similar to: Noises on Batphones

Hello! I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950 server with 8 gigs of RAM. The 2 other ports on the R4T1 go to our 2 PRIs. In this configuration, we have trouble maintaining stability. It may be fine

The TDM410P doesn't support 'hvac', only the obsolete TDM400P supports that option was for the old phones that have a neon light (or equivalent LED+ZENER ciruit). Are other phones off the TDM410P (other than the VTECH) working, or is the Vtech the only model with VMWI available to you. I'm not able to check at the moment, I have copied the asterisk-users list, someone else may

Hi, Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] & same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry

Hi, Is it possible to create a clear zaptel channel which doesn't require to be picked up? The requirement of my client is to open a clear channel to a recorder which starts recording certain message. Currently the channel which is created by zaptel requires the other end to answer the call, and the recorded can't answer, so the channel get hung up after a certain number of rings. Zaptel

I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they could share? I have tried the version at voip-info<info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP&view_comment_id=14768#Troubleshooting>for the 7941/7961 but unfortunately /var/log/messages shows in.tftp stops sending after

Guys. Anybody using sipuras 2002 knows if there is a way to make the phones connected to it to autodial an extension when the phone is picked up? For example, if the phone is on a police booth (building entrance) and you want the guys to just pick up the phone and make the phone auto dial the receptionist extension without the guys having to dial anything (ala batphone). Is this possible with

The Asterisk Development Team is pleased to announce the first release of DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0. 2.6.0 is a feature release which: - Adds support for the TE820 8-span card to the wct4xxp driver. - Decrease load time of analog cards supported by the wctdm24xxp driver. - Adds sysfs object model to facilitate persistent span numbering and early loading of modules

The Asterisk Development Team is pleased to announce the first release of DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0. 2.6.0 is a feature release which: - Adds support for the TE820 8-span card to the wct4xxp driver. - Decrease load time of analog cards supported by the wctdm24xxp driver. - Adds sysfs object model to facilitate persistent span numbering and early loading of modules

My setup is this: Analogue phone attached to a Linksys PAP2 | Asterisk | VoIP provider I have put the PAP2 in 'batphone' mode where when you pick it up it immediately dials the 's' extension in the pap2_incoming context in Asterisk, where asterisk answers the call and does a DISA(no-password, internal). I do this because it means I can centralise all of my dialplan logic in

Does anyone know of a way to make a standard analog phone plugged into an FXS port do something other than get a dialtone when you pick it up? For example, if the phone should automatically ring someone or play a greeting when picked up without having to enter an extension? - Robert -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

How can I let asterisk immediately dials a trunk when off hook?

Hello For users who 1) don't have a QoS-capable ADSL router and 2) would like to run Asterisk with a couple of SIP trunks, I was wondering what hardware is recommend to run any of the main open-source *WRT projects to which Asterisk has been ported: (http://en.wikipedia.org/wiki/List_of_wireless_router_firmware_projects Thank you.

This might be a newbie question... I'm using a SIP trunk and trying to get DID line information on an incoming call. All I hear is a nice lady saying 'Zero' - then the call continues... Any suggestions? thanks Todd exten => s,n,Set(DIDID=(<${FROM_DID}>)) exten => s,n,SayNumber(DIDID) or exten => s,n,Set(FROM_DID=${EXTEN}) exten =>

My analog card, uses a PCI slot and a 12V power connector, is configured with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS ports but I can't dial out from them. Is extensions.conf where I need to make changes? [root at robin asterisk]# cat chan_dahdi.conf [trunkgroups] [channels] [phone](!) usecallerid = yes hidecallerid = no callwaiting = yes usecallingpres = yes

Our smallest network of systems has only four computers connected via Gigabit Ethernet.? The oldest and most stable platform is an eight year old Dell E520 running CentOS 6.8.? We often try out applications on this Dell/CentOS machine before moving them to other systems on our other networks. Last night, one of our users decided to create a single, 228GB home directory tar archive on an empty,

Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after

Hi all, I''m trying to install Xen on a dell E520 computer. I''ve a network problem while rebooting on kernel vmlinuz-2.6.16.29-xen, all do work well, dom0 (ubuntu 6.06) start but I have no network at all. When booting on ubuntu, the NIC which is a Intel Gigabit Ethernet card drived by the e1000 modules does work correctly, but when the kernel xen is booted, trying to use the

I just saw the release info for the latest SYSLINUX and improved USB keyboard support. Checking the mailing list archives it looks like you don't have access to hardware with the problem. I have a DELL E520 (amusing enough one which was shipped with Ubuntu preinstalled) with which I've had SYSLINUX/USB keyboard problems. The 'basic' system works, but when it tries to chain? into