search for: klaverstyn

...lwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 channel=>1-10 callgroup=1 pickupgroup=1 immediate=no David Klaverstyn Systems Administrator Information Services, Asia-Pacific Intergraph Corporation Level 3, 299 Coronation Drive Milton, QLD 4064 AU P 61.7.3510.8951 F 61.7.3510.8901 david.klaverstyn at intergraph.com <mailto:david.klaverstyn at intergraph.com> , www.intergraph.com.au <http://www.intergraph...

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct.

...ify=yes host=sip.gotalk.com disallow=all allow=g729 ;GoTalk Inbound [09xxxxxx] username=09xxxxxx type=user secret=xxxxxxxx fromuser=09xxxxxx host=sip.gotalk.com context=from-vsp canredirect=no Registration string is register=09xxxxxx:xxxxxxxx at sip.gotalk.com/09xxxxxx David Klaverstyn Systems Administrator Information Services, Asia-Pacific Intergraph Corporation Level 3, 299 Coronation Drive Milton, QLD 4064 AU P 61.7.3510.8951 F 61.7.3510.8901 david.klaverstyn at intergraph.com <mailto:david.klaverstyn at intergraph.com> , www.intergraph.com.au <http://www.intergraph...

...all from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David Klaverstyn -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130219/7c9d1d62/attachment.htm>

Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm not sure if I have a configuration problem or not. I am unable to dial out. When I try to dial in I can hear the phone ring on the dialling phone but Asterisk does not register anything. In zaptel.conf I have loadzone = au defaultzone=au fxsks=1 In zapata.conf language=au context=from-pstn When I do: zap show channels I get: Chan Extension Context Language

I hope this doesn't already exist, but I couldn't find anything to help. I am installing a brand new Asterisk server, and want to use the Polycom IP331 phones. Does anyone have any steps on how to configure these? I have softphones working just fine, but for some reason I can't find a clear step by step on provisioning the Polycoms. Any help is greatly appreciated! Mark J.

Hi All, I'm in the process of writing an install script and I would like to change some settings for the install process but I don't want the user to go into menuselect and make the changes manually. Is there a way to make the changes to menuselect from the CLI? As an example, selecting the iLBC codec. menuselect codec ilbc on Regards David. -------------- next part -------------- An

Hi All, I have installed newt and newt_devel but dahdi_tool will not compile/install. I'm trying this with dahdi-linux-complete-2.5.0.2+2.5.0.2. does anyone have any suggestions as to what I am doing wrong? Regards David. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. -------------- next part -------------- An HTML attachment was

All, I have a Polycom 650 phone, when turned on displays "Checking application". Can any give me some information as to what is wrong? I have copied the CFG files from a 601 phone to work with this 650. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070429/daadda2a/attachment.htm

Hi All, Where can I find copies of the app_rxfax.c, app_txfax.c and apps_Makefile.patch. They don't seem to be located at soft-switch.org anymore. I am currently trying to compile Asterisk 1.2.26.1 and need the fax components. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, How can I force an anonymous SIP connection from a certain IP address to use a specific context rather than the default one defined in sip.conf. I am using Asterisk 1.6.0.9 Regards David Klaverstyn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090619/12c2472c/attachment.htm

What is everyone using in Asterisk 1.6.x to retrieve data from LDAP. The version of app_ldap I have only works with Asterisk 1.4.x http://www.mezzo.net/asterisk/app_ldap-2.0rc1.tgz Without a way to get data from LDAP stop me from using Asterisk 1.6.x Regards David. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

Hi All, I'm hoping someone can recommend a method to integrate Microsoft CRM with Asterisk. Preferably an open source product otherwise a commercial product. Regards David Klaverstyn -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130716/f931d763/attachment.htm>

The Telco in Canada is been real painful. I was wondering if anyone has installed a Digium TE1X0P card in Canada and if their Telco was so difficult. The Telco will not provide us a service until they see a FCC or DOC number for the equipment ware are connecting to their service. If have found "FCC Part 68, ANSI/ITA-968-A, Including Amendment A1 and A2 Industry Canada CS-03"

Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI <--> Asterisk <--> T.38 <--> ATA <--> Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt

Can someone please help me with a problem that I seem to have with this Polycom 601 phone. It will not see my TFTP server and keeps saying "Could not contact boot server, using existing configuration". I have Linksys phones that use the TFTP server without any problems but this Polycom will not see or use it. Please Help. -------------- next part -------------- An HTML attachment