[asterisk-users] Does any one uses PortSIP VoIP SDK?
Does it working good with RFC standard? Or where can I get a crack version? Thanks -- _ -- 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] Video not working with PortSIP SDK
Hi, I'm setup the Asterisk 1.4.33 and try test it with the PortSIP SDK( www.portsip.com), but seems the video does not works. When I make the call from PortSIP SDK Demo to GrandStream GXV3140, it's working fine if no video codec selected. If make call with H.264 codec, the PortSIP got 503 service unavailable response from Asterisk, why? Thanks -- _ -- 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] Asterisk desktop tools for OS X
Count me in. On Jan 17, 2008, at 6:25 AM, Devraj Mukherjee wrote: Hi everyone, I have been long working on a project (http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- 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] Voicemail to Email on Blackberry
Blackberry's won't do it. check https://www.blackberry.com/knowledgecenterservices/efaq/ResultPage.asp?FAQId=34 search for "audio attachment" On Jun 8, 2006, at 12:00 PM, Kerry Garrison wrote: Is there any setting in the voicemail that will send the voicemail file in a type that is recognized on a Blackberry? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
I'm willing to bet the phones that are stalling have the most active computer users attatched to them. I wouldn't advise having the computer running through the phones port. To me that is asking too much out of the $100 phone.Run each device from it's own port on your switch.On Jun 7, 2006, at 9:36 PM, Daniel Salama wrote:They are extremely casual web surfers. Just have their Outlook client opened checking email every minute. Email traffic is very low.They are all connected to the same switch. It's a Netopia DSL router/modem/switch for the BellSouth DSL service. The computers are connected to the PC port behind the GXP-2000.Any suggestions?Thanks,DanielOn Jun 7, 2006, at 8:49 PM, list mail wrote:What do they do on the internet? Heavy surfing, large transfers, myspace. How are these units connected to the network? Are they passing through the same switch?I don't think it is the phones...On Jun 7, 2006, at 12:32 PM, Daniel Salama wrote:Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun 7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (66ms / 500ms)Jun 7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (68ms / 500ms)Jun 7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now TOO LAGGED! (1114ms / 500ms)Jun 7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now REACHABLE! (90ms / 500ms)Jun 7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1077ms / 500ms)Jun 7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (72ms / 500ms)Jun 7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now TOO LAGGED! (1077ms / 500ms)Jun 7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now REACHABLE! (73ms / 500ms)As you can see, it only happens to a couple of their phones and at random times. They're behind a DSL circuit. I don't know if it's because their DSL line is going up/down. They don't necessarily claim their Internet goes down, however, they are not constantly check it.What would you (or anyone else) suggest?Thanks,DanielOn Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:Do you have multiple phones going down at the same time? If so, monitor them with "qualify=500" in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail to Email on Blackberry
Blackberrys won't.https://www.blackberry.com/knowledgecenterservices/efaq/ResultPage.asp?FAQId=34search for "audio attachement"On Jun 8, 2006, at 12:00 PM, Kerry Garrison wrote: Is there any setting in the voicemail that will send the voicemail file in a type that is recognized on a Blackberry? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite
Sounds like they crippled the phone for cellulars sake.On Jun 7, 2006, at 10:35 AM, Jon Schøpzinsky wrote: Hello Olivier Ive been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work.That means, that you cant just walk down to your local airport, and use the IP part of the phone on their network.You have to have a non nat local server, to get it to run.Other than that, the phone can accept calls both from cellular network and IP network, and actuatly works quite well, both for cellular and IP traffic.But you cant do seamless handover, for example when you walk out of the building. You have two different numbers, your mobile number and your IP number And these cant automaticly be transferred. Hope this answeres your question RegardsJon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] På vegne af Olivier Krief Sendt: 7. juni 2006 16:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite 2006/6/7, Jon Schøpzinsky [EMAIL PROTECTED]:Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon What do you mean by " users has to have some local equipment from the telco" ? Do you think Nokia E60, E61 and E70 are appropriate for Fixed Mobile Convergence (each mobile phone being reachable at the same time from inhouse PBX and Telco's mobile network without any handover or roaming between both networks) ? Regards -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: GXP-2000 (steer clear)
I read all the stories and did the homework, but one client wanted under 100/phone. I explained the issues and we went ahead. I have had not one problem with 18 units purchased from VOIPsupply. They sound very good via handset. Can't comment on the speaker phone quality. For the money, they are good phones in my opinion. On Jun 7, 2006, at 8:26 AM, Francesco Peeters (Asterisk) wrote: On Wed, June 7, 2006 14:09, Louis-David Mitterrand said: On Tue, Jun 06, 2006 at 11:26:20PM -0400, Daniel Salama wrote: Well, these are encouraging words :) You're basically telling me that I should tell my client to buy other phones. I agree that you cannot compare these phones with Cisco or Polycom. After all, like you said, what do you expect for under $90. However, the fact is that my client just recently invested in these and it will be hard, if not impossible, for me to tell my client to swap them for Polycoms or something else at a much higher cost. I have heard complaints from my client about the speakerphone and they are now, I guess, getting used to picking up the handset :). I have heard any echo problems so far. What bothers me the most is that the phone stops working often (multiple times per day). By this I mean that my client won't be able to dial anything successfully. As soon as 3 or 4 digits are entered, they get a fast busy. To solve it, they need to reboot it. It sounds as if these phones were running Windows instead of Linux :) Anyway, what firmware did you use that solved so many of your problems? I've only had bad experiences with these phones and steer clear of them. In the same price range you can now get the Thomson ST-2030 or Polycom 430 for a much, much better user experience. Where do you purchase the Thomson or Polycoms for a comparable price as the GXP2000? I'd like to buy an ST2030 or 430 for under EUR 90 myself too! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] GXP-2000
What do they do on the internet? Heavy surfing, large transfers, myspace. How are these units connected to the network? Are they passing through the same switch?I don't think it is the phones...On Jun 7, 2006, at 12:32 PM, Daniel Salama wrote:Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun 7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (66ms / 500ms)Jun 7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (68ms / 500ms)Jun 7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now TOO LAGGED! (1114ms / 500ms)Jun 7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now REACHABLE! (90ms / 500ms)Jun 7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1077ms / 500ms)Jun 7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (72ms / 500ms)Jun 7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now TOO LAGGED! (1077ms / 500ms)Jun 7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now REACHABLE! (73ms / 500ms)As you can see, it only happens to a couple of their phones and at random times. They're behind a DSL circuit. I don't know if it's because their DSL line is going up/down. They don't necessarily claim their Internet goes down, however, they are not constantly check it.What would you (or anyone else) suggest?Thanks,DanielOn Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:Do you have multiple phones going down at the same time? If so, monitor them with "qualify=500" in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
