similar to: "Slash Tone" at pstn cut-though?

Displaying 20 results from an estimated 10000 matches similar to: ""Slash Tone" at pstn cut-though?"

2009 Aug 18
1
Play Fake ring in phpagi
> I'm going blind searching - maybe you know? > > During the execution of a script I want to play fake ring to caller. > Both of these examples complain of missing option: > > $agi->exec("Ringing"); > $agi->exec("Playtones ring"); > > Notice: Undefined variable: options in > /var/lib/asterisk/agi-bin/includes/phpagi.php on line 326
2009 Sep 20
2
different verbose level for full log than to console?
Is it possible to have a different verbose level full log than to console output? I'd like to keep console verbose at 1, but now full log is at 1 also. Bart -------------- next part -------------- A non-text attachment was scrubbed... Name: bhfisher.vcf Type: text/x-vcard Size: 253 bytes Desc: not available Url :
2006 Jun 03
2
ADIT 600 <=> Asterisk Help
I've been reading the Google searches trying to understand how to tie together Adit 600 to Asterisk to provide 2 way service. I'm about blind from reading. I assume, the answer is using MGCP between the boxes. However, the examples I found don't really explain fully enough to know how to modify examples to work for me. I'll have in the ADIT with T1's. There is a CMG and
2007 Mar 23
1
Noob question regarding PCI 2.x & TDM400P Card
I have some old PC's I want to build as a test box - It's up and running OK now. Now I installed a TDM400P and there is nothing I can do to get the card to come up. My guess is the box is not PCI 2.2 compliant or does it need to be to see the card? Thanks, Bart Here's what I know: Processors 1 Model Pentium III (Katmai) CPU Speed 551.37 MHz Cache Size 512 KB System Bogomips
2006 May 23
4
What about T400 T1 cards?
Can anyone clue me in about these T400 T1 cards I see advertised? I hear they are Digium Clones. Is there some reason to avoid these? How do they compare to TE410P's for example. Bart
2007 Sep 03
1
ADIT 600 & CMG <=> Asterisk question
I've searched but can't find an answer as to how many MGCP paths can a single ADIT/CMG card support? It appears it's only 24 ports, maybe 48. What I'd like to do is install 6 Telco T1's into a single (or more) Adit 600 and route inbound calls towards asterisk. Can I have more than one CMG in a single chassis? Or maybe you know of a better way to connect T1's to
2006 Jun 28
0
Dial Tone + E&M
Maybe one of you can help me with this: We have T1's that come from both MCI and Global Crossing as uses channelized (24 Ports per T) with inband (DTMF) ANI and DNIS delivery (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk Server. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail
2002 Mar 20
2
Excludes not working
Hello, I'm a relative newbie to rsync, I use it to backup (mirror) a bunch of Windows boxes. I mount them using samba (to avoid having to install the cygwin version on each box) then run (using rsync 2.5.2 run from crontab) rsync -vuaz --delete-excluded --exclude-from=/root/bin/rsync.exclude \ --modify-window=2 /mnt/pc/machine/share /backup/machine/share It works like a charm, except
2008 Dec 21
3
A method to determine PSTN Call Provider?
I'm looking for a solution to determine if a PSTN call to a zaptel channel was originated from a VoIP provider or not in real time. I'd like to use the callerid(num) to reverse match to the provider. Does anyone have a clue how I could do this? TIA Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:
2009 Oct 10
3
Method to use SOX inside a Dialplan
I'm trying create a feature that allows a callers to add more speech to his recording. I think this can be done inside a dialplan, but I can't find an example of how to do this. Basically,after he records the primary message, a menu would play asking if he wants to append to this message. If yes, then he would record a temp file with the additional message and when done, I want SOX to
2007 Sep 07
1
Channels in use?
I'm using version 1.2 and need a method to detect the number of channels in use from inside the dial plan. I'd like to count total channels system-wide, but even better if I can determine for a selected extension also. I've searched the wiki, and don't see such a function that does this. Any ideas? Bart -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410
2009 Oct 01
1
DTMF problems during a message play
I'm using the latest asterisk-1.4.26.2 and no zaptel trunks used, all SIP. I have one user that is having problems once he connects to asterisk. He's dialing from his home phone (pstn) to a Vitelity DID (SIP Trunk) which goes to my asterisk IVR. If he presses a dtmf during any message, the press is ignored unless the press was a #, 0 or *. Otherwise, he needs to wait for the
2009 Oct 05
2
Method to downgrade asterisk
I currently have asterisk-1.4.26.2 installed and working. It was sugguested I try asterisk-1.4.25 to see if it fixes my SIP dtmf problems. What is the method to downgrade? Do I just do in the asterisk-1.4.25 folder: make clean ./configure make install Or do I need to 'make clean' in the asterisk-1.4.26.2 first then move to the asterisk-1.4.25 folder and do ./configure & make
2008 Oct 02
1
DTMF
How can I know for sure if SIP Trunk Provider is sending DTMF 'inband' or 'rfc2833'? And more importantly if they could be sending both? If I specify 'inband' should they honor that? Thanks, Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081002/3b34d38d/attachment.htm
2008 May 22
1
Telco intercept prompts
Does anyone have all the Telco intercept prompts (numbers and such) with voice inflections to simulate number referrals and disconnects I could download? TIA, Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080522/08b001ce/attachment.htm
2005 Jul 15
1
SYMBOL NETVISION II NP-3010
I was looking at these SYMBOL NETVISION II NP-3010 VoIP TCP/IP WIRELESS PHONES - I know they have been discontinued. Am I asking for trouble to buy some of these for use on Asterisk? TIA Bart
2006 May 18
0
E&M and Dial tone
I'm a bit confused about how to handle this. I have Asterisk sitting in the middle between a Qwest Long Distance T1 (Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic D/240SC-T1 card. The Qwest T1 originally was connected to the Dialogic card directly. The signaling was set to E&M Wink Start because Dialogic used this as its default settings, so it just worked
2009 Sep 16
1
ACR Anonymous Call Rejection
Does any have or can point me to /ACR/ Anonymous Call Rejection message I can download? The one I found was not not too clear. Thanks, Bart -------------- next part -------------- A non-text attachment was scrubbed... Name: bhfisher.vcf Type: text/x-vcard Size: 253 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090916/c3b682d5/attachment.vcf
2009 Oct 07
1
DTMF Issues
I have a block of DID's that I ported to Vitelity about 7 days ago. The problem is if a POTS caller dials into the system, his dtmf is not heard at READ() or Background() while a prompt is played. After the prompt is finished, then dtmf is heard. I've been working with their support, but it still not resolved. SIP callers are not effected. Yesterday, I purchased a DID from
2006 Jun 17
0
E&M + Dial tone
Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system correctly