAX ATA? Their free softphone is GREAT https://www.virbiage.com/products.php Regards Anders Svensson ___ --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] IAX ATA
Yes I was interested to test them. They are not available on the link you submitted either Anders From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francis Ballares (VoIPware.ca) Sent: den 13 oktober 2005 15:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX ATA Are you looking on purchasing one? francis www.VoIPware.ca On 10/13/05, Anders Svensson [EMAIL PROTECTED] wrote: Hi! Has anyone tested this IAX ATA? Their free softphone is GREAT https://www.virbiage.com/products.php Regards Anders Svensson ___ --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 -- Regards, Francis Ballares E-mail: ballares (at) gmail.com www.VoIPware.ca ___ --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] IAX ATA
We are Voip distributors in Europe so we can buy them from Manufacturer cheaper. Anders From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francis Ballares (VoIPware.ca) Sent: den 13 oktober 2005 17:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX ATA I have other IAX ATA's available at VoIPware.ca - I have tested them personally and they work great. thanks, Francis www.VoIPware.ca On 10/13/05, Anders Svensson [EMAIL PROTECTED] wrote: Yes I was interested to test them. They are not available on the link you submitted either Anders From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Francis Ballares (VoIPware.ca) Sent: den 13 oktober 2005 15:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX ATA Are you looking on purchasing one? francis www.VoIPware.ca On 10/13/05, Anders Svensson [EMAIL PROTECTED] wrote: Hi! Has anyone tested this IAX ATA? Their free softphone is GREAT https://www.virbiage.com/products.php Regards Anders Svensson ___ --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 -- Regards, Francis Ballares E-mail: ballares (at) gmail.com www.VoIPware.ca ___ --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 -- Regards, Francis Ballares E-mail: ballares (at) gmail.com www.VoIPware.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TTL
Hi! Perhaps newbie but I cant find somewhere to set the TTL for sip registration when * acts as client Regards Anders Svensson ___ --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] telephony that just works
What you are looking for is a pc2phone dialer. This can be preconfigured with all settings and when it connects to your * it ask for username and password or just a pin. There are many of these out on the net. Most is however locked to a provider but you will also find many that you can buy with your settings in them. Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of lenz Sent: den 10 oktober 2005 13:28 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] telephony that just works Hello list, I am looking for a way to have multiple remote Windows users download a package and get connected to *. My idea would be that they run a simple app, it connects without any setting to an * box (maybe via IAX) and then people press a button to talk. It would be okay if they had to enter a username and password, but not more than that. Looking for such software, I keep finding how much easier for a non-technical end-user is to download skype and have it running than downloading a softphone, creating an account, configuring the softphone and then dialing the required number. Having a way to use skype as a terminal would be nice, but I fear it's impossible by now (see http://www.skypejournal.com/blog/archives/2005/03/skype_strategy.php ). So, anybody has experience of something that could be used, repackaged, modified or you-know-what that could be helpful in this case? And don't you think a IAX intercom could be somehow useful? :-) Bye l. -- Assum est, versa et manduca. ___ --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
[Asterisk-Users] AAH. only 1 ring
Hi! I have problem with my AAH. I have set up a sip channel. It works perfect both ways with one exception. When someone calls in I only get 1 signal. The caller have normal ringtone until message is played. Anyone who can help? Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming sip
Hi! I use AAH and have 2 sip peers. First one is working perfect both ways. Now I have set up another on and it works perfect for calling out but I get busy when I try to call in. If I use an IP-phone connected directly to the provider it is no problem. Anything special to think about when you have more then 1 provider? Regards Anders Svensson ___ --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] DLINK DVG-3004S
Hi! What exactly are you looking for regarding the fxo gateway. Perhaps we can help you Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: den 5 oktober 2005 23:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] DLINK DVG-3004S Does anyone have any experience using this DLink quad FXO SIP gateway with Asterisk? I'm still looking for an analog interface that I can live with, having tried X101p, SPA-3000 and TDM400...all with less than desirable results. 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 fwd 54245 ___ --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
