similar to: FXO recommendation

Hello I have just set up Asterisk Asterisk 1.4.21.2 on a CentOS 5.2 machine. I am using the OpenVox B200P ISDN card. My problem is that even though chan_misdn module seems to be loaded correctly with Asterisk (I can see it using 'module show' command) the misdn commands are not available to me in the CLI so I cannot tell if my box is correctly interfacing with the ISDN card Any ideas

Good Afternoon, Can someone please explain what Y-cords are available out there and how they can be used with Aastra or other VoIP phones? Maybe with or WITHOUT headsets? Isn't a Y-cord traded for soft Barge in these days? Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Is anyone aware of any progress on this bug? http://bugs.digium.com/view.php?id=8763 Not only is the channel randomly disappearing during idle periods, it vanishes during a call as well. No indications in dmesg, syslog, asterisk or anything. Only cure is to rmmod and modprobe again. I'm currently on 1.4.0. Any ideas would be greatly appreciated. Cheers, Kyle -- Kyle Gordon

Hey all, I'm running the latest and greatest CentOS4 here, along with Samba and Winbind coupled to an Active Directory server. It's all working smoothly, bar one little bit, the idmap gui and uid directives. It appears to be ignoring them completely. I've pasted the relevant directives below... winbind separator = + idmap uid = 10000-20000 idmap gid = 10000-20000 winbind enum users =

Hi all, I've got a device that seems to become unreachable for about 2 minutes, every hour. From what I can tell, it isn't due to network or server issues. Any ideas? TIA. -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701

Chris you might know the answer to my HUUUUUUGE problem A few weeks ago you posted this message: "I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which is currently working happily with an SDX Index phone system. I have to replace this phone system shortly and I've been trying to get a * system working for some weeks now. I have configured the dial plan (which works)

Hello asterisk-users, My SIP phones are in pickupgroup, and if some of them ringing from other phone can pick up with *8 as usual. But I want to know if this happen. I've tried the a extension, but seems not working. Any other idea? -- Best regards, Gergo mailto:csibra at gmail.com

Hey guys, just thought I'd let you know that I'm giving away a copy of "TrixBox Made Easy" on The Asterisk Blog <http://www.asteriskblog.com>. Check it out. -- www.AsteriskBlog.com Your home for easy to learn Asterisk stuff. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've read alot of comments on the SPA-3000, many if not all saying they had echo issues, but I've not seen anyone comment on the SPA-3102. Does anyone have any comments or issues with these? Tim

Hi all, I have a problem with the dialing tone in PAP2: When making a call, I can hear the calling tone 5 times and then it stops. The called party still hears the call but not the calling party. I've playing around with different parameters on the PAP2 web config with no success until now. Anyone has seen the same probelm? Thanks, Jose

Hi, I have a queue with one agent added using AddQueueMember (FAO|Local/1001 at from-sip|0||Agent/602). My extensions.conf is [general] static=yes writeprotect=yes autofallthrough=no clearglobalvars=no priorityjumping=no [from-sip] exten => 100001000,1,Dial(SIP/100001000,,t) exten => 1001,1,Dial(SIP/1001,,t) exten => 1002,1,Dial(SIP/1002,,t) exten => 1003,1,Dial(SIP/1003,,t) exten

Hi, I have a Linksys PAP2-NA connectd to my asterisk. I would like the device to add 2 characters in front of the dialled number always when it send the call to my asterisk. I dont know how to do that. I will summarise my requirement. My friend dials 1-210-1234345, i want the asterisk to get 55-1-210-1234345. Can someone help me to add this dialplan. Thanks in advance Dan -------------- next

Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install. Anyone can help me. Tnx Giordano -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.0/639 - Release Date: 18/01/2007 18.47 -------------- next part

Hello asterisk-users, There are .call files, with their own syntax, ant they works. But I have a problem. The voip-info.org says: "... If the call answers, connect it here ..." that means, if the called people picks up the phone, he/she hear ringing, until the "caller" picks up the phone. But what can I do, to connect the call before it answers, so the when the called

Hello Asterisk, This is only a test, because I can't start new thread in this list... -- Best regards, Gergo mailto:csibra at gmail.com

Hi, is anybody out there who can set the outgoing caller id on ISDN (CAPI or misdn) channels? I've tryed everything what I found in forums, os voip-info.com but no luck. I use a fritz card with CAPI in my first installation (1 BRI), and a hfc 4 port bri card with misdn on other. The first installation have p-t-mp configuration, the second one is p-t-p. Both configuration is EuroISDN in

As I did two years ago, "I'm posting a new thread with the "Top Posting" subject" rather than hijacking the "Paging for Praying" thread. Two questions: 1. Steve K: What do you mean by "/coat"? 2. How do we change rule #5? --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -------------- next part

Hi, The move to sRGB is nice, is there any interest in adding an interface to lcms, http://www.littlecms.com, to allow gamut matching? I can think of a lot of instances where I would like to render a figure as it would appear on my printer. This is probably best done as a separate package though, at least at first. Nicholas Martin Maechler wrote: >>>>>> "Paul" ==

I have achieved much of what I wanted to achieve using Wine, with one exception. I was hoping to be able to use Shockwave content, but despite installing Firefox and the Shockwave plugin, it does not work - it seems to stop after the promotional film before the actual requested content. I know there are "pay" solutions to this problem, but was wondering if anyone had found a free

Dear all, I tried to make a call with extensions.conf. exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN}) exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME}) exten=> _00[1-9].,102,Hangup But the 2 and 102 will not be executed. So I can get the correct answered time via 2. Is any idea about it? Is it the problem of my ZAP channel's configuration? My zapata.conf is as below: