Re: [Asterisk-Users] Custom Menu Not Working
When adding the details in AMP for when caller dials 3, I have referenced it using 'custom-myapp,s,1', and if I go to 'extensions_additional.conf' I see the following line under the rest of menu item info that was created : exten = 3,1,Goto(custom-myapp,s,1) ; and in the extensions_custom.conf file I have [custom-myapp] exten = 3,1,SayDigits(1234) exten = 3,2,Hangup() Change to this [custom-myapp] exten = s,1,SayDigits(1234) exten = s,2,Hangup() since you send it to s,1 hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound of breathing
while using iax and a soft phone, the sound of breathing comes through so clearly that it has started bothering me. Earlier I was amazed at the quality, but now feel it is irritating. Wondering if there is a way to cut it down. I am in the process of exploring using iax for a call center, but this sound of breathing is a disappointment. Don't put the microphone right in front of your mouth. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conecting to asterisk server through NAT usingIAX
What is the simplest configuration to allow external clinets tocontect to my server. For me it was this entry in iax.conf [client1] type=peer usernamename=client1 secret=test context=sip host=dynamic allow=all Just a side note : you can't connect mutiple IAX clients simultaneously with the same registration. Else, only the first one registering will be successfull. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed to forward a call to X-Lite....
exten=2,1,Dial(capi/720:078***) exten=2,1,Dial(SIP/mateo01,15) On asterisk CLI, type show application dial ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference between 2 lines
Is there a way to make a join conference between 2 lines? like when you have 2 incoming calls and you merge them together with you? how can you do this on * if its possible? Transfert them both to a conference room, then join that conference. At least, that's how I would do it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conecting to asterisk server through NAT using IAX
I use linksys router. Now, I am trying to connect from outside to my asterisk server. I use Diax as iax client. For some reason I cannot connect to my server from outside. On my router I forward those ports to my asterisk server. 5060-5063 1-2 5036 4569 For IAX, only port you have to forward is 4569 UDP Notice the UDP, not TCP I'm using Linksys WRT54G and it works without a hitch. It works ok with broadvoice, but clinets cannot connect to the server. This is my iax.conf file [general] port=5036 well, here's your problem, port=5036. This is not the standard IAX port. comment that line or replace it with port=4569 hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Prompts with no sound
Yesterday I had sound problems with the voice prompts, I couldnt hear them, so I rebooted the system and voila, I was able to hear everything.. so I went to bad.. and I just woke up and tried the system again and its back!!! I dial the voicemail system and I cant hear the voice welcome.. I can hear any voice prompts Has anybody had this kind of problems? Only thing I can see is that you have a codec problem. Maybe you are allowing a codec that isn't supported by your phone. Maybe I'm wrong, but at least check that. Get on the console and see what codec the call is being handled with hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hi Newbie question
I am in Nebraska, US. I have broadband cable connection at my home. And I have friend and family in other country. Using asterisk and some hardware is it possible for me to call to landlines to other countries. whiout the need to go through or take any service from say Vonage or any other service provider. No. if you want to call them on the PSTN (landlines), your call need at some point to be interfaced with the PSTN. If you want to talk to them without paying longdistance fee, you can always get them VoIP phones (hard or soft) and connect them to your asterisk. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I exchange datas between two Asterisk servers ?
I'd like to establish way to exchange data between two remote Asterisk server. Something like call over IAX and send some structured data. Any advice ? I don't know if this could be done thru an IAX call. What you could do is something like this : - have a php script on one server that POST the data in xml to your other server or - have a php script on one server that GET the data in xml from your other server at least, that's how I would do it hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MultiLine Sip Phones
Come to think of it, why don't soft-phones have web interfaces? If you have a web accessible phone please tell me about it, off-line too, I need to increase my inventory of phone models. The snom sofphone they just released as a web interface, just like the real one. In fact, I think it's a nice thing. Maybe I'll do this in a future release of my softphone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wiki down?