[Asterisk-Users] Outgoing busy
I have a problem. Incoming calls work without problem but I cant call out. Using AAH.Gets a busy tone Anyone who can see a mistake in Outgoing settings context=from-pstn host=ipkund1.rixtelecom.se insecure=very nat=yes secret=xxx type=peer username=0406082250 Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial pattern sort order
Hi! Is there a simple way for an * newbie to force * to use different sip-trunks for different calls. I have 2 siptrunks, one for inland calls and one for international calls. All in country numbers starts with 0 and all international starts with 00. This I have configured in the outbound routing. But * always use the incountry trunk because the 0. dialpattern is also true for international calls How to fix this? Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto attendant
Hi! Where can a newbie find some info about how to set up an auto attendant extension? Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk behind nat
Hi! How do I configure my * to have a remote extension if the asterisk is behind a nat? Regards Anders Svensson ___ --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] SIP-CPE Gateway
Just to clarify. These products are not produced by this company, its Taiwanese brands. The SIP-CPE Gateway is a rebranded VodTel MOSA 3700 Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Michaelson Sent: den 3 oktober 2005 18:12 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP-CPE Gateway Has anyone used the GSM-SIP gateway product produced by a company at sipcpe.com? Any comments? ___ --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
[Asterisk-Users] Outgoing rout dialpattern
Hi all! How do I create a dialpattern in an outgoing rout that sends all calls starting with 00 AND calls starting with 1-9 to same trunk Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream GXP2000
Hi! Cant get my Grandstream GXP 2000 to register on my AAH. All other Grandstream units work fine. Something extra to think about? Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transcoding
Hi all! Is it possible to have a setup with a server only dedicated for transcoding from ulaw/alaw to G729. What is the capacity of a server like that in simultaneous calls? Regards Anders ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Required hardware
Hi all! We have to setup 2 *servers. Now I am interested in possible capacity. Server 1. Should be used for getting traffic from our Telco using IAX and send it out using SIP. No transcoding, ulaw both ways. What is possible capacity on 1 server using required hardware? Server 2 will pick uo traffic from 1 Tollfree number sent to us by IAX with ulaw codec. Must be transcoded to G729 and sent out as SIP. What is possible capacity on 1 server using required hardware? Regards Anders Svensson CTO BoBas Communication Glimminge 2045 S-280 60 Broby Sweden Phone: +46 (0)40 608 22 50 Cell: +46 (0)703 17 13 06 MSN: [EMAIL PROTECTED] Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NAT
Hi! Finally I have been able to install AAH and its up and running. I am behind a router and believe I have to configure this somewhere but cant do this with AMP. Can somebody hint a newbie about how to do it Regards Anders Svensson ___ --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] NAT
Yes that I have done. But don't I have to configure the gateway ip somewhere? anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Sent: den 29 september 2005 19:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] NAT At 12:14 9/29/2005, Anders Svensson, wrote: Hi! Finally I have been able to install AAH and its up and running. I am behind a router and believe I have to configure this somewhere but cant do this with AMP. Can somebody hint a newbie about how to do it You should be able to do it in AMP. What extension are you using? Extensions - NAT (change from never to yes) Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT
Hi Thanks! 2 questions Can externalip be a dns-address? How do I configure the Incoming settings in the siptrunk? I can call out using the trunk but get busy tone when I try to dial in. Use AAH Thanks Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Samy Antoun Sent: den 29 september 2005 20:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] NAT Anders, There are 2 ways to acomplish this: 1. Keep your Asterisk box behinde your router. In this case you need to do this: - Port Forward to Asterisk Box UDP 5060 UDP 1-2 - sip.conf (Change as needed) externip=xxx.xxx.xxx.xxx localnet=192.168.1.0/255.255.255.0 - Extensions nat=yes qualify=yes 2. Configure your Asterisk box as a router. In this case you need: - Asterisk box must have 2 NIC's - A network switch - Setup instructions can be found at: http://samyantoun.50webs.com/asterisk/firewall/firewall.htm Hope this helps. __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.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 ___ --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] NAT
