similar to: Voicemail: Timestamp suddenly reverted to GMT!!

Asterisk Users: I've been working hard on app_voicemail2 which is an enhanced scalability version of app_voicemail. Specifically, its features are: * Highly improved internal architecture (maybe someone else can actually code on it) * Foot print for getting mailboxes from DB (for Vonage) * Segmentable mailboxes, allowing you to truly multihost voicemail for multiple companies

I'm sure there must be a setting I'm missing somewhere, so I thought I might was well ask here. Conversations are punctuated by sudden replacement of a given syllable or so of conversation with a DTMF tone. I would hope perhaps there's some kind of setting that has to do with the way it detects inband DTMF? I'm pretty sure it's an artifact of this particular ATA; my

I wonder if anyone out there has a perspective on this. There are a welter of tickets out there on the matter, most of them closed. This problem began for me over a year ago, and continues up to the latest versions I've installed (1.6.2.13). It happens randomly, and the suggestion on one of the bug tracker tickets that it is instigated by a small network leg looks to be on point to me,

I have /var/spool/asterisk/voicemail NFS mounted from another server. Everything is fine, until I simulate an NFS server failure, by shutting down the NFS server process. At this point, Asterisk becomes almost non-responsive. It won't even process a 'sip show peers' command correctly. It displays a few lines of text, pauses for several seconds, and then displays the rest. When a call

Wondering, since they appear to be plentiful on eBay, whether I could get a Nortel 350 to use to learn my way around ADSI. The vendor claims that these are "generic," and looking through the archives I wonder if that means that they might be unlocked in the sense that the word is meaningful to asterisk. Of course I am green as could be on this topic, so this question may even be a

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

I am wondering if there are any tricks getting the Cisco ATAs to do "distinctive rings" via the ALERT_INFO variable? I have seen some contradictory information in the Wiki, and I tried the example there. I then sniffed the connection between the server and the ATA and didn't see the header sent like it is "supposed" to be. If someone out there has a handle on this and

Hi all Looking for a solution to create a flexible voicemail solution in Asterisk without the need to preregister the voicemail users (via databases etc etc). Scenario: All incoming calls are voicemail calls however the dialled number (called party) does not necessarily have a voicemailbox configured in the Asterisk system. I am looking for * to do the following: * Call comes in *

Hello all, I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to send the voicemails as attachment to e-mails and delete the voicemails from my PBX once it has been sent. But, I don't have a running MTA here even on the PBX itself. I just want to send the e-mails to my GMail account from my PBX. Can I just use the mail or mailx command to send the e-mail and attach the

Folks, I'm trying to get Asterisk to load my voicemail configuration from MySQL. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database I restarted Asterisk, but no luck: the voicemail.conf does not get updated. I started with a sample voicemail.conf that I found on the Wiki. Or was it from Voicepulse? I can't remember. For initial testing, I

I'm having tons of trouble getting the fromstring to work in voicemail.conf. I've tried both voicemail and voicemail2 but the emails still seem to be coming from asterisk pbx. Has anyone had any luck with this? ================= Here's my voicemail.conf: ; ; Voicemail Configuration ; [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=wav49|gsm|wav ;

I see a lot of chatter in the archives about intercom and paging, but has anyone addressed zone paging? Each of the 50 telephones in a large clinic would be members of one or more paging zones. Someone could then page Dr. X in zone #1. Would this be possible with analog phones? SIP? Thanks, Mike

Hello As far as ive understood, you can just write Exten => s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and "${AVAILSTATUS}" would return "1" Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Brian Capouch Sendt: 27. juni 2006 09:10 Til:

Could someone provide some steps for troubleshooting Realtime? I can't see any signs that it's working. I followed and double-checked a few different guides around the net, but haven't been able to figure it out. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I wonder if someone out there could loan me a peek at their sip.conf? I have conflicting advice, for instance, about whether or not to use "nat=1" and also whether or not the ATA should be registering with the instance of asterisk it is going to be using to dial out. Thanks in advance. B.

I wonder if anyone could send me a pointer to technical specs and pricing information. I got a mail today from an acquaintence that contains what I believe is some serious misinformation, referring to the 7960 as their new portable 802.11b SIP phone. A quick search of eBay would seem to refute that. I hope this is an OK question to ask. . . Thx. b.

The Vonage service is offered with a SIP Cisco ATA device for connection to an analog phone. Is it possible to connect the Vonage service directly to the Asterisk PBX bypassing the ATA and FXO card? Are there other services that offer this capability or something similar to IP dialtone? Thanks Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am using Audacity to record some voice prompts. The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. I know that some of this comes with the territory, but I wonder if there is anyone out there who does this routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting

I got a call from an ISP friend tonight who said he is getting calls from people who are getting signed up with Vonage. Instead of sending them ATA186s, apparently they're receiving something made by Motorola. They apparently work significantly differently than the Cisco units, and there have been some problems. Anybody know anything further? Thx. B.