is the wiki down again? It looks like it ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home ... the next step
And is there a specific _next_ place ( URL/URL/Wiki ) to continue to get [EMAIL PROTECTED] configured? Actually I'm testing at home since it is not considered a good thing to experiment with our business' lines. :-) http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zaptel.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax.conf config and iax based clients
Try using context (with a trailing T!!) in your config, and lose the spaces around the equal sign, just in case. Well, I was wondering why the error log showed that the phones where in default context. That just show that I should never answer before my first coffee ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Strategy for a stable IAXy
So, which way to go? IAX or SIP? IAXy or Sipura? I prefer by far IAX All ip phones use SIP right? Nope, now there's IAX hardphone, like there : http://www.iaxtalk.com/ hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax.conf config and iax based clients
correct your dialplan. something like this [from-iax] exten = 105,1,Dial(IAX2/QIax1,20) exten = 106,1,Dial(IAX2/QIax2,20) exten = 107,1,Dial(IAX2/QIax3,20) hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] transferring a IAX call into a conference
If someone calls me in on my faktortel number I cant transfer them to the conference call room. It literally disconnects them each time I transfer? Why is this? What can I do to prevent this. Any CLI log from when you try that ? Help us helping you :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone..easy to use ?
Im very new to this, so unsure what softphone I should use ? Can anybody provide me a link with a good Softphone ? (for windows) http://www.marccharbonneau.com/asterisk/mediaxphone.php Supports gsm, ulaw, alaw see also : http://www.voip-info.org/tiki-index.php?page=Asterisk%20IAX%20clients http://www.voip-info.org/wiki-VOIP+Phones hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
Can someone give me a simple rational explanation why a $5 analog handset gives me no echo whatsoever on an analog PSTN line, but PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require software-based echo cancellation. Surely a $5 analog handset does not have an echo canceller. The echo I mean is when I hear myself while talking to another party. When you talk on the PSTN with an analog phone, in fact you have echo, but it's coming back so fast, that you think that you just ear yourself while you are talking. No mix in the fact that you are talking on a VoIP phone, that takes the voice, encode it in the proper codec, send it on the network to your * box, * decode it, plays it on the PSTN line, takes what it ears, encode it back in VoIP, send it on the network to your phone that decodes it and play it back to you. Now, this adds a little delay, that'S why you ear yourself talking just after you actually said it. This delays make it so that you ear it in echo. While when you are directly on the PSTN, the echo comes back so fast that you ear it almost at the same time that you say it. When you are going only VoIP to VoIP, you don't have echo at all because there's no analog link (that's where the echo is) I hope I explained it well enough. Please correct me if I'm wrong 1. It is not in the Asterisk box because IP to IP calls do not suffer this malady Exactly 2. It is not from the Central Office to my premesis because my $5 analogue handset works without echo. Also PRI ISDN works without echo. Listen more closely, you'll see that there is echo. With echo cancellation, you ear the echo only for the first seconds of the call. Then the echo cancel is trained enough to suppress it hope this help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as VoIP gateway
I want to interconnect 2 pbx switches from to distinct location via an internet vpn using asterisk as VoiP gateways. The problem is what interfaces i must use between asterisk servers and pbx switch (FXO or FXS), and why? You must use FXS ports on *, then plug these in you PBX as phone lines. Then you can route calls thru thoses lines To your PBX, it will be just more lines available FXO (Foreign eXchange Office) is for connecting phone lines FXS (Foreign eXchance Station) is for connecting phones I hope I have the right terms in the definition, I just woke up and didn't finish my first coffee hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie help/pointers required - configure xlite with asterisk
I just want one of my incoming numbers to go to an IVR service that will allow me to select what I want. For example Press 1 for Mike, 2 for Karen, 3 for other, 9 for voicemail etc just put your incoming line in a context where you have a s extension Something like this (not really good one, just wrote it from the top of my head) [incoming-menu] exten = s,1,Answer exten = s,2,Playback(welcome-message) exten = s,3,... look in the sample config that came with asterisk, you have samples in there And remember, google is your friend ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web based Asterisk management tool