Thanks!! Problems solved. Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Samy Antoun Sent: den 29 september 2005 20:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] NAT Anders, There are 2 ways to acomplish this: 1. Keep your Asterisk box behinde your router. In this case you need to do this: - Port Forward to Asterisk Box UDP 5060 UDP 1-2 - sip.conf (Change as needed) externip=xxx.xxx.xxx.xxx localnet=192.168.1.0/255.255.255.0 - Extensions nat=yes qualify=yes 2. Configure your Asterisk box as a router. In this case you need: - Asterisk box must have 2 NIC's - A network switch - Setup instructions can be found at: http://samyantoun.50webs.com/asterisk/firewall/firewall.htm Hope this helps. __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk installation
Here comes a newbi question. I have got a used Dell Optiplex GX240 P4 1.7 Ghz with 512 Ram. I have downloaded [EMAIL PROTECTED] and burnt a bootable CD. But the Computer doesnt start the installation. What can be wrong? The computer is empty with anything on the harddrive. Is it something wrong with the CD or something else. Please help ANders ___ --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 installation
It seems that it must be the cd that is bad. The Dell has an option to hit F12 to manually decide bootmedia. But even if I choose the cd nothing happens. It works with other bootable cd's without problem. I have never done a bootable cd before and now I am not sure if I do right. What emulation shall I have? None or harddrive? I have done both but same result I have burnt both with Nero And Easy CD creator. Lot of wasted cd's. :-)) Thankful for all help!!! Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: den 28 september 2005 22:49 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk installation Actually with Optiplexes (Optipii?) they have a hidden partition on the HDD that runs the F2 setup, if the HDD was formatted you're hooped for F2 setup. The key combination you are looking for is ctrlaltf8 which allows you to manually specify boot media during POST before the OS starts loading. Be quick, 'cause these things boot fast. hth -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 28, 2005 2:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk installation Try going into the BIOS and changing the boot order so your CD-Rom drive is first in the boot sequence. Make sure your IDE cable on the CDRom is not loose, and make sure it is jumpered properly as a slave on your primary IDE BUS, or master on your secondary IDE BUS. When you reboot, you should at least see the led light up on the CD-Rom and it should spin up, if it's doing both those things and still not booting, you probably have a bad CD, or a defective ROM Drive. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory Anders Svensson wrote: Here comes a newbi question. I have got a used Dell Optiplex GX240 P4 1.7 Ghz with 512 Ram. I have downloaded [EMAIL PROTECTED] and burnt a bootable CD. But the Computer doesn't start the installation. What can be wrong? The computer is empty with anything on the harddrive. Is it something wrong with the CD or something else. Please help ANders ___ --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 ___ --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 installation
There is 2 files. BOOTCAT and BOOTIMG Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: den 28 september 2005 23:26 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk installation Let me ask you one major thing. Look at the CD filesystem on another computer. Did you perchance burn the ISO as a file on the CD instead of burning the ISO image to the CD? --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Anders Svensson -Sent: Wednesday, September 28, 2005 5:18 PM -To: 'Asterisk Users Mailing List - Non-Commercial Discussion' -Subject: RE: [Asterisk-Users] Asterisk installation - -It seems that it must be the cd that is bad. The Dell has an -option to hit -F12 to manually decide bootmedia. But even if I choose the cd -nothing happens. It works with other bootable cd's without problem. -I have never done a bootable cd before and now I am not sure -if I do right. - -What emulation shall I have? None or harddrive? I have done -both but same result - -I have burnt both with Nero And Easy CD creator. Lot of -wasted cd's. :-)) - -Thankful for all help!!! - -Anders - --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Colin Anderson -Sent: den 28 september 2005 22:49 -To: 'Asterisk Users Mailing List - Non-Commercial Discussion' -Subject: RE: [Asterisk-Users] Asterisk installation - -Actually with Optiplexes (Optipii?) they have a hidden -partition on the HDD that runs the F2 setup, if the HDD was -formatted you're hooped for F2 setup. - -The key combination you are looking for is ctrlaltf8 -which allows you to manually specify boot media during POST -before the OS starts loading. Be quick, 'cause these things -boot fast. - -hth - --Original Message- -From: Cory Andrews [mailto:[EMAIL PROTECTED] -Sent: Wednesday, September 28, 2005 2:37 PM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: [Asterisk-Users] Asterisk installation - - -Try going into the BIOS and changing the boot order so your -CD-Rom drive is first in the boot sequence. Make sure your -IDE cable on the CDRom is not loose, and make sure it is -jumpered properly as a slave on your primary IDE BUS, or -master on your secondary IDE BUS. - -When you reboot, you should at least see the led light up on -the CD-Rom and it should spin up, if it's doing both those -things and still not booting, you probably have a bad CD, or -a defective ROM Drive. - -Cory J Andrews -Partner / Purchasing -+++ -VOIPSupply.com - Everything you need for VOIP -454 Sonwil Drive -Buffalo, NY 14225 -+++ -tf voice - 800-398-VOIP X22 -l voice - 716.630.1555 X22 -f - 716.630.1548 -e - [EMAIL PROTECTED] -AIM - b2Cory - - - -Anders Svensson wrote: - - Here comes a newbi question. - - I have got a used Dell Optiplex GX240 P4 1.7 Ghz with 512 -Ram. I have - downloaded [EMAIL PROTECTED] and burnt a bootable CD. But the Computer - doesn't start the installation. - - What can be wrong? The computer is empty with anything on -the harddrive. - - Is it something wrong with the CD or something else. - - Please help - - ANders - - --- - -- - - ___ - --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 - - -___ ---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
RE: [Asterisk-Users] Satellite Broadband and VOIP
What provider to use depends of course of witch country you live in. We have a lot of customers in Africa who use iwayafrica. Many of the providers block Voip because they have own Voip service. For US we use New Era Systems, Inc Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Anthony C. Delfin Sent: den 26 september 2005 12:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Satellite Broadband and VOIP Hi Sean, We operate a VSAT network here in the Philippines (using Shiron, FDMA Bandwidth on Demand) and offer VoIP using asterisk. We do not sell our voip to our gilat clients since gilat has a higher latency (since it uses TDMA). Try to look for a satellite provider that has an average (to your country of voip destination) latency of below 600-800 ms and it must be consistent. Also since, satellite has low upload bandwidth, try to have QoS behind the satellite modem and prioritize VoIP traffic cxpcman wrote: Sean Rima wrote: I live in a very rural area, BB access will never happen and the only choice I have it Satellite. I seen from a post to this list that Gilat sat modems are not recommended. Is this still the case or is there another alternative? Sean ___ --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 Well is not recommended because of the seektime . the information you send and recive have a delay no matter how fast your conection is .. so you gonna hear the voice out of time . wire have a lot faster response times than air soo... ur choice ___ --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
[Asterisk-Users] Codec routing?
Hi! I asked this question a couple of days ago but got no answer so I try again. Is it possible to route a call in * based on used codec, meaning that if a user use G723 that call is routed to siptrunk 1 and a user using G.729 is routed to siptrunk 2? Regards Anders Svensson ___ --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] Best Voip provider
There is also www.talkycallshops.com Very good rates, no monthly fee Unlimited number of numbers. Voip DIDs in 22+ countries. Anders From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nana Tandoh Sent: den 25 september 2005 12:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best Voip provider Termilink Digital Voice www.termilink.net On 8/13/05, jonny hashem [EMAIL PROTECTED] wrote: what is the best voip provider that provides good service ,good voice quality and good rates . any one havean experience with voip providers advice me. Regards; jonny Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Voip provider
For the moment only sip. But they will set up an * to be able to provide IAX too but it will take some time yet. Mail to [EMAIL PROTECTED] and explain what you want and they will get back to you tomorrow. Its bedtime in Sweden now ;-) Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shawn Rutledge Sent: den 25 september 2005 23:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best Voip provider On 9/25/05, Anders Svensson [EMAIL PROTECTED] wrote: There is also www.talkycallshops.com That looks interesting. Do they offer iax service or sip only? Do you have any .conf example? ___ --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
[Asterisk-Users] Seperate siptrunks
Hi all. Is it possible to get * to send calls to different sip trunks depending on what codec the incoming call use? This to avoid transcoding Anders ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CPU load
Hi! Here comes a newbi question. I now that transcoding of codecs take a lot of cpu load. But if I want to receive all traffic as IAX and then want to send it out as SIP. Is it the same? Requires a lot of CPU and RAM? Regards Anders Svensson ___ --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] Which codec?
