search for: digum

Displaying 20 results from an estimated 43 matches for "digum".

Did you mean: digium
2004 Feb 03
2
Pictures of new multiport FXO/FXS from digum
...Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of woody+asterisk@solutionsfirst.com.au Sent: Monday, February 02, 2004 11:06 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > thisemailaddressisbogus@risehigh.com > Sent: Saturday, 31 January 2004 8:56 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-U...
2004 Mar 31
5
3-4 port FXO card recommendations
...tem (Nortel Venture). I presently have three POTS lines. I would use a VOIP provider, but now are presently available in the Toronto, ON, CANADA area that support user owned hardware/software. I need a 416/647 area code number. In looking at FXO cards to atteach to the lines, I've seen the Digum single port FXO card and their multipoort FXS cards. So far, the cheapest route seems to be to buy three Digum FXO cards; however, the system I want to use them it only has some three/four PCI slots. Having one card with three FXO ports would be useful. I'm also open to using an external de...
2011 Mar 24
4
Issues with Digum Repos / AsteriskNOW & Bad Packages
I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error
2005 Aug 17
0
Xten & Digum TDP FXO card: No sound
I have a tdm 3xfxs and 1xfxo, aslo I have a setting with 1 snom 190 and 2 xten line. I can call from the snom to the ptsn line at the fxo port ok. I can call from the ptsn to the xten lite phone. I can call from the xten lite to snom but what I CAN`T do is; Call from xten to ptsn. When I dial from the xten, I can hear the dialed party, but he cannot hear me... Tips? Help? What I'm
2008 Apr 05
2
IAX IP Phone
Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost.
2005 Feb 24
1
Which Codec(s) to use..?
Hey Everyone, I am playing around with my * box, and I have a few different phones hanging off it it right now. I have a Cisco 7960 capable of g729, ulaw and alaw, I have a Cisco ATA186 with a Panasonic cordless phone attached to it, I have a Digum IAXy with a dumb analog phone attached to it, and I have a Linksys PAP2-NA with an AT&T 959 analog phone attached to it. I also have several IAX2 connections, one to NuFone and one to another provider. My question revolves around which codec to use. I purchased 10 licenses of the g729 from D...
2005 Jul 25
3
Zap channel configuration problem
Hi, I would like to use a digum card to call an external number through my PSTN. I think that I have a problem in the configuration. Asterisk returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' I use Fedora core 3. I installed libpri, zaptel and asterisk. I plugged my line on the FXS module (gre...
2012 Apr 11
4
Dahdi-2.4.0+2.4.0 means ??
Hi, can anyone tell me what does that 2.4.0+2.4.0 means in dahdi release numbering ??? 2.4.0????? regards upendra. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120411/9b160392/attachment.htm>
2009 Mar 06
2
question about MeetMe performance.
hello, I will do a server to do a lots of conferences (MeetMe). I want to know that if I dont use a digum card, the limit of simultaneous calls is harder without a card than with a card ?if, yes, how harder is the limit? thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML atta...
2005 Sep 08
2
Transfer calls from cellphone
...ature a) with *, but are b and c possible? Also, I would probably be using a channelbank to reuse old analog telephones and connect analog fax-machines to the PBX. I.e. PSTN | PRI | TDM Card | | | Channel Bank | Asterisk | LAN How much do I need to worry about echo? Should I consider the Digum TE411P for the extra $1000? Arnar
2006 Nov 21
2
Answer Machine Detection
...text but i can't get it to work, just on inbound context like whe i use the application Answer before AMD, but i need to make AMD to do the detection on an outbound predictive dialer integration. Follow are the inbound and outbound examples. My current environment is Asterisk 1.4beta3 and a Digum TE105P with ISDN E1. Have any one managed to do answer machine detection already? [outbound] exten => _x.,1,AMD exten => _x.,2,Dial(SIP/${EXTEN}@10.1.1.203,,tT) exten => _x.,3,Wait(2) exten => _x.,4,Set(RECORDEDFILE=${CALLERID(num)}.wav) exten => _x.,5,Record(${RECORDEDFILE},,,skip...
2009 Apr 24
3
timing source problem
...ble. So - here the questions... - is it possible to do what i want to do ? - do you think timing=0 in zaptel.conf will work ? - would it be possible to connect a xorcom 2 PRI channel bank to asterisk to handle the qsig line between the two ? Or will the xorcom then also take the timing from the digum cards - telco lines ? any hints would be nice... many thanks Wolfgang
2005 Feb 02
4
new install
hi, i got an error while running the asterisk -v error message: error while writing audio data ===== R.B.Roa Traffic Management Engineer PhilCom Corporation __________________________________ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail
2007 Jan 28
1
T1 Wire Level Tapping
I am trying to do a wire level tap on T1 equipment using digum equipment. So far most call monitoring hardware for call centers try to stay on the analog side requiring a lot of rewiring. I have already posted to the list about T1 "bridging" using DAC's support in the zaptel drivers. I still don't know if I can spy on channel information s...
2006 Feb 09
1
Static problems with Asterisk + Polycom phones
Hey all, I'm having problems where there is significant static when making SIP -> PSTN calls. SIP -> SIP and SIP -> VM calls are totally clear and fine. Here's the setup: Polycom 601,501, and ten 301s. Digum 2400 TDM card w/echo cancelling, 12 FXO ports. The TDM card is on IRQ 5 with nothing else on it. Server Specs: Asus P4P800E Deluxe P4 3.0 Ghz 1 GB Ram 80 GB SATA HD - There is no static when using a normal phone direct to the 66 block. - The sound is also a bit low, and bumping the volume on the...
2003 Sep 05
1
T1 - A little guidance needed to get started, What order to do zaptel, zapata...
...onencted to the Fujitsu PBX that I built with mgetty/pppd and have the lines provisioned the same way as those dial-up server, ESF, B8ZS, and E&M wink start, so I have confidence in the guys who set up the PBX. I've built a loop back plug for T1 and looped it both directions, I can get the Digum T100P card to go green as well as the PBX port. So I have confidence in the wiring. I don't understand the relation ship between zaptel and zapata and wether I need to config a dail plan or anything else in asterisk before I get the T1 up. So do I start with /etc/zaptel.conf and the zaptel mod...
2005 Jul 05
1
(no subject)
...to have no effect. 3. Ensure your not getting any NMI's. Which I did, and I was not accumulating NMI"s.. 4. Run"hdparm -t /dev/[Hard Drive Device]" and notice if you hear crackles, pops, or loss of audio. I did get substantial interference with the audio when running this. Digum had me run the zttest application, which showed every couple of interations I would get a reading of 99.3 or 99.4 where they say anything below a 99.89 is bad. I can fax from FXS to FXS on the same card just fine. Which I think is an interesting point. Maybe it has to do with data passing across...
2003 Sep 26
2
Creating a SIP gateway for use behind NAT
...ster with FWD? 2) Am I basically going to be using a blank configuration on the GW box, and having the second * box with all the fancy dial plan stuff, or the other way around? 3) Am I looking at using the switch => command? If so, an example would be fantastic, I've looked around on the digum list with google, but can't seem to find anything, perhaps I'm just using the wrong search words? Also, does my logic seem to make sense? Again, I would have just loved to put the TDM400P card in the GW machine, but unfortunately the dd line to make the CPU go to 100% just isn't ve...
2005 Feb 09
2
Problem with meetMe
I try to use meetme app after reading manual i compile and install zaptel with ztdummy when i make lsmod i have ztdummy 2532 0 (unused) wcusb 20064 0 (unused) zaptel 179168 4 [ztdummy wcusb] usb-uhci 26348 0 [ztdummy] usbcore 51616 0 [wcusb usb-uhci] after it i recompile asterisk and after it i have
2006 May 29
8
E1 hardware for asterisk
Hi all, I need your lights :) There are many hardware provider for E1 cards on the market, what's your exeperience with E1 and what's the preferred provider for Asterisk out of Digium? Olivier