All I want is the web config tool ! Apologies if I am misunderstanding you here, as I say I am quite new to this and need to get up to speed fast For an Admin only web based product, is AMP my only option ?? If you want a web based config files editor, I have done one. But it's not advanced as AMP. Mine only let you vew/edit the different config files. It's like manually editing the text files on the box, but you do it thru a webpage. I that is what you want, I can send it to you. It's basically just 2 PHP pages. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web based Asterisk management tool
Why not share with the community? I do, like I do with my IAX2 softphone. It's just that I haven't took the time to make a webpage that explains what it does and provide a link to download it. I already send it to peoples on this list that asked for it. Anybody want it, just email (privately, since this list is already pretty busy) As soon as I have some free time, I'll do a page for it, I promise Sharing is caring ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] First Call straight to my extension
Thanks anyway. But what files is this? that I have to play with extensions.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone
I have specified an SIP extension (many, actually) in the sip.conf file but I cannot get DIAX to register with Asterisk. I've tried changing just about every variable I can while troubleshooting. One thing that is kind of suspect is what comes up after I have it re-read the config files: you are configuring it in the wrong place. DIAX is not an SIP phone, it's a IAX phone setup your account in iax.conf or, get a sip phone, like X-lite : http://www.xten.com/index.php?menu=productssmenu=xlite hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom Phones Volume
They work great, EXCEPT I have to have the volume turned all the way up in order to hear the conversation on the other end. I don't know if there's a fix, but I experiences exactly the same thing. This is the only negative thing I can say about the snom phones. Other thant this, they are really great phones If somebody know how to fix this, it would be more than welcome ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Encrypted VOIP?
If I remeber correctly, Mark Spencer is working on encryption in IAX2 hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd behaviour between Grandstream and Xlite
Whatever codec I choose in Xlite, when calling the Grandstream it always uses the GSM codec even if it is greyed out. Whatever codec I choose in Xlite, when getting called by the Grandstream it always uses ulaw even if it is greyed out. and what about the phone config in sip.conf ? what codec do you allow them to use ? I think * doesn't care what codec is grayed out in X-lite, her use what sip.conf tell him he can hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI with IAX
Does the MWI feature work with IAX2? I have read where it should but cannot get the indicator to work on any of the IAX softphones that I have tried which have this feature. I even did an IAX debug and did not see where and indication was sent to the phone when it registered. From what I know, * return a pseudo-boolean value with your registration acknoledge. - If you have messages, it returns 65535 - if you have none, it returns 0 - if you don't have a voicemail box, it returns -1 MESSAGE COUNT : 65535 there you have it, that means you have messages N.B.: it's doesn't work in MediaX phone because iaxclient's dll only raise a text event with a message that say if your registration is accepted or rejected. Exactly why is a good question. I only trap this message and update the indicator accordingly hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] First Call straight to my extension
Anybody with an idea how will I set me Asterisk to send call straight to my extension with out playing demo-Congrat MENU. Comething like this exten = s,1,Answer exten = s,2,Dial(IAX2/2001,20,tr) exten = s,3,Voicemail(u2001) exten = s,4,Voicemail(b2001) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new install
hi, i got an error while running the asterisk -v error message: error while writing audio data Well, that's the least verbose email message I've seen this week. Can you put more precision ? like what version of asterisk, the exact message you got on the CLI Help us help you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: 911 and Cops knocking on my door
The PSTN lagging would make sense and would my CDR reccord still show that 5911079 was dialed? Yes, the CDR won't show the w HTH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Developing an IP Phone
I just thought this link might be interesting to some of you. I know it's m$ware but please hold back the flames. http://msdn.microsoft.com/library/en-us/wcetarget5/html/wce50oriDevelopingVoIPPhone.asp That's fine if you want to develop an SIP phone, but if you want an IAX one, you can take iaxclient and compile it as a DLL. I did that and now I'm using it with Delphi. My phone is almost done :) I'll post it here when it's ready (really soon) N.B.: if somebody want the DLL, I'll be glad to share ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to compile iaxclient with MinGW/Cygwin
Can you give me some councils? Sorry I can't help you, I tried myself but couldn't manage to compile it. Somebody else on this list managed to do it, maybe he will jump in and provide some help (Preston, are you listening ? ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Developing an IP Phone
