similar to: What happened to my subscription?

I am testing 1.4 branch on OSX (10.4.8) and although it's running and passing calls ok, I am still not able to connect using asterisk -r. When I do open a CLI using asterisk -r, it appears to start up normally, but then is non responsive to commands (exit works though?). I am currently running SVN-branch-1.4-r46716. Any ideas on why this might be, or how to figure out how to fix it?

Probably just me being dumb, but I am trying to update my asterisk to the latest (1.2.5) When I go to my /usr/src/asterisk I type: make upgrade make install This seems to be doing it's thing, but when I type show version from the console (after a restart) I still get: Asterisk SVN-branch-1.2-r7231 built by root @ notdeadyet-imac.local on a Power Macintosh running Darwin on 2006-03-04

Martin: I had seen your other post and sent you a message off-list, but I never got a response. What do you feel is the most lacking that does not make it ready for a production enviroment. - I've been using a SIP deskphone in my office and usually some sort of ATA at my house, both as the primary phone. I've also had mobile phones from almost every carrier. Each one of these devices

Hi, I have a spiffy new gateway which seems quite promising. It's the Audiocodes MP114 FXS_FXO (2 of each). I have got it configured and working reasonably well, but have a couple of issues. 1) Asterisk 1.2.13 voicemail seems to be hung up on by the gateway after 10 seconds. This isn't asterisk saying it's quiet for 10 seconds, it's the gateway deciding it's time to go

YUP, this is the way that asterisk works. It is going to quelch all DTMF that goes out via a SIP gateway via asterisk. I spent a long time working this through and it has to do with the way that asterisk deals with DTMF and the DSP.c module that sits inband to the RTP/audio stream. There is a flag called DSP_DIGITMODE_NOQUELCH that is broken that might allow the inband DTMF after answer to work

Hi Friends, I have successfully implemented Intercom, Voicemail and International dialing using Asterisk. Now I want to connect my PSTN Lines to Asterisk server. I have 3 PSTN number (lines) to connect to Asterisk. For this, I want to use Sipura SPA-2100. Is my decession is correct or not? Is there any disadvantages with this Sipura SPA-2100? Please tell me. Thank you. Regards, Chandramouli

Ok, I am back from my thanksgiving holiday, and I find there was a big snow storm here in Seattle. Apparently during the storm there where multiple brown out/black outs. I have struggled since day one to get a high quality PSTN gateway configured with my very long loop and Mac based asterisk. I originally tried the HT-488, which had multiple issues, and was unacceptable. I then purchased

Hi, What does this mean when I have something like: > qda.boot <- boot(train, qda.bootstrap, R = 500) Error in qda.default(structure(data.matrix(x), class = "matrix"), ...) : Rank deficiency in group M with my qda.bootstrap() looks something like: > qda.bootstrap <- function(data, index) { + boot.qda <- qda(x = data[index, 2:9], group = data[index, 1]) + qda.pred

I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a

Hi, i am trying to find an appopriate R function which will estimate the bias associated with the apparent error rate of my logistic discriminant model (groups = 2, covariates = 3). I have read that bootstrapping can be used for this. Does anyone have any ideas on how I can go about doing this? I have quite a small sample size for one of my groups and I realize this may cause some problems, but

Hi, i am trying to find an appopriate R function which will estimate the bias associated with the apparent error rate of my logistic discriminant model (groups = 2, covariates = 3). I have read that bootstrapping can be used for this. Does anyone have any ideas on how I can go about doing this? I have quite a small sample size for one of my groups and I realize this may cause some problems, but

Is there a way i can change my subcription email address, without unsbubbing and resubbing myself? Cheers, Mathew -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20030221/c9149135/attachment.html>

Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want

I did a quick check between what I have set and your settings below. You can try the following and see if it helps In your endpoint: bind_rtp_to_media_address=yes With best regards Florian Floimair Innovation - Software-Development - VoIP & DevOps COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstra?e 51 Tel: +43-662-85 62 25 Fax: +43-662-85 62 26 http://www.commend.com Security

Asterisk Project Security Advisory - AST-2014-004 Product Asterisk Summary Remote Crash Vulnerability in PJSIP Channel Driver Subscription Handling Nature of Advisory Denial of Service Susceptibility Remote

Asterisk Project Security Advisory - AST-2014-009 Product Asterisk Summary Remote crash based on malformed SIP subscription requests Nature of Advisory Remotely triggered crash of Asterisk Susceptibility Remote

Asterisk Project Security Advisory - AST-2014-004 Product Asterisk Summary Remote Crash Vulnerability in PJSIP Channel Driver Subscription Handling Nature of Advisory Denial of Service Susceptibility Remote

Asterisk Project Security Advisory - AST-2014-009 Product Asterisk Summary Remote crash based on malformed SIP subscription requests Nature of Advisory Remotely triggered crash of Asterisk Susceptibility Remote

http://grandstream.com/BETATEST/HT488_496_386/

Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). Anyhow, I can see that the gateway is passing