[Asterisk-Users] ipvolution t1 cards
Has any one used the Ipvolution tdm120 cards i am intrested to know how well it works and how well the on board dsp's work. ___ --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] CVS-HEAD Compile Problem
I ran into the same problem the other day and just went back to non head version It would be nice to figure out why it does this. - Original Message - From: Nico Giefing To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, August 19, 2005 9:20 AM Subject: [Asterisk-Users] CVS-HEAD Compile Problem I have a little Problem, I will compile asterisk CVS-HEAD but after 20 second of compiling i get the message as shown at http://pastebin.com/340654 about 1000 times. Do anybody know a solution for this? Thanks a lot Nico ___ 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 7750
- Original Message - From: Mark Johnson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, June 21, 2005 8:56 AM Subject: [Asterisk-Users] Cisco 7750 I have read of people attempting to do this, and I just wanted everyone to know about what we've discovered about the Cisco 7750. If you don't know what it is, it's basically a blade server. I have 1 power blade, 1 alarm processor, 2 system processing engines and 1 multi-service route processor. We just got asterisk running on this today!!! Just dont let cisco know We haven't tested the T1 with it, yet, but I pretty sure it will work OK. All of the FX ports work beautifully right now. The big deal about this for me is that I have battled over and over again with interrupt issues with Digium hardware. This is sweet because all the T1 processing including echo cancellation should be done on the route processor. Asterisk doesn't have to do much of anything. so im guessing that all of the t1/fx ports are configured in the system processor and just talk sip/mgcp to the route proccessor. That sounds like a pretty sweet setup If you could only get cisco to sell you the hardware without having to buy the software. ___ 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] RJ45 instead RJ11 in Digium's TDM20B card help meplease
just plug it into the rj45 - Original Message - From: Kumara Jayaweera [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, June 14, 2005 12:46 PM Subject: [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help meplease Dear all, I am happy to tell you that I received a Digium's TDM20B card for my Asterisk box today. but the phone jack is RJ45 instead of RJ11. So Please I need precise instructions to connect a phone to this card. please, assume that I have a phone (a normal analouge phone connected to the one end of a cable with an RJ11 jack (at the phone side). and now I want to connect the other end to the Digium's TDM20B card. what is the wire combination/sequence for a successful installation? please define the pin no.s accurately. You advice/support is highly appriciated. Thanking you Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 multiplexer (or ?) for failover in largeinstallation
It sounds like your looking for a t1 protection switch with will do what you want It can switch t1's for failover or loss of signal on a t1. These are usualy rather expensive and Might work 100% in your example because they will only switch over if th actuall t1 goes down. So If your server dies/locks up and doesnt tell the t1 to go into alarm it will still think it is up and not switch to the other. - Original Message - From: Mike [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, June 13, 2005 10:35 AM Subject: [Asterisk-Users] T1 multiplexer (or ?) for failover in largeinstallation Hi, Please forgive my terminology, still a bit new to T1s and such. I'm looking for a way to have 5 T1s from a carrier terminate into some type of box (multiplexer?), then be able to plug 7 asterisk servers into that box (each with single port T1 card) and be able to have 2 * servers go down at any given time and not actually have the carrier see that anything has happened. Obviously if a * server crashes the calls on it at the time will drop, but then once the box (multiplexer?) sees that a T1 is down (between the box and asterisk) it will terminate those DS0's on another T1. Basically some type of hunting/pooling/load balancing. Anyone heard of anything like this? Or am I off my rocker? Thanks, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voiceblue gsm/sip
does anyone know of any vendors that sell the voiceblue gsm gateways in the US and have any experience with any providers who it will work with I know that cingular and ATT have gsm networks but there are some linke sprint that are mostly cdma I am wanting to use some of these for some remote sitesand wondering if any one has had any experience with them in theUS. Thanks Trey Scarborough ___ 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] livevoip