Yes, I would !!! If you wan't, I can send you a demo of my phone. Just drop me a private email, since this list as enough email traffic as it is ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Client
I'd like to develop an IAX - client. Does somebody know where can I get the source code for an IAX client? Please be more precise : is it a Windows/Linux/MacOSX/other that you want ? Anyway, have a look here : http://iaxclient.sourceforge.net/iaxcomm/index.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Softphone
For all the peoples that wanted to test my windows IAX2 phone, I've put it up on a server where it can be downloaded. For the ones that wanted the DLL, it's available on the same page. For the DLL, I will post a list of the functions in it and the parameters it expect as soon as I have some free time. All comments (good or bad) are welcome The phone can be used mostly with the keyboard : - Dial with the keypad, press ENTER to start the call - # is replaced with / for convenience - Pick up a line with the function keys (F1 for line 1, F2 for line 2) - ESC to hangup currently selected line The phone is self contained : the DLL is in the EXE as a ressource, will extract it if not found. Waiting for your input http://www.marccharbonneau.com/asterisk/mediaxphone.php ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Developing an IP Phone
DO you know of companies that will re-brand ip (sip/iax) phones? Have a look at my phone, I can rebrand/modify/etc http://www.marccharbonneau.com/asterisk/mediaxphone.php HTH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outlook Integration
Unfortunately the PBXtray app only works with our systems, and we cannot sell it separately. It is not released under the GPL because there are no modifications to Asterisk or any related software for it to run. Let me get this straight... Only works with their system, but there is no modifications to Asterisk for it to run ? I don't get this one ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home and Zap Channels
I have one more question that I can't seem to get straight, The ZAP channel phone, I can't dial any other extentions from it, I just get a fast busy. Same if I dial 9 to use the outside trunk. It works great from the SIP soft phone, but I can't seem to get the FXS phone to behave. In your zapata.conf, where you defined your FXS port, you have to put it in the right context so it as access to the other extensions. Just put the same as your softphone ex.: extension=localstations HTH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x100p issues + TDM400P
Does TDM400P wildcard has FXO and FXS ? This card as 4 ports, and on each port you can put an FXS or FXO module. So you can make any combination : 2 FXO and 2 FXS, 4 FXS, 4 FXO, etc HTH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange Crash
memtest86 is a nice tool and if you go to their site(http://memtest86.com), they have an ISO bootable image there also. Knoppix also can be used to test memory On the boot prompt just type memtest and it will start the test HTH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcement to caller when called party has picked up - without initial Answer()?
How do you want to play something on the line without answering it first ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] redirect different phone number to different IP phone
I have a simple question but I cannot find the answer. I have a line with 2 different phone numbers I want to redirect each phone number called to a different IP phone Example Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2 Someone calls 5551235 and the call is redirected to IP phone 192.168.0.3 Just put both incoming lines in a different context and have an extension s,1 that dials the phone you want. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Am I missing something really basichere?????helpwith Asterisk@home {Scanned} {Scanned} {Scanned}
I have no idea what AMP configurator is? http://amp.coalescentsystems.ca/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Moh in meetme doesn't work if I transfer to meetme
It's hard to tell without seeing your config files, but ... your first trace show : -- Started music on hold, class 'default', on IAX2/[EMAIL PROTECTED]/1 while your second trace show : res_musiconhold.c:466 moh_alloc: No class: random So it looks like the context the tranfert is from as MOH class defined as random but you don't have a class random defined in musiconhold.conf, but default is defined. HTH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi Asterisk Server Transfers
Anyone know a good IAX phone (not softphone)? Only phones I know that support IAX can be found there : http://www.iaxtalk.com/index.php ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Softphone
You should have a look here : http://www.geocities.com/babarnazmi/middlepage.htm IAXClient ActiveX Control-IAX2 ...full IAXClient engine for creating IAXClients standalone or web based phones... HTH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P/TDM22B dialing issue
If I'm not mistaken, add a 'w' in your dialstring to make asterisk
wait half a second like this :
exten = _1888.,1,Dial(Zap/g1/w${EXTEN})
you can put multiple w to make it wait longer
HTH
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Re: [Asterisk-Users] Inbound analog Telco line not answered
