Re: [asterisk-users] [asterisk-dev] Rgd Zaptel code for Asterisk
I need one more help can u plz explain what zaptel does it exactly. This discussion probably needs to move off the dev list. Zaptel is the set of driver modules that connects hardware from Digium and other various vendors to Asterisk through chan_zap. This information is available if you look for it. http://www.voip-info.org/wiki/view/Zaptel William ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE210P Vs TE220P difference
Dear all anybody have idea of this 2 card and performance vise which one is suggestable ??? If you had done a little bit of legwork, you'd have noticed that the TE220 has a PCI-Express interface while the TE210 has a 3.3V PCI interface. There is no difference between them in terms of performance. Which one you purchase depends on what interfaces you have available in your server. William Moore ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE220B
Has anyone ever had any problem with the TE220B card with it showing up as four ports instead of two. I RMA'd the first one with the retailer (Digium tech advice), but I just got another brand new card and it is coming up as four ports again. The card identifier is showing 0420 when I do lspci. Has this happened to anyone and if so is there a fix? I don't know why, but your PCI subvendor ID seems to be set to the wrong value. Unfortunately, it's probably not modifiable with the tools at your disposal. There is a way to make this card work properly, but it will make any 420's in the system act like 220's as well. Open up wct4xxp/base.c and search for 0220. You should find yourself in the pci_device_id structure. You'll need to modify the 0220 to be 0420 and you'll also need to comment out the line above it that contains 0420. After that, run make install and reload the wct4xxp.ko driver. Your TE220 should work fine with those modifications. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receiving SIP calls without registeration and dynamic IP address
* I was asking if the endpoint send a call, and it has a username and password typical to that configured in SIP.conf file, then should this end point being registered or not? If you are only *SENDING* calls to asterisk and not receiving, you do not need to send a registration. You only need to send a registration if you want to *RECEIVE* calls from asterisk. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi port IAX Gateway
On 6/26/07, Mike Hammett [EMAIL PROTECTED] wrote: I am looking for a gateway that has several FXS ports and uses IAX. I have a need for 16 ports, but will accept 6 or 8 port gateways as well. Here is an 8 port gateway that should suit your purposes: http://www.digium.com/en/products/hardware/asteriskappliance.php Unfortunately, I think they're only selling the developer's kits at the moment. I don't know when the retail version will be out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kore dump
On 6/27/07, Ed Nuñez [EMAIL PROTECTED] wrote: What is a god Windows application to read core dump files? No. Core files must be examined on the same system that created them. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Execute ChanSpy
On 6/19/07, Carlos Garcia Mujica [EMAIL PROTECTED] wrote: With what command can I execute chanspy throw Asterisk Console. THANKS. You are less likely to be answered if you spam the list. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p and te110p configuration.
On 6/14/07, Matt Scott [EMAIL PROTECTED] wrote: I purchased FXS modules so that I could terminate the machines or faxes (eg just like a standard phone) the outgoing/incoming channel will be be provided by my E1. I hope I have the right modules for the job? You do indeed have the right modules for the job. FXS modules terminate phones and the like. Below is what you'll need for your basic configuration. zaptel: loadzone = uk defaultzone = uk span = 1,0,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 # note that fxs modules use fxo signaling because they're acting like the telco fxoks = 32-35 zapata: [channels] language=en usecallerid=yes hidecallerid=no callwaiting=no callwaitingcallerid=yes restrictcid=no usecallingpres=no threewaycalling=yes callreturn=yes transfer=yes cancallforward=yes echocancelwhenbridged=yes echocancel=yes musiconhold=default rxgain=0.0 txgain=0.0 signalling=pri_cpe switchtype=euroisdn immediate=no overlapdial=yes pridialplan=unknown prilocaldialplan=unknown group=1 context = from-pstn callerid=asreceived channel = 1-8 context = from-local ; again, fxs modules use fxo signaling signalling=fxo_ks cidsignalling = v23 cidstart = polarity channel = 32-35 Another thing you need to check is that the card itself is in E1 mode and wcte11xp is loaded before wctdm. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM bus extension.
