similar to: avm fritz 2.6

Hi all, I've been searching the wiki and google for a couple of days now but cannot find any reference to a timing source on OpenBSD. I have * CVS-v1-0-02/15/05-21:54:52 (I always do a cvs -q up -Pd before compiling) running like a charm on OpenBSD 3.6 Now I want to setup some IAX trunks to work and 3 friends and some meetme rooms but it looks like I need a zaptel timing source. Anyone can

Hi, I read a lot about US providers that can terminate a PSTN number for you and offer IAX or SIP connectivity. Does anyone know such a company in The Netherlands ? I read about Unet. Anyone with experience with them ? Any information is welcome. -- Michiel van Baak http://lunteren.vanbaak.info michiel@vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

Hi all, I been searching on the wiki and google for ENUM in NL. All I could find were some docs from the Dutch Financial Department about taskforces and plans. But it all links to dead pages and no-longer-connected phone numbers. Is there anyone who knows some more about Dutch ENUM stuff ? If it's all dead, anyone interested in setting it up together with me ? Hope to get some answers,

Hi all, My BRIstuffed 0.2.0-RC7k is running fine on my debian box for voice calls over ISDN2. Now I want to implement receiving incoming faxes into my setup so I did a google and some reading on the wiki. I got the spandsp 0.0.2pre10 package compiled and installed, patched asterisk's apps makefile and compiled * again. This all worked out fine. When integrating the RxFax into my dialplan the

Hi, If it is really true that the voip-info.org website is hosted on a DSL connection without static ip, I have a server in managed.com datacenter that can host it. I still have some ip's free, so tell me if you want to use it. Bandwidth will be on my cost the first terabyte every month. Server has plenty of space left on the HD. I offer this for free, heck, I even offer mail domain with it!

Hi, Anyone here using sipgate.de ? It worked for months, but for a couple of days now I'm unable to register with them. My account is ok, because I can login to the website. Asterisk keeps showing me: Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n) I looked at the sip debug stuff, and all I

Heya list, I need some advice/experience. Some of our customers are asking us about DECT solutions for their asterisk install. Some others will not go to asterisk if there won't be a DECT solution. They now have a Siemens or a Samsung PBX. Those PBX-es come with a DECT basestation and optionally repeaters etc. All those basestations speak some own protocol to the PBX, so we cannot use them

MD -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michiel van Baak Sent: September 12, 2008 2:08 PM To: Asterisk Users List Subject: Re: [asterisk-users] Setup speed dials on Cisco 7921 On 13:15, Fri 12 Sep 08, OCG Technical Support wrote: > I've added lines like this: > > > >

Hey, We have asterisk 1.2.4 (old I know) with a couple of snom phones, a couple of grandstream phones and around 65 philips dect stations. Now the problem: All calls do peer to peer RTP except the calls from dect station to dect station. snom to dect or dect to snom do peer to peer. So the sip config looks fine because all the 'static deskphones' honor the REINVITE and start talking to

Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- Michiel van Baak michiel at vanbaak.eu http://michiel.vanbaak.eu GnuPG key:

Hi, Anyone here has any experience with the Nokia E70 and asterisk ? I read on the nokia website this phone is capable of talking SIP and do Presence based on SIP/SIMPLE. Please share your experience, I'm thinking of getting one but want to be sure I can use it with * before I do. Thnx. -- Michiel van Baak michiel@vanbaak.eu http://michiel.vanbaak.eu GnuPG key:

> -----Original Message----- > From: Michiel van Baak [mailto:michiel@vanbaak.info] > Sent: 26 May 2005 20:22 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] CRM integration (was RE: CallerID) > > > Anton, > > My script is not connecting to the manager interface. > The php script is run as agi script as first when a call > comes in. The

Tzafrir Cohen already mentioned it in a reply to someone else: There's a bristuff-users mailinglist now. If you are interested in bristuff or are using it consider subscribing to it. The webinterface is here: http://lists.three-dimensional.net/mailman/listinfo/bristuff-users The list address is: bristuff-users@lists.three-dimensional.net This mailserver is running greylisting, so you might

I would like to hear stories from people using TAPI, CTI or CallerID software with Asterisk. What are you guys using, setup examples, etc. Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do it. Are you running callerid software? Did you stumble into problems like using tapi and callerid software returned both the callerid and calledid? Hope you can help me out with

Does anyone know if the 7912G (which the wiki says can do either sccp or sip) uses the 7940/60 sip firmware? I ask this because the only firmware I can seem to find on TAC for the 7912G is sccp, no sip...if it takes it's own firmware and doesn't use 7940/60 firmware, can someone point me to the right location for it? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone:

Hello! I'm curently using * with two old Teles PCI card (wich, btw, were hard to install and make good use of) with ISDN4Linux. The sound quality is simply perfect. However both dialing in and out through the ISDN line, there seems to be a _little_ bit of echo that eventually gets on your nerves ! Also the echo seems to get a _little_ bigger after a minute or so into the conversation. Now,

I would like to have asterisk pass along the caller ID phone number to a database server on a my local network (the same network that the * server resides on ) so that our customer service app. can pull up customer data automatially. Asterisk passes along caller ID to the phones fine, can someone tell me how to make it pass this info to my database server? Any suggestions would be greatly

Can any one tell me how to install CAPI for Fritz cards on a Mandrake 9.2 system. I have been working through http://www.voip-info.org/tiki-index.php?page=Asterisk%20How%20to%20conne ct%20with%20CAPI and just cant seem to get it working. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I wonder which PRI interface card is most stable and supported for EuroISDN and Asterisk ? Are they stable enough ? Any tips ? Thanks in advance, regards, Rob.