similar to: Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.

Thanks. I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again - but only one direction. -David ________________________________ From: asterisk-users-bounces@lists.digium.com

hi, Longing for your help. I came into a problem ,Now I want to configure asterisk sip peers from MYSQL database dynamic, flolling the introduction of asterisk realtime,i set the cofiguration of sip users,but I need to cofigure sip peers too. Where I can find some infomation about cofiguring sip peers? What is the difference of configuration sip peers and configuration sip

On Friday, March 31, 2006 3:52 PM Chris Earle wrote: > I'm wondering if I should be using zapHFC with my Junghanns card > instead of qozap? Why would you want to do that? Sorry if I missed the start of the problem but qozap is what you want to use with your Junghanns card (at least if you want to stay with zaptel/libpri). > Everyone always mentions zaphfc -- mostly I >

> Appreciate the replies everyone -- really > > I'm wondering if I should be using zapHFC with my Junghanns card instead > of > qozap? Everyone always mentions zaphfc -- mostly I guessed because they > are > using a zaphfc-compatible card ----- but *maybe* I should try that instead > of qozap??? > > And yep -- totally know about the module load order thing and

Hey, anyone know if it's possible to receive faxes through a Junghanns bristuff quadbri card? In germany, currently I have faxes coming in on DID line into QuadBRI and then passing to Digium TDM400 (analog) and into faxmachine. But the reliability of TDM card is spotty, so I want to maybe just accept faxes in on ISDN card and save on asterisk system ...? keeping digital signal strong ...

Hi Alessio Yes, the problems you report do seem similar to the issues I had. I found on the Dells that the audio prompts were very choppy and played slower than normal. Occasionally there would be 'bursts' oav a second or so of 'good' audio. I also suspected IRQ issues but the Dell Mobos had no way of adjusting them. Best thing is to try and get the card on its own unshared

Hi, I have debug off (debug level 0) why are the following lines showing up in '/var/log/asterisk/full' Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '765e20595817e9897b77cff23f821cc5@10.0.0.254' of Request 102: Match Found Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '4858cde16223cc0716e325921a8a0654@10.0.0.254' of Request

Hi, Reading from www.voip-info.org, i can see that "Junghann's chan_capi is now part of bristuff <http://www.voip-info.org/wiki/view/Bristuff> , as of version 0.3.0-pre". What does that really mean ? Shall I understand I can share a Junghanns QuadBRI board between 2 CAPI-enabled software (like a 0.3.0-pre bristuffed Asterisk for instance) ? If positive, what are the pros and

As usual, we'll be jawing about any and all asterisk-related subjects with the usual gang and any new people are always welcome, regardless of your level of expertise. You can even come and ask questions, it's guaranteed to be a more pleasant experience than it will be on IRC ;) http://VoipUsersConference.org/topics.php IRC; Freenode.net #voip-users-conference

I have set up a * box with two hfc ISDN pci cards using mISDN both in TE mode with PmP mode. (using $MODPROBE hfcpci protocol=0x2,0x2 layermask=0xf,0xf) I have no problem dialing out by explicitly naming the mISDN port, ex: Dial(mISND/1/${EXTEN},60) or Dial(mISDN/2/${EXTEN},60) But it does NOT work when specifying the mISDN group: exten => _(outpattern),1,Dial(mISDN/g:TEmode/${EXTEN},60)

Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register => 092779077:XXXX at 85.119.188.3 This the output of SIP show peers : asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status

Actually it worked, but only after I defaulted all the settings on the phone and let it pick the config up fresh. Anyone know if there is any headset config options to default to headset/speaker? Thanks Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lee Archer Sent: 03 January 2006 14:49 To:

Hi. I'm trying to get a Beronet QuadBRI card work with bristuff drivers. Though qozap module loads right, all card spans are in deactivated status. I'm quite sure my configuration is correct and using a single BRI card instead of the quadBRI the status is active and I can place and receive calls. On Beronet installation manual I read that Beronet and Junghanns cards are identical in

The HAVE_MMAP_FILE macro comes from when you configured llvm. The configure tested if your system is giving access to a various resources required by llvm. LLVM needs the MMAP functionality by your system. Some times, if the required resources aren't present, we use other resources. However, in this case the building process is stopped, because the functionality can't be implemented

Dear Mr,Ms; I am planing for a custom IVR, for example to act as a simple installer! I mean there is some choice via 0-9 and # as *Next* and * as *Back* button. is there any way for me to detect if the caller pressed # vs * on Dialplan ? -- Regards, Ali R. Taleghani 0936 322 4069 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Im using Asterisk 1.4.11 and I want to proceed some time and date operations in my dial plan. (for a time shifted callback). Should look like: CURRENT TIME + x minutes. Of course it should increase the hours for example in this case: 10.59 + 5 minutes = 11.04 I guess I've to use the math function in 1.4 but how can I manage easily the time operations? Kind Regards, Erik

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=GB18030" http-equiv="Content-Type"> <title></title> </head> <body bgcolor="#ffffff" text="#000000"> Hello,<br>  I built llvm1.4 with gcc3.4.3 and glibc2.3.2 on a redhat 9.0 machine,

Hi All; Is there an GUI for Asterisk that can help in showing the call flow (who is in progress, who is connected, called number, ...)? I was think in AsteriskNow does this? Any advise? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font size="-1"><font face="Bitstream Vera Sans"><br>     Hello,<br> <br>     I am trying to have dns <b>automatically</b> configured through

Hi, I'm testing/debuging the migration from mbox to maildir in a mailserver running Dovecot 1.0alpha5. I migrated all 472715 emails(in a test server) and later noted that dovecot could not get the same flags/status of the emails as in the mbox. I was using perfect_maildir.pl and I think its not dovecot fault. Seems to me that perfect_maildir.pl convert the emails in a way that dovecot could