similar to: ring group containing external 10-digit numbers

Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work. I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized. Any other thoughts on how to solve this are also

Ok, Here is what I got working: A call comes in from a Zap line. 5 SIP extension ring if nobody picks up, the call is transfered to a cell phone number. That works. I not want to add a playback of a file ("Please waite while you are being transfered") before transfering the call to the cell phone. How can I do this? Andre

My TE411p does not seem to like to have two PRIs from different telcos (span 1 and span 2). I can get one working, but not the other. However, if I use vpmsupport=0 when loading the wct4xxp module, they both work. But here is the problem, vpmsupport=0 disables the on board echo cancellation. Any ideas? BTW, here is zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23 dchan=24

Hello, I was wondering if anyone had any ideas to a problem I am having. I have asterisk installed, and running. All extension phones are analog on FXS ports. The phones call each other and even transmit caller ID information but they DO NOT RING. Here is my setup: T400P Quad span T1 card currently connected to a CAC I w/ 12 channels of FXO for incoming lines and 12 channels of FXS for

I'll start by saying I'm a trixbox user, and a new one at that, so hopefully you can respond to me on those terms. I have a user who works from home 1 day a week. On that day I'd like for him to be able to connect with a softphone and be reachable by just dialing his extension as we normally would. I could set him up a new extension, then he could forward his phone there on

Could someone explain to a non-US dummy the following phrases I have seen on the list. "I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout." "We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6." What is 6/6? What is US48? What is blended? What is MOU? What is NPANXX breakout? -------------- next part --------------

Is it possible to configure * so that if a caller reaches voicemail for someone in Engineering, but doesn't want to leave a message they can press zero (0) and reach the Engineering Secretary or if they are calling someone in Accounting and reach voicemail, pressing '0' would reach the Accounting secretary, not the Engineering secretary? Don Pobanz

Modem connect speeds on calls through * seem to be lower than calls made through the telephone company lines or our old Rolm PBX. All data calls have 2 wire analog modems on both ends. For my set up I have channels of a Zhone channel bank tied to 2 modems. The Zhone channel bank interfaces my * server with a T400P card. modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -

Hi, Have looked around for info about this: <http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail If we are using 5 digit extensions (10102: 10 for the company, 102 for the extension), where can we put something so that "102*" goes straight to voicemail without waiting while the

After upgrading from 1.2.x to 1.4.x call detail records are not being written to /var/log/asterisk/cdr-csv/Master.csv In cdr_manager.conf I have [general] Enabled = yes Apparently there is something else that needs to be configured for call detail records in 1.4.x. Can someone point me in the right direction? Don Pobanz

I have a normal backup phone (and an alarm panel) sharing the POTS line with the Digium X101P FXO: | | Wall |>---+------X101P FXO as Zap/5 | | | Phone & Alarm Whenever the Phone is used, Asterisk sees a 'false ring' signal immediately when the phone is hung up. The Alarm panel dials out nightly at around 1AM, and each time it completes the call, Asterisk

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

Hi all, I am having a problem on to different boxes and to different channel banks. I can't get dial tone out of either one. I can still send and receive calls but no DT. This is the error that I get: Jun 8 15:23:54 WARNING[5021]: chan_zap.c:6283 handle_init_event: Unable to play dialtone on channel 95 asterisk-1.2.9.1 zaptel-1.2.6 zapata.conf: signalling=fxo_ls context=gene

Thanks anyone, I found the problem in rhconfig.h. After the fix I successfully compiled zaptel. V. --- Ron Wellsted <ron@wellsted.org.uk> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Have you built your kernel on that machine? > > The errors suggest that while the kernel sources are > installed, the > kernel has not been built. > > Check on

I did cvs update on asterisk, zaptel, libpri as of today (November 5, 2003). I also did 'make clean' on each of them. My previous version of asterisk was cvs of September 15, 2003. No other changes have been made to my system other that these updates. when running 'make asterisk' the following error appears term.c:55: conflicting types for `term_color'

When you have a bunch of analog phones that you want to connect to asterisk, but those analog phones have no transfer button, what are the options to allow the phones to transfer a call? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507)

Did anybody know, Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? Thanks a lot for help. Nico Giefing -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/3b657058/attachment.htm

Is there something special that needs to be done to allow a T100P/T400 to respond to a remote loop request? I am having some issues with a Point to point T1 line using zaptel ppp. This line gave us small problems when we had a pair of Cisco 2600's on either end but now with the zaptel ppp it is going down every couple of minutes for 15 intervals. The phone company says the line is good to

Below you will find, what I believe to be a typical setup with a T100P card. My question is - 1. Is this correct? 2. What kind of phones would be needed here... (Would you have to use Digital phones) And if so what would you recommend. PRI/T1----- | | | ---------------- | | | Channel Bank | | |

xen 3.0.2 dom0 (kernel 2.6.16-xen) always crashes, or rather freezes, whenever I copy more than 40GB of data. i checked syslog and there is no indication at all why it crashes. this happens whether the OS is FC5 or mandriva 2006, and is also independent of hardware, although we have mostly dell poweredge (1750, 2850, etc.) i am wondering if there some workaround. thanks! System Coordinator,