similar to: Enterprise users of asterisk?

Displaying 20 results from an estimated 3000 matches similar to: "Enterprise users of asterisk?"

2006 Jan 06
2
SPA-3000 is translating vocal sounds into DTMF
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
2010 Oct 30
2
Exceptionally long queue length queuing . . . .
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,
2003 Jul 27
3
Nortel 350
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
2006 May 29
4
app_conference DTMFs?
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
2006 Jun 16
17
Voicemail with NFS
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
2006 Jun 26
4
Oh oh. Micro$oft just noticed VoIP
It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.
2006 Jun 27
2
SV: Error in config sample for GoToIf?
Hello As far as ive understood, you can just write Exten => s,n,GotoIf([${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:
2003 May 09
2
Configuration for ATA186 behind a NAT?
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.
2003 Jul 22
2
Cisco 802.11b VoIP phone?
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.
2003 Sep 05
2
VONAGE or IP Dialtone
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:
2003 Oct 06
2
Anyone else use Audacity for prompts?
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
2003 Dec 07
1
Vonage sending Motorola gear now?
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.
2008 Jul 16
2
New transactional enterprise e-commerce web site
We are close to releasing a high-volume, multi-lingual, e-commerce site in Ruby on Rails that has previously been written in an alternative Java framework (e.g. servlets and JSP). This is a real-time 24/7 betting and gaming web site based out of Gibraltar in Europe. When we embarked on this project, we really wanted to find examples of other high-volume web sites that had adopted RoR. We did
2005 Sep 06
4
Working example of ALERT_INFO with Cisco ATAs?
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
2007 Mar 01
5
Asterisk Realtime
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:
2004 Jan 17
6
Zone Paging
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
2005 Mar 18
15
Meetme2 compilation problem
Hi All, I am trying to compile meetme2 in my asterisk box and getting the following compilaton error. Please help me to sort it out. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function)
2007 Apr 06
2
regular expression
Dear R-List, I have a great many files in a directory and I would like to replace in every file the character " by the character ' and in the same time, I have to change ' by '' (i.e. the character ' twice and not the unique character ") when the character ' is embodied in "....." So, "....." becomes '.....' and
2006 Apr 07
5
[OT] Centrex Question
I haven't dealt with Centrex for a long time, and one of my customers is being courted heavily by a Sprint salesperson. Am I not correct in assuming that each "line" of Centrex corresponds to an "extension" in the PBX world? This site has 2 POTS lines and 5 extensions, and they told me that for the same thing they're paying right now (~$40/POTS line) they will be
2003 Oct 20
16
A software FAX modem
Hi all, I would like to announce the availability of an initial test version of a totally software FAX facility, suitable for use with Asterisk. This is a first public test release, so don't expect a solid polished product just yet. People have shown interest in what I am doing, and here is the evidence that it is not vapourware. If the notion of a software FAX machine is new to you,