Re: [Asterisk-Users] Broadvoice asterisk patch
I was just about to ask a similar question having just received the message. I'm more concerned about someone trying to spread a virus or something like that. You have to admit that the URGENT, INSTALL THIS message with an attachment pretty much screams virus, even if its not. I tried calling Broadvoice support but they want me to leave a message for them to call me later. Can anyone comment on the validity of this message? thanks, Ryan Wilkins On Nov 10, 2004, at 2:54 PM, [EMAIL PROTECTED] wrote: Just received this from broadvoice, anyone know if this patch will become part of the CVS tree? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice asterisk patch
I agree that sending a patch out via email blindly is not the appropriate method. It would have been much better to send the email as they did but provide a link to download the patch from the Broadvoice website. This would help verify the authenticity of the patch and not cause the discussion that it did. In any case, the patch has been positively identified as being genuine. On Nov 10, 2004, at 5:14 PM, Michael Giagnocavo wrote: That's not the point. The point is distributing patches via email is a horrible way to do patches, and teaches users to just trust what comes in the mail. It should be put on a site that's trusted and easily verified and a notice of that sent out. Even Microsoft has this down. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS Router/Software Suggestions
For what it's worth, I just shutdown my PC based gateway last week and replaced it with an Efficient Networks 5861 ADSL router. The 5861 is billed as a Business Class DSL Router. It comes with Stateful Firewall, DHCP, NAT, VPN, and QoS (WFQ), among other things. I have not setup the QoS yet, but then I've been able to carry on VoIP conversations without issue, even on a well utilized line. The idea was to find a suitable replacement with the PC-based gateway crashing on a semi-regular basis. Instead of getting another PC, I just replaced the whole thing. There are obviously some trade-offs but for my situation, it works well. You said you use RoadRunner so this router wont work for you, but perhaps you have been entertaining a switch to DSL. I use SpeakEasy and have had very few problems with them. If you, or someone else reading this, decide to go the 5861 route get version 6.1.001 firmware. The older firmware isn't that great. You can find the 5861 on eBay for cheap. I bought mine for $13.01. Info on the 5861 can be found somewhere on http://www.efficient.com and the firmware can be found by searching for it on Google. Ryan Wilkins On Oct 12, 2004, at 11:00 AM, Matthew Boehm wrote: I've got a Linksys BEFSR41 at home with RoadRunner service. I'm pretty sure it doesn't do QoS. I'm using WinXP Pro and not sure if it does QoS. I'm using SJ Phone and...(follow the pattern). I have to stop all network traffic on my machine if I want to have any hopes of making a clear call. But I shouldn't need to do that, right? Because somewhere the data packets should be getting queued and my voice packets should be having top priority right? How can I ensure this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS Router/Software Suggestions
Not with an area served by Covad. Speakeasy uses Covad to deliver the DSL service and Covad recently introduced their OneLink service which does NOT require an active phone line for DSL services. What they do is charge $6/mo over the cost of regular service and run a dry pair to your location. I'm in the process of switching to OneLink and dropping my POTS service, in favor of VoIP. On Oct 12, 2004, at 11:38 AM, Matthew Boehm wrote: Switching to DSL would require me to get a phone line, which kinda defeats the purpose of doing VoIP. =) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys PAP2-NA
It's probably something akin to the older H.323 VxWorks based units where the hardware is custom designed, but the software is the same or very similar. No sense reinventing the wheel if someone has code written already and is willing to license the code. Does anyone know anything about the ESS Visba 3 chipsets, or ES3890F in particular? That will be the microcontroller that runs the software. A datasheet on the ES3890F would be useful is anyone has it. Ryan Wilkins On Fri, 24 Sep 2004, Andres wrote: Andres wrote: Nope. The SPA firmware does not load on the PAP2-NA, but the functionality is identical. There are 2 things that I immediately prefer on the PAP2 over the SPA: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys PAP2-NA
This begs the question, again, that someone else posted originally.. what about loading SPA-2000 or PAP2-NA firmware in the PAP2? If it's the same hardware, there shouldn't be any reason not to try it. Ryan Wilkins On Wed, 22 Sep 2004, Brandon Patterson (peering) wrote: This is about big business. No ILEC is going to just sit idle and watch billions in revenue go out the window. It will be interesting to see if port blocking ever becomes an issue. Did I buy my Internet service with out without restrictions? H. Cisco sells to Telco's and Cable guys. Vonage has serious $ so I would expect serious business moves on their part. There is no way the excuse you were given is true. The real reason? Vonage probably went nuts and told Cisco to fix the problem. I know I would, how about the rest of you? There are other providers of hardware starting to gear up for mass production of terminal adapters. It will all happen before the end of the year. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting
On Thu, 16 Sep 2004, Ben Wern wrote: I've already run into some trouble with Broadvoice. Broadvoice support tells me that support isn't really available to BYOD plans, which I suppose I understand given the variety of devices out there. I'm hoping that someone on Asterisk-Users has seen the two issues I'm running into and has a suggestion. They don't officially support Asterisk, but when I've called for support the gentleman asked if I was running Asterisk and then gave me some ideas as to what the problem that I was experiencing was related to. The first issue I'm seeing is that incoming caller id shows the number as out of area and the name shows as 147.135.8.129;bvoice I don't have this problem with other incoming SIP providers -- is there some tweak I need to make Asterisk see CID information from Broadvoice? I've not seen this. While I've not connected up a CID capable phone to my phone adapter, the Asterisk debug output clearly shows the proper CID name and CID number when a call comes in. I'm running Asterisk 1.0_RC2 with a Sipura SPA-2000 as my analog phone adapter. Ryan Wilkins ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oh323 loading error
You might try setting P_PTHREADS=1 in your Makefile. I'm not actually certain if this will work, but it can't hurt anything. Ryan Wilkins On Sun, 15 Aug 2004, Krystian Filiks wrote: I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 load_modules: Loading module chan_oh323.so failed! Can anyone tell me how to fix this, or what that mean? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oh323 loading error
No. Look in the Makefile of the oh323 driver source. Search down through the file for #P_PTHREADS=1 and remove the #. Then recompile. See if that helps your situation any. It may.. or it may not. On Sun, 15 Aug 2004, Krystian Filiks wrote: Would the command make P_PTHREADS=1 opt do the job? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oh323 loading error
Out of my league.. It may.. You can always try it. On Sun, 15 Aug 2004, Krystian Filiks wrote: I think that I found it, I'm compiling PWLIB with ./configure --with-pthreads Do you think this would do it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP -h.323
Yes, it can.. I'm doing it at my home. My current setup is Asterisk-1.0-RC2 using the oh323 driver. I have a SIP connection to Broadvoice talking to Asterisk. I have a e-tel (now Qtelnet) H.323 VoIP telephone adapter as my end point talking to Asterisk. For processing sake, you may want to keep your codec the same all the way through. Originally I ran G.711u on the SIP connection and G.711a on the H.323 connection. It worked just fine but the logs always said something about transcoding between u-law and a-law. I reset the H.323 link to G.711u and now it says nothing about transcoding. In theory you would lose a bit of audio quality in the translation process. In reality I don't really know. email me privately if you want a sample config. Ryan Wilkins On Fri, 13 Aug 2004, Yiannis Costopoulos, Web2Net Solutions Ltd. wrote: is there a definite answer if asterisk can pass calls between SIP and h.323 protocols? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