On 5/25/07, Alex Balashov [EMAIL PROTECTED] wrote: In reference to an old post from 2002: http://www.marko.net/asterisk/archives/0203/0103.html How does one go about doing this? I think what mark was referring to there is dynamic spans. They actually work over a standard ethernet network. They are configured in zaptel.conf and zapata.conf just like any other zaptel device. Also, what is the present status of the OpenSS7 stack in Asterisk? What can it do now? The SS7 stack in asterisk is still under development. Any comments mattf? And is there any possibility in the future of developing a DS3 card for it, if only for the purpose of mostly DACSing? Which is still a level of intelligent call control on the TDM bus that is highly in demand for VoIP applications that require PSTN interconnection. This is probably a question for Digium's marketing/sales team. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks
On 5/25/07, Matthew J. Roth [EMAIL PROTECTED] wrote: List users, This post contains the benchmarks for Asterisk at low call volumes on similar single and dual-core servers. I'd appreciate it greatly if you took the time to read and comment on it. Are you recording memory figures as well and have you checked the total used memory? Or did I miss it somewhere? Thanks for doing this, scalability testing is always good. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Integrated T1
On 5/24/07, Jeremy Mann [EMAIL PROTECTED] wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? Yes, zaptel will create a device node for you. Take a look at the set-hdlc tool in zaptel and the less common channel types in the default zaptel config file (rawhdlc is one, there are also others). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Integrated T1
Here's a link that will get you most of the way there: http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration If you have any issues with setup, I recommend you contact Digium's support to help you since I'm sure they've had the most experience with it. On 5/24/07, Alex Balashov [EMAIL PROTECTED] wrote: On Thu, 24 May 2007, Jeremy Mann wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? The Zaptel/Asterisk infrastructure can definitely break particular timeslots out of the T1 for voice, but it is not my impression that any existing WAN drivers for Linux support Digium cards or cohabitation with Zapata and can give you a serial data interface on other channels. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Echo
On 5/22/07, Asterisk [EMAIL PROTECTED] wrote: Hello all, One of our clients reported that they are experiencing echo in SIP calls (even on internal ones). What do you think could be causing echo in internal SIP calls? We're using Polycom telephones, do you think they could be causing it? By nature, pure SIP connections will not generate echo. This is an acoustic problem on one or both ends. If your client is using headsets, make them get better headsets. It also might be that they absolutely love speakerphone and the room's acoustics are bad. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial out issues.
On 5/22/07, Matt Scott [EMAIL PROTECTED] wrote: Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work well. Error Message: [May 22 11:01:48] WARNING[9179]: channel.c:3024 ast_request: No channel type registered for '(Zap' [May 22 11:01:48] WARNING[9179]: app_dial.c:1090 dial_exec_full: Unable to create channel of type '(Zap' (cause 66 - Channel not implemented) It says channel not implemented. Did you compile asterisk before or after compiling and installing libpri? If you type module load chan_zap.so, what is the output? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial out issues.
On 5/22/07, Morgan Gilroy [EMAIL PROTECTED] wrote:
In your dial lines you have an extrac comma (,)
exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
should be
exten = _9xxx,1,Dial(${OUTBOUND}/${EXTEN:1})
or
exten = _9xxx,1,Dial,${OUTBOUND}/${EXTEN:1}
Good catch Morgan!
___
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Re: [asterisk-users] PRI got event
On 5/16/07, Oscar Atienza [EMAIL PROTECTED] wrote: Hi all, I have 1 Card Digium TE412P and 2PRI E1. I have more problems with drops lines. The asterisk log is this: May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 May 16 10:52:26 WARNING[4465]: chan_zap.c:2287 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 This indicates an unstable d-channel. Try changing dchan in zaptel.conf to hardhdlc. If that fixes it, you are missing interrupts for one reason or another. I would also advise that you call Digium's tech support. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Module wctdm24xxp not found - TDM808P on debian
On 5/11/07, Juliano Fernandes Schroeder [EMAIL PROTECTED] wrote: I'm trying to configure a TDM808P card on debian. When I modprobe wctdm24xxp i get this error FATAL: Module wctdm24xxp not found. FATAL: Error running install command for wctdm24xxp I think i have successfully compiled the zaptel drivers, and the card appears when i do a lspci 02:06.0 Ethernet controller: Digium, Inc. Unknown device 0800 (rev 11) Subsystem: Digium, Inc. Unknown device 0800 Flags: bus master, medium devsel, latency 64, IRQ 3 I/O ports at e800 [size=256] Memory at fe20 (32-bit, non-prefetchable) [size=1K] Expansion ROM at 2000 [disabled] [size=128K] Capabilities: [c0] Power Management version 2 I've searched for a solution with no success. Another problem is that when i do a ztcfg -vv i get Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected I don't know if the errors are connected and would appreciate some help. Did you make install in the zaptel source directory? If you didn't, it didn't put the kernel modules in your kernel's module directory and run depmod, so your system doesn't know about the modules. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10 FXS - Channel Bank or PCI Card?