This is a good link http://www.erlang.com/calculator/lipb/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Journo Sent: den 23 september 2005 11:20 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Which codec? Is there a guy somewhere on how much bandwidth each codec uses, along with the advantages and disadvantages of each one? Dan Journo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Double cpu
Hi! Probably another newbie question. Is it possible to run * on one processor and MySql on the other in a double cpu server? Anders ___ --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] Cisco Ip phones
Have you tested Aastra. Works great with * and reasoable pricing Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: den 20 september 2005 20:57 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco Ip phones On 20:38, Tue 20 Sep 05, Florian Overkamp wrote: Hi Sander, Sander wrote: Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap Cisco phones work fine using SIP, good reports have also been seen with SCCP/Skinny, although my own experience on that is limited. We use SwissVoice a lot and others have reported great success with Polycom. I been using some Cisco phones for a while now. I started with converting them to SIP so they could connect to * Now with chan_sccp I reverted them all back to SCCP and they work awesome. Too bad they are so darn expensive, otherwise I wouldn't use anything else. Just my experience :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] internet connection between Africa and Europe
Many of the satellite companies block voip because they have the sevice for sale them selfes. And dedicated satellite internet is VERY expensive. We arranged a 512/512 connection today for a callcenter in Nigeria and they will pay 6000 usd per month. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan de Konink Sent: den 15 september 2005 21:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] internet connection between Africa and Europe Check out google with: VSAT Africa, lots of companies provide IP links overthere. If it is good enough for voip... I don't yet know. Stefan On Thu, 15 Sep 2005, Jean-Michel Hiver wrote: Stéphane LAVRI a écrit : Hi I'm looking for a company who can provide me an Internet connection between africa and Europe. 'Africa' and 'Europe' are both rather big, so what you're saying doesn't make much sense. Pehaps if you outlined your requirements a bit better, you could get some useful advice. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T.38 ATA
The MOSA 3700 family from Vodtel have working T.38. They come from 2 to 16 ports. Can be bought on www.bobascom.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moody Sent: den 14 september 2005 14:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T.38 ATA Can anyone recommend me ATA device that REALLY has T.38 built in. While I have not tested it myself (one just arrive for me try out), I have been told that the Mediatrix products have a working T38 implementation. Of course my suggestion would be check with the provider tho you plan to use the product with and see what they suggest/have seen work before. This is the base product... http://www.voipsupply.com/product_info.php?manufacturers_id=16products_id=334 I don't think the current Sipura firmware (for any model including the 2100) supports T38 yet. J ___ --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] Skype purchased by Ebay 2.6 Billion
Skype prices is not that low. F.ex buying price today for Argentina Buenos Aires is between 0,0050 and 0,0056 Euro and Skype charge 0,0170 euro. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: den 13 september 2005 11:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion On Mon, Sep 12, 2005 at 06:00:25PM -0700, Matt wrote: anyone knows how skype provide world wide call service to regular phones by voip at such low rate? is this by partnerships with various * isps? Skype do deals with local telcos, they use G.729/SIP for SkypeIn/Out between them and the local player. Steve -- NetTek Ltd Fax +44-(0)20 7483 2455 Skype / In stevekennedyuk / UK +442088167166 / US +13106518226 Vonage UK +442079932612 / US +13108577715 / UK mob 07775 755503 Personal Blog http://stevekennedy.blogspot.com Euro Tech News Blog http://eurotechnews.blogspot.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 ___ --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] Configuring SIPURA 2002 to work wih Asterisk
Have you read this article? Its about Sipura 2000 and Asterisk but have much valuable info. http://voxilla.com/modules.php?op=modloadname=Newsfile=articlesid=39 Anders From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sander Sent: den 11 september 2005 09:31 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Configuring SIPURA 2002 to work wih Asterisk you can try to post your sip.confso someone can help the sipura spa 2002 works perfectly with asterisk Sander Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Paul Conn Verzonden: zaterdag 10 september 2005 23:15 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] Configuring SIPURA 2002 to work wih Asterisk Im setting up Asterisk for the first time. I purchased a SIPURA 2002 ATA to connect with the Asterisk server. In the /var/log/asterisk/messages log I keep getting an error indicating wrong password. Below is the error I am receiving. Note that the IP address and username has been modified for security. Sep 10 15:56:22 NOTICE[24099] chan_sip.c: Registration from 'John Doe sip:[EMAIL PROTECTED] ' failed for '192.168.1.5' - Wrong password In the sip.conf file under the extensions I have the secret set the same way as the password in the SIPURA 2002 GUI under the LINE 1 parameters. Anyone successfully configured the SIPURA 2002 to work with Asterisk OR does anyone know of any help documents (other than the SIPURA PDF) that explains the configuration of the 2002 for use with asterisk? Thanks! Paul ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Required hardware
Hi! What hardware is required for Asterisk to handle 30 simultaneous calls? All sip to sip. Regards Anders Svensson CTO BoBas Communication Glimminge 2045 S-280 60 Broby Sweden Phone: +46 (0)40 608 22 50 Cell: +46 (0)703 17 13 06 MSN: [EMAIL PROTECTED] Email: [EMAIL PROTECTED] ___ --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