If your doing that many mins a month you could probably go to one of the bigger cariers yourself Level3, global crossing, broadwing, ATT, williams to just name a few - Original Message - From: VOIP Consultant [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 09, 2005 11:49 PM Subject: RE: [Asterisk-Users] livevoip They have been down for 3 days now. Althought I prize their concept, perhaps I am wrong in thinking they are close to the only option. Will anyone know of another carrier that can provide me with (for about 500,000 minsxmonth so far): 1. on the fly dids for $1 2. IAX 3. on the fly 800 setup 4. Level(3) -- Level(3) -- Level(3), I can't stress enough how important a consistent carrier is 5. all incoming/outgoind on the 1.2 range 6. Excellent international long dist rates and quality Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kellner, Peter Sent: Monday, May 09, 2005 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] livevoip Yes. I have never had any downtime that I know of but the quality is consistently below average. Because they are iax, it is easy to set up. If you are interested, I'd buy there minimum ($30 I think) and test it out and see what you think. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JD Austin Sent: Monday, May 09, 2005 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] livevoip Anyone use livevoip? opinions? -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Install Asterisk on CCM MCS-7835 Server
I am almost positive that these are just HP/Compaq servers - Original Message - From: Walid Azab [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, April 28, 2005 3:07 PM Subject: [Asterisk-Users] Install Asterisk on CCM MCS-7835 Server Hi All, I am replacing Cisco Call Manager with Asterisk. As you know CCM is on a MCS 7835 Server which comes with a custom version of Windows. Does any one know how to install Linux on that H/W. My guess is that someone must have tried the same thing before. I know how to install Linux however I cannot get passed the H/W limitation. Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Number Portability and VoicePulse
you can always move your phone number it is a federal regulation in most areas that all providers all for number portablility - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 1:32 PM Subject: Re: [Asterisk-Users] Number Portability and VoicePulse Sounds like good questions to ask VoicePulse. - Original Message - From: James W. Brinkerhoff [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 1:14 PM Subject: [Asterisk-Users] Number Portability and VoicePulse I have two questions regarding number portability... 1) If I port a DID over to Voicepulse, can I then move that DID elsewhere somewhere down the road. Or does voicepulse now OWN that DID? 2) Can I take a DID assigned by Voicepulse and transfer it to someone else? If not, why? -jwb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Network Sniffing Calls for recording
yes it can be done easily take a look at this little program http://vomit.xtdnet.nl/ vomit - voice over misconfigured internet telephones - Original Message - From: Steve [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 07, 2004 7:54 PM Subject: Re: [Asterisk-Users] Network Sniffing Calls for recording -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 07 June 2004 06:29 pm, Chris Albertson wrote: Many people use ethereal to capture network packets. I've used it to debug SIP sessions. www.ethereal.com/ In theory one could re-contruct a phone converstion from logged packets but it might take some effort and you'd need to be pretty smart to find the packets from a call from Joe early last week in the morning some time. Not really hard. If you have his IP. Of course the first problem is that unless you are recording all the time you cannot go back to some point in the past. However, if you did record traffic you can filter it on f.ex port 4359. You could see how much traffic you have at any point in time. The latest version even let's you graph it live. You could also run tcpdump to a file and review it later, but ethereal is much more powerful, and with the filtering abilities you can be as specific as you want. As far as following a conversation it can also follow a network session. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFAxQ4yljK16xgETzkRAjSTAKC1LAhVUxyv3KX4CSBGoYFhiUVgaQCgvyKa 2gHfG55Jx/IVTc6B3K9bNfE= =v/EL -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ipphone voicemail problems
Im having a little problem with voicemail and my cisco phones i was wondering if anyone might have seen this before and let me know whats going on. it spits this out and then my cisco ip phone reboots im using the latest cvs and a cisco 7910 phone WARNING[1234379840]: File res_adsi.c, Line 205 (__adsi_transmit_messages): Unable to send CAS -- Playing 'vm-login' (language 'en') Skinny [EMAIL PROTECTED] went on hook WARNING[1234379840]: File app_voicemail.c, Line 1907 (vm_execmain): Couldn't read username == Spawn extension (internal, 850, 1) exited non-zero on 'Skinny/[EMAIL PROTECTED]' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