Asterisk Message -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at inbound-analog,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at inbound-analog,s,1 still failed so falling back to context 'default' Looks like your zapata.conf define the context for the incoming lines as inbound-analog, but I don't see this context defined in your extension.conf So, asterisk fallback to the default context, in which you defined : [default] include = ext-local exten = s,1,Playback(vm-goodbye) exten = s,2,Macro(hangupcall) -- Executing Playback(Zap/1-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'en') -- Executing Macro(Zap/1-1, hangupcall) in new stack See, * just follows what you told him ;) I think you wanted to put from-pstn as the context of you Zap channels HTH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updating Asterisk
Just tried it. Show version still shows: Connected to Asterisk CVS-v1-0-12/21/04-14:14:46 Well, only thing I can see is that your CVS download didn't went right, or you downloaded it into a different place, because you're not even at 1.0.4 Follow these simple steps to update you tree : # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout -r v1-0-5 asterisk # cd asterisk # make clean; make then, stop asterisk # make install then start asterisk HTH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone
(IAXPhone): I suppose you're talking about Steve Sokol's phone If so, then this phone only support gsm. Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm packets - it should transcode if necessary ? I have enabled gsm and ulaw in both configs, but it seems not sufficient. Yes, * will transcode, but you specified in the IAX Phone config that you allow this one tu use gsm AND ulaw, so instead of transcoding, * just tell the IAX Phone to switch to uLaw, since the originating party sends it in ulaw. Just change your iax.conf to only allow gsm on the IAX Phone like this : disallow=all allow=gsm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone
thanks for info. Which iax softphones are using newer iaxclient ? What is the best iax softphone from this point of view ? I don't know for sure, but I think iaxcomm and DIAX are most up-to-date. I'm almost finished building my IAX softphone that is based on a recent version of iaxclient (one from the beginning of this month). I't will be free for non-commercial use. I'll post it here when it's available ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone calling Zap/1 on TDM20B rings and answers but not hear voice
Could you give us the output of the console when you try the call ? That would help us to point you in the right direction. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power Alarm Error - Help
I have been getting the following message in Asterisk and it shuts Asterisk down, needing a reboot. Power alarm on Module 2 I have (1) TDM400P with (2) FXS (2) FXO cards (1) X100P card Any ideas? Since nobody answered, I'll guess something :) Did you plug the power on the TDM400P ? since you have FXS ports, you need to plug it in ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound analog dialing with Internet Line Jack (fwd)
I've been trying to setup asterisk with an Internet Line Jack card for sometime. I've been successful in configuring asterisk to handle incoming calls, make calls between sip phones, call the asterisk demo, and even When the call comes in, what's the channel * reports handling ? Something like Zap/1 maybe ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Install Method
I would suggest you go with the easy road : - install CentOS : http://www.centos.org/ - then download Asterisk 1.0.4 (latest stable) : ftp://ftp.asterisk.org/pub/asterisk/ - install it by following this document : http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation then play with it, read some more, play again, etc... your best source for info is http://www.voip-info.org/wiki-Asterisk, google and when you don't find the answer in one of these two, this list is your last hope :) Hope this help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Webmin Module for Asterisk
There is already one, you can find it here : ftp://ftp.asterisk.org/pub/asterisk/webmin But I never managed to make it work, maybe it should be updated Anybody wanna take the challenge ? :) BTW, I've done some web pages that show you your configuration, and let you edit the text files in your browser. If you want it, drop me a message ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: how to manage Digium TDM04B outgoing calls
Last concern about making my channels in a group and add that group in my dial plan. How can I make sure it will start with channel 4 and not pick a random one between the 3 channels as I'm pretty sure if I put in my dial plan a group having channel 2, 3 and 4 it might do the opposite and start with channel 2 then if it's busy switch to 3 and then 4 instead of 4 then 3 then 2 no? I think * start with 1, then 2, ... until it finds an available channel. I you really want it to start with 4, then 3 ... I think just re-managing your lines so that you primary number (line 1) is plugged in port 4, and vice-versa, then put all those lines in the same group, and tell * to dial by this group, it would solve your problem. If I'm wrong, please correct me ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Ten lite troubles.