On 5/9/07, Gavin Henry [EMAIL PROTECTED] wrote: Hi All, What do you recommend? I was looking at: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html But it will be 3 PCI slots. You could do it in one slot with Digium's TDM2400P (you would actually have to get 12 channels since they come in groups of 4). It tops out at 24 channels terminated at an amphenol connector, so you'll need a breakout box if you go this direction. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Daemontools and holidays macro
You may want to consider renaming daemontools as it is also the name of a windows program that allows you to mount CD/DVD ISOs, so there could be some confusion. On 5/2/07, Steve Totaro [EMAIL PROTECTED] wrote: Vicente Aguilar wrote: Hi I've recently released the daemontools scripts I use to run both Asterisk and Flash Operator Panel, and a macro to tell whether today is a holiday or not and jump to different dialplan places accordingly. They are here: daemontools scripts: http://www.bisente.com/blog/2007/04/27/spanish-asterisk-y-daemontools-spanishenglish-asterisk-and-daemontools-english/?lan=english is-holiday macro: http://www.bisente.com/blog/2007/04/30/asterisk-holidays/?lan=english Hope you find them useful. Any feedback or improvements will be appreciated. :) Regards, Thanks, Email moved to my Useful Asterisk Stuff Folder Thanks, Steve Totaro www.asteriskhelpdesk.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
On 5/3/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Yes, Asterisk will do the conversion from SIP to IAX and back again if necessary. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] echo cancellation and ztdummy
On 4/23/07, Patrick Fortin [EMAIL PROTECTED] wrote: Are echo cancellation parameters useful when using the ztdummy driver and no physical card ? No. The echocan software and hardware only cancel hybrid echo. They do not cancel acoustic echo that would be generated by voip phones with bad speakerphones or bad headsets. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3rd T1 of quad card won't change signaling
On 4/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Apr 19, 2007 at 02:44:21PM -0700, Jay Wilton wrote: Hello, I'm trying to set the 3rd span of a new digium quad card as a EM T1 for Faxes to a Hylafax server. The 1st and 2nd spans are working as PRIs. When I start asterisk, the logs show a signaling error and chan_zap.c dies. I also get an error that it can't read the gains but they are the standard shown below. 2.6 kernel, Debian Stable, * 1.2 svn from feb 2007 my procedure: make changes to zaptel.conf zapata.conf rmmod wct4xxp modprobe wct4xxp ztcfg -vv #shows 1+2 span as PRI, 3rd span as EM If ztcfg -vv spits out something about CAS signaling, try ztcfg -vvf. It forces a reconfigure on all of the spans/channels. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pci 2.2 - pci-e x16
On 4/20/07, Olivier [EMAIL PROTECTED] wrote: 2007/4/20, asterisk [EMAIL PROTECTED]: Hi, Does anyone know if it is possible to plug a tdm400p pci digium card into an pci-e 16x slot ? np Olivier meant no here as well. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trigger a wake-up call from the shell?
On 4/17/07, Donovan Niesen [EMAIL PROTECTED] wrote: I have set up a script that ensures certain services are up on my Asterisk box (Trixbox 2.0). I would like it to trigger a wake-up call if certain conditions aren't meant. How might I accomplish this from the shell? Take a look at call files. They allow you to generate a call from * to a phone and then do whatever you want with the other end (play a message, connect you to a tech, etc.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TE205P and channelbank
Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? In my experience, I have never had an issue with faxes and Digium's cards, but I'm sure many people will beg to differ. also, would I need a crossover to the channelbank or is it a patch lead like the connection to the PRI You will most likely need a crossover cable. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
You seem to have misplaced your message/comment/question. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T100P -- TE120P
On 4/9/07, Carlos Chavez [EMAIL PROTECTED] wrote: No, that particular model is not able to use an E1. You need a TE110P which is able to select between both. They used to have an E100P card that was E1 only, both were replaced by the TE110P. The TE120P is the updated version of the TE110P and is much more compatible than the TE110P. It also has a hardware echocan port (the TE110 doesn't). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] Upgrade 4 to 8 Analog Lines Question
On 4/9/07, Jim Freeze [EMAIL PROTECTED] wrote: Or a new Digium TDM880B replacing the old TDM40B for only one IRQ... Do you know if this board will fit in a 2U machine? The TDM800P is about the same height as the TDM400P and is about an inch longer, so you should have no problem putting it in the same slot as the TDM400 was in. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE120P and Unknown Signalling Method
On 4/2/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: I have a brand new TE120P card that I have installed and asterisk is not starting as I am getting the error: ERROR[5054] chan_zap.c: Unknown signalling method 'pri_cpe' Make sure you have libpri installed and that it is the right version for your version of asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] modem passthru
On 3/20/07, Mark Farver [EMAIL PROTECTED] wrote: I suspect the issue is caused by the echo canceller, since I believe the issue appear about the time we turned echo cancellation on (for the IAX users). We don't need echo cancellation for PRI to PRI calls. I've looked around, but I am finding conflicting opinions on what the echocancelwhenbridged line does. Some say it turns off the echo canceller for TDM to TDM calls if set to yes, some say if it is set to no. Which is correct? Mark, The name of the option is fairly clear. If you want echo cancellation even when the call is bridged directly from card to card, set the option to 'yes'. Otherwise, set the option to 'no'. This option should be 'no' in the majority of cases. William ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 and 3.3v pci
On 2/12/07, Paradise Dove [EMAIL PROTECTED] wrote: my card has just fxo modules and is put in a 3.3v slot. when running modprobe wctdm24xxp it waits for ever and dmesg shows Resetting the modules what could be the problem? when i put this card in another system with 5v slot it works fine. I would call Digium's tech support. They open in 20 minutes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium S101I as a traveling companion
On 2/11/07, Joseph [EMAIL PROTECTED] wrote: I have a few questions with regards to Digium S101I adapter. I would like to use it as a traveling companion, plugging it into various networks (all behind firewall I assume). I'll be registering it into my asterisk server (behind firewall, port 4569 will be open). 1.) If I plug that small adapter S101I into another network will that network have to have port 4569 open as well? As long as the server is on a public IP, you don't have to do anything to the local network you're on. All data should pass normally. 2.) If I'll have two of these units registered to my Asterisk server (both units in different places). When I make a call between these two units S101I, does the connection (the bandwidth that is being utilized) goes through my server or directly between these two adapters? I'm not too sure on this one. I read some info somewhere about IAX being able to send voice data directly between endpoints, but I'm not sure whether or not the S101i supports it. William ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 and 3.3v pci
On 2/11/07, Paradise Dove [EMAIL PROTECTED] wrote: does TDM2400 work on 3.3v pci slot? Yes, all of Digium's analog cards are dual voltage and can work with either 3.3V or 5V slots. You just need to make sure you have an extra molex connector if you're going to be using FXS modules on the card. William ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] On what distribution is www.asterisknow.com based on ?
What is the package manager used? And what is the added value compared to the well maintained debian based asterisk ? Hi Maxim, AsteriskNOW is built on top of the R-Path linux distribution which uses conary as the package manager. There is no difference between the version of Asterisk included with AsteriskNOW and the source code obtainable from asterisk.org. It is meant for those who do not wish to or know how to administer their own linux server. William ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap calls
On 1/10/07, Jay Moore [EMAIL PROTECTED] wrote: Can I repeat channels like that or will it cause Asterisk to choke? If I can't do it that way, can someone suggest a way to do it? It will not cause Asterisk to choke, but when you assign the group the second time, it replaces the first assignment. I think you will want to specify your groups as such: group=1,9 . . . group=2,9 etc. William ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Background switch to different context
If you type show application background on the *CLI, you can see all the options listed there. The last optional argument is the context that you want to use to look for extensions. On 12/28/06, Time Bandit [EMAIL PROTECTED] wrote: I am using the Background() function to ask for the extension, but the extensions are in a different context. Is there a way to tell Background() to look for the entered extensions in another context other than the currently running one? in that context you can do include = other-context hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quadBRI beronet card: how to specify which ISDN channel to use to make calls
Giorgio, I believe the syntax for mISDN is mISDN/port:channel/number. In other words, replace your - with a :. On 8/25/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, I have a quadBRI beronet ISDN card. Is there anybody who knows how to choose the channel to make calls? I tried with Dial(mISDN/1-1/) to choose channel 1 of port 1 but without success. TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No CLID from PSTN using X100P FXO Card
However, reinstalled the box from ground up and installed 1.2.10 and now
CLID isn't working at all. The PSTN line is still transmitting it, as
I've plugged in my Uniden cordless with CLID and it shows up fine on
there, but getting absolutely nothing inside the ${CALLERIDNUM} and
${CALLERIDNAME} variables.
In 1.2.10, these variables have been changed to a single function.
The new way to access those would be ${CALLERID(name)} and
${CALLERID(num)}.
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Re: [asterisk-users] Dialplan or matching
On 8/18/06, David Cook [EMAIL PROTECTED] wrote: Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches sort of like the SPA's can? Tollfree numbers for example. I can have a line for each combination: exten = _1800NXX, Dial, exten = _1866NXX, Dial, exten = _1877NXX, Dial, exten = _1888NXX, Dial, But I want to do is something like this: exten = _18[0678][0678]NXX, Dial, . This syntax is valid and would work for what you're doing, but as you said, there is a chance of logic error in it. Or to prevent the logic error which albeit small, the above would create: exten = _18[00,66,77,88:2]NXX, Dial, .. (representing that the next 2 chars must equal any of '00'.'66','77' or '88' As for this syntax, Asterisk does not respect the [], so it parses the 66 as the priority. I have no idea how to properly do this in one line if it is possible. Kinsey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
