similar to: one-waysilence during calls

I have a preoblem with my snom 320 phones. I have 5 snom phones installed and all of them have 5.2 firmware. All have same settings in the advanced panel. On 2 phones when I press the hold or transfer key nothing happens and * does not start the musiconhold. In the The hold and transfer keys are set as F_R and F_TRANSFER correctly as the others. Other snoms and gxp-2000 work ok. Any ideas?

Hi, Since a week ago I am trying to get chan_misdn working with asterisk 1.4.19, using HFC based ISDN card on Linux 2.6.22. My setup is done as detailed on wiki and FAQ. * mISDN and miSDNusers are 1.1.7.2, unpacked, build and installed. After installation and misdn-init, I have this: aragorn:root/pts/1: # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) ->

Hi, I have an asterisk box connected to a PRI LINE, some extensions are trunked by IAX to another box that's connected via ISDN BRI to a PBX. That's what's happening call comes in via PRI to the first box and is sent to the other box exten => _N.,1,Dial(IAX2/sip:pippo@trunk/*${EXTEN}) the other box rings a pbx (simulating an ISDN call from a BRI line) exten =>

Hi, I've configured a new Asterisk 1.6.1 box to replace an old Bristuffed 1.2 Asterisk. Since then, it happens that forwarded calls are not presented the way they used to be. It seems that now, some endpoints are displaying the original caller id (that's what I'm trying to achive), while some are displaying the redirecting number : so if A calls B, B forwards to C depending on where

Hi Folks, I've Asterisk Bristuffed up and running behind an Auerswald Commander Basic ISDN PBX on the internal ISDN Bus (BRI/PTMP). The HFC Card works marvelleous for outgoing calls (as the parallely installed avm fritzcard with chan_capi does), but when I'm trying to call in, I get a short ring signal and then the connection is terminated. This does not happen with chan_capi and

Hi We've recently set up and are using with success 1.0.7 using a Junghanns quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works very well, however we're getting cases where sometimes the call just drops. >From setting more verbose modes we get a log which is shown below. The problem seems to be the maxretries message which comes from chan_iax2. We are using

Hi, I have a 1.2.4 * box with two HFC modems using chan_modem_i4l and several SIP phones and ATA's. We have a terrible delay on calls between the PSTN (isdn BRI) and the SIP phones. All internal calls are fine. My first thought was that the transcoding could cause the delay but all of the SIP phones default to ulaw so there should not be any transcoding needed. I also checked the load

Hello Marc, Could you post the install details for the Diva PCI? I have a Diva Pro PCI 3.0 and I am looking to get it working... I am running Debian Testing. So far it appears there are a few things floating around but I am not certain which direction to go. I know I need to load the firmware first. I did manage to load the modules ok. Thanks, Greg -----Original Message----- From:

I'm trying to disable call waiting for Linksys SPA-941, but unfortunately as far as I have seen, there are no parameters on the web interface regarding this feature. I just want callers to hear the busy tone when the called party is at the phone. Probably I can accomplish this by using the "disable call waiting" in asterisk as well, but I have not been able to find any

Why is using ulaw or alaw an unlikely scenario? I wouldn't use anything but ulaw\alaw. The Bells can compete on price and will if they have to. Where they CAN'T compete is quality. If there were something better than 711, I'd offer that. Well, there is 722, but not many things support it. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original

Hi- Because a majority of my customers are in Europe, I've gotten quite used to working with ISDN (PRI) and BRI on a regular basis. Recently one of my customers asked me if I could terminate a few lines locally here in the USA (California), so I called up SBC to enquire as to how much it would cost to install a BRI here. Although the rates were reasonable (except the installation), I got

Hi list, I installed 2 HB8 cards each of them with a Quad Bri modules in a HP 360 G6 running Debian Squeeze. Here is an output of dmesg wafter server has booted: [ 9.784123] wctdm24xxp 0000:0b:08.0: PCI INT A -> GSI 31 (level, low) -> IRQ 31 [ 11.847073] bnx2: eth0 NIC Copper Link is Up, 1000 Mbps full duplex [ 11.847600] ADDRCONF(NETDEV_CHANGE): eth0: link becomes ready [

Hi all, finally yesterday I was able to set up my Asterisk machine and now it works greatly. Assuming I use Asterisk also to manage a BRI via a HFC based ISDN card, I can dial outgoing calls via ISDN but I still can't answer incoming calls from that BRI. I read that special extension 's' is for analog only. What do I have to put in extensions.conf to make Asterisk answer incoming

I have a question, I have to provide a solution for an office that will be almost abandoned, and there will be one or sometimes two persons 2 days a week. The main number however should be preserved. They have several ISDN BRI connections, most of which will be dropped. Only one will be retained, for 2 reasons: 1) It has the ADSL link 2) The number has been the main contact number for over 20

Hello, I'm having a problem with significant audio dropouts occurring in voicemail messages left via an ISDN-BRI trunk. Dropout durations are as short as 15ms and as long as 200-300ms. The audio that is recorded, appears to be otherwise complete, just with frequent holes punched in it. The same trunk has no problems with audio files played toward it from voicemail, nor interacting with

Hi All, I have setup 2 asterisk servers using AAH 2.8. I have configured a IAX2 trunk between the 2 servers using the guide on dumbme. Trunk is not using register string and no authentication. In my dial plan I have 7XX numers on server B and 6xx numbers on Server A. Calls from my SNOM phones are ok between the extensions on the 2 servers. In server A I have a eicon 4 port BRI card connected.

I'm trying the latest bri 0.1.0 RC2 drivers. In announce I see implementation of so long waited Transfer feature. But I can't make it work. When the person who is making transfer after talking with second party press "R" second time to establish 3 way call the person to which call supposed to be transfered being disconnected. Any ideas whats wrong? Thanks, Dmitry

We have quadBRI configured 1 port in TE mode 2,3,4 ports in NE mode. We are trying to place a call from the phone connected to BRI card port #4 to city number through ISDN line connected to port #1. Number successfully dialed. Person on the other end answering the line. But conversation can't last more then 10 seconds. Below is a log of such call. Its not clear for me why we appear in

We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This should get rid of static on the line (at least any static generated by our half of the circuit), right? I am a total virgin to ISDN. I understand that we need two BRI circuits to provide four voice channels, and that the hardware to speak to the BRI circuits can be passive or active, with the active type being much more

Hi all, Originally tried to post this without being subscribed, apologies if the list gets this twice. One of my users has a problem with many of his calls via my Asterisk? server. He describes the problem as having the sound slowly fade out and then fade back at a regular frequency. Has anyone experienced this before? Can you suggest a potential cause? The client is an IAX2 soft