similar to: CVS HEAD won't compile for me

I read on the Wiki that Asterisk Realtime requires CVS HEAD, but I've also discovered that not everything on the Wiki is 100% accurate (that's not a knock, but with a program that is changing as fast as Asterisk, it's impossible for the documentation to keep up). Is it true that Realitme requires CVS HEAD? TIA, Jeff Heath

Where are user's voicemail passwords stored and how does the asterisk administrator change them? TIA, Jeff Heath

Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client "subscribes" to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Any ideas are welcome.

For bridging VOIP with PSTN Lines Which one is giving better performance SIP or IAX ? I am looking at a result without NAT in picture ? Can some body give details from experiance ? Also with single SIP/IAX channel can I use more than one call at a time ? Thanks Sandeep

Hi, I'm new to asterisk, and have a uniden UIP 200 that I got off of ebay. I'm having trouble getting the phone to register with asterisk. I've tried a few different settings. I'd be extremely grateful if someone with a similar setting could give me the sip.conf block for the UIP and the settings you're using in uniden.txt. Here's what I have currently: IP of phone

We're proud to announce Astricon 2004 - the first Asterisk user's and developer's conference! * Where? Atlanta, USA * When? September 22-24, 2004 The conference is arranged in partnership with Digium.inc and the keynote speaker is Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara

Friends, Please send all offers for help *off list* to us at info@astricon.net. Do not disturb the list with offers of your services, please. I repeat: Only the Developer's Meeting will be considered for broadcast at this time. In order to enjoy the conference, you will simply have to be there. It's an IRL experience - meeting all the other Asterisk user's from around the globe,

Astricon 2004 is the first Asterisk user's and developer's conference, to be held in Atlanta, Georgia in September. See http://www.astricon.net ** We will soon open for early bird registrations! ** To get a very low price, I recommend that you participate as a speaker. In fact, speakers doesn't pay any fee at all to participate in Astricon. **** Send us your speaker's proposal

I feel this is a slap in the face for those of us that have been here and I don't feel I should HAVE to pay to be certified... I think me and MANY others are about to walk out of the project over this. I have already spoken with many people that are close to the project. You're hurting US and our ability to make money. I still know the code better than most of the people that will be

We're now opening up registrations for the Astricon tutorials again. We've been able to move to new conference rooms within the same hotel. Register on line at http://www.astricon.net We're sorry for the inconvienience our recent closing of the tutorials may have caused you. You are welcome to contact us at info@astricon.net to change your reservation to include the tutorials. There

I just built libvirt 0.9.3 on my Ubuntu 10.10 box. It appears to have built correctly. When I try "virsh -c qemu:///session list --all" I get the following: error: Failed to connect socket to '@/home/heath/.libvirt/libvirt-sock': Connection refused error: failed to connect to the hypervisor After that, libvirtd isn't running as user heath as it did when I was using

Well I'm stunned no one has suggested a webcast option. I mean we aren't talking a bunch of people unable to grasp the concepts of chat/voice/vision sessions with a log in/remote display capability. If you think this is an option let me know I have someone who has some software they wouldn't mind stress testing as a trial. Cheers, Dean > -----Original Message----- > From:

Thank you very much for all feedback on Asterisk Sunday News! This is the last issue for June. This week I'll go on holiday and will be back with more news in early July. My kids are getting summer leave this week and we'll be visiting the south of England for a while. Another part of Europe that still use their own currency. If you think there's an European standard, you're

Dear All, I am totally new in this arena and I am still waiting for my installation process on freebsd to finish, but I wanted to make sure of the following: - Call routing between IP telephones can be done regardless of who made the phones? - Asterisk does support SIP V2? - it does act as SIP Proxy and Register? -- Thx MAG -------------- next part -------------- An HTML attachment was

Hi all, I'm trying to see if I can get the hint priority working with my polycom 500. So far I have 2 </reg> entries with the same sip registration, one is labeled as private, the other as shared. I have set the hint priority before anything else in my dialplan for my extensions. As it stands, I have two registrations on the phone, one has a half greyed out phone icon, the other

I notice these notices running cvs head: NOTICE[6154] chan_sip.c: No field 'From' present to copy Any ideal what make this notice? -- respectfully, Joseph =============== ---------------------= ********** =

Hello Asterisk users around the world! Recently, I have been working with pretty large Asterisk installations. 300 servers running Asterisk and Kamailio (OpenSER). Replacing large Nortel systems with just a few tiny boxes and other interesting solutions. Testing has been a large part of these projects. How much can we put into one Asterisk box? Calls per euro invested matters. So far,

I have a large list of matrices and a vector that identifies the desired matrices that I would like to rbind. However, I am stuck on how to get this to work. I have written some code below to illustrate my problem: # 3 simple matrices a<-matrix(1:9,3,3) b<-matrix(10:18,3,3) c<-matrix(19:27,3,3) #this is the type of list of matrices I am dealing with

I have been working with this for a wile and I have been watching the list for about a month on this subject, to no avail. I am wondering if anyone has successfully configured asterisk for clients to connect to it when the clients are behind nat. I mean successfully because I can do everything except for audio, my audio is only one way. I am asking so I can determin if I will be continuing

Dear Asterisk Community, Asterisk 1.0 was released at Astricon 2004, in September last year. It's been almost a year and we haven't been able to go ahead and release a new version. Now is the time to try to move forward again. As we've outlined before, the process is this: -------------------------------------------------------------------- * Code freeze: At this point, we'll