[xlite1] ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend username=xlite1 secret=guessit callerid=Jane Smith 5678 host=dynamic ;nat=yes ; X-Lite is behind a NAT router ;canreinvite=no; Typically set to NO if behind NAT context=remote-station disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw allow=alaw Just ajust to your environnement (username, password, context) HTH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on SIP -- not on analog.
Echo will only happen when you go from digital to analog. Here is more info on this : http://www.voip-info.org/wiki-Asterisk+echo+cancellation http://www.voip-info.org/wiki-Asterisk+echo+analog+lines http://www.voip-info.org/wiki-Asterisk+Echo+Avoidance And remember : Google is your friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't initiate a call with X-Lite.
to which entry have to corespond Domain/Realm parameter in X-lite just put the same as your SIP Proxy, that is your Asterisk box address ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] is asterisk a good solution?
I need a SIP server which allows upper registration. We want to offer the possibility to make voip calls to our users, allowing to call to pstn. But only starting calls, not receiving. Do you think asterisk could be a good solution? Of course it is, you can make what you want with it. That's the power of open source right there :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Communication Between Phones... I can't test :(
My question is : when Desk1 call Desk2 , server (desk3) will authentificate phone but i want to known if Desk3 use bandwitch during communication? depends if Desk1 and Desk2 are behind NAT, and if you configure your accounts to let them reinvite or not. If you let them reinvite, and if they can talk to each other directly, no, Desk3 won't use bandwith, except for the authentification. HTH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS tagging - can Asterisk do this, if not, what do you recommend?
This DLink switch will prioritize data if it has already been tagged by either the ATAs or Asterisk. In reviewing the Asteisk documentation I can not see that Asterisk offers this functionality (of course I could have missed the information). So my question is, does Asterisk offer the ability to mark the voice data with the proper tags so that our switch can prioritize the data through our network, or if it can't what hard ware form sip.conf and iax.conf ;tos=184; Set IP QoS to either a keyword or numeric val tos=lowdelay ; lowdelay,throughput,reliability,mincost,none have a look here http://www.voip-info.org/tiki-index.php?page=Asterisk+QoS ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?
I know as a Canadian I'm not interested in a list Just for Canadians -- It's just fragmenting the help available for very little benefit. I do, however, appreciate the thought. -A. I'm a Canadian also, and I second that ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 lost after reboot
Hi My card is working, but when I reboot the machine, most of the times it is not working, I get ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6) To make it work again I have to shut down, remove the card, reboot so kudzu will remove the config. shut down again, put the card back in, reboot, now kudzu see it, I choose Ignore and then it's working again (until the next reboot). I'm on WBEL 3.0 and the card is not sharing is IRQ. Is anybody else having this problem ? When kudzu see it (as a Jens Schoenfeld Intel 537), what should I choose ? Is there something I can do to prevent this from happening ? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P NO BATTERY Poopy???
doh! i assumed the x100p and TDM400p worked the same, because i thought was able to do both on that card...well thanks for the help :( Side note : you just have to get 2 FXS modules for your TDM400, the card can use FXO or FXS modules, and you can mix them as you wish ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No more loading asterisk...
Jan 15 16:37:24 ERROR[7573]: chan_iax2.c:7486 load_module: Unable to bind to 0.0.0.0 port 4569: Address already in use That is because you already have something listening on the 4569 port. I think another instance of Asterisk is already running ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly repeats registering to * server
Is the reregistering normal behaviour for an external client ? Yes, IAX default behavior is to register every minutes or so, external or internal If I'm wrong, please someone correct me ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote Voicemail Retrieval...
I want to listen to voicemails on my * box from a phone that is not local to my pbx. I.e., from my cellphone or my PSTN work line etc. I'm aware that I can forward VM to email or use a web interface but that is not always practical. Other than doing an IVR type arrangement or a phone number dedicated to VM access is there a way to do this? On my old POTS line I used to be able to call my line and simply punch * during unavailable message playback to go to the equivalent of voicemailmain(). Is there a way to do this in *? You can include the voicemail extension in your incomig-line context That way, while you are in the main menu, you could punch 8500 (or whatever extension is you voicemail) At least, that's the way I did it Hope that help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on a notebook?
Yes, * can run VOIP-only. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ports to open behind a NAT
From searching the list archive I have come up with the following list 22 for SSH Should this be TCP, UDP or Both? TCP 5060TCP Only 1 -2 UDP Only Is this info correct or is there other ports or port type corrections above? Yes, for SIP it is correct ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] not sharing IRQ's
just to make sure: when i have zaptel devices on my box and i also use meetme and iax2, do i need to have USB device enabled and it's modules loaded? No your zaptel device will provide the needed hardware timer the USB timer hack is for when you don't have any digium card ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the best and easiest flavor to be used with Asterisk.
try this : http://www.whiteboxlinux.org/ On Wed, 12 Jan 2005 16:58:29 +1300, Imran Sadiq [EMAIL PROTECTED] wrote: Could anyone please advise me on the best flavor of Linux on which Asterisk is easiest to install. I am currently using RH8.0, everything over the IP works fine but when I want to call a physical line I can only have conversation for about 3 sec and everything freezes after that. I have to hard reset the machine to bring it back up. Any suggestions will be greatly appreciated. Thanks Imran Sadiq Systems Engineer Tel: +64 9 377 8282 World Class Support for any business Fax: +64 9 377 7900 with between 7 and 70 computers. Mob: 027 286 9269 LANcom Technology Limited: 25 Union St, Auckland, New Zealand ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do i talk to the IAXy...? (Newbie Alert)
I just tested another way... On windows, install Cygwin, download iaxyprov, make, and you can run it under Cygwin It works, I just provisioned my IAXy with it ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] kind of urgent
I have * working on FC2 with SATA drives. I would wait to go FC3 untill it matures a bit. Hope this help On Thu, 6 Jan 2005 19:32:24 +0200, Shoval Tomer [EMAIL PROTECTED] wrote: Hi all. Can anyone comment why shouldn't we use FC 3 for an * production system? I'm not looking to start a distro war, but we just found out that redhat 9 (and FC 1) don't support SATA drives, and apparently FC 3 does. We are only familiar with red hat and are in a point in time that switching distros is not available. The guy installing the system is already on location. Yes, I know we made a silly mistake. Please help us... Thanks. Shoval ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM4000P with 4 FXO's not picking up ringing lines
maybe a stupid question but, did you include Answer in your dialplan ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: phones with two ethernet ports
Snom 190 and 220 also ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 8 pstn lines+ on Asterisk supported hardware.
I have an Asterisk with 2 TDM with 8 FXO modules and I don't have any problems. One thing to look for is that the cards don't share any IRQ. Use a motherboard where you can assign IRQ to the PCI slot. I used an Intel board. Hope this help On Mon, 3 Jan 2005 19:43:12 +0200, Hadi Jadallah [EMAIL PROTECTED] wrote: Hi all, I have this project that requires me to use 8 PSTN lines and possible more. I was thinking 2 TDM cards with FXO modules. The I got to read the Qs about FXO/FXS cards thread and that scared me. Can anybody recommend anything that is known to work ok with no mysterious problems? I was thinking OpenSwitch12 cards. What do you guys think? Any help is appreciated. Regards, Hadi -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.290 / Virus Database: 265.6.7 - Release Date: 12/30/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
