search for: michiel

Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring.

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

...>_write($fp, "Context: ".$this->context); $this->_write($fp, "Exten: ".$this->whotocall); $this->_write($fp, "Priority: 1"); $this->_write($fp, ""); $this->_write($fp, "Action: Logoff"); where $this->user_device = "SCCP/michiel", $this->context = "default" and $this->whotocall = my mobile nr If I run this script my Cisco starts ringing, so far so good. If I pick up the Cisco, it gives me a ringing tone, so far so good. When I pick up my mobile, the Cisco keeps giving me a ringing tone, like it doesn&...

...troller: def do_revert page = Page.find(params[:id]) page.revert_to!(params[:version]) flash[:notice] = page.title redirect_to :action => 'edit', :id => params[:id] end When I look at the data in the object, nothing seems to have changed. Thanks in advance, -- Michiel Sikkes michiel.sikkes@gmail.com _______________________________________________ Rails mailing list Rails-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org http://lists.rubyonrails.org/mailman/listinfo/rails

...DN lines. We have one annoying problem tho. There is an audio delay on outgoing/incoming calls. The delay is 2 to 3 seconds. Internal calls from sip to sip is no problem at all. Any idea what is wrong with our setup ? If you need more info and/or config files please ask. Met vriendelijke groet, Michiel van Baak Terrazur

I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

...(outgoing-budgetphone) exten => _0XXXXXXXXX,2,SetCallerID(31557110304) exten => _0XXXXXXXXX,3,Dial(SIP/31557110304/${EXTEN}) exten => _0XXXXXXXXX,4,Congestion exten => _0XXXXXXXXX,104,Busy And this is wat I get on the CLI when I call my cellphone: -- Executing SetAccount("SIP/michiel-d5bd", "outgoing-budgetphone") in new stack -- Executing SetCallerID("SIP/michiel-d5bd", "31557110304") in new stack -- Executing Dial("SIP/michiel-d5bd", "SIP/31557110304/06XXXXXXXXX") in new stack -- Called 31557110304/06XXXXXXXXX...

...> etc... How can I access the array/object 'resultset' in my submit action in the survey controller? I've tried using @resultset in the 'submit' view or params[:resultset] in the 'submit' action, both seem to be nil. Hope someone can help me. Thanks in advance! -- Michiel Sikkes michiel.sikkes@gmail.com _______________________________________________ Rails mailing list Rails-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org http://lists.rubyonrails.org/mailman/listinfo/rails

...nd on the internet and found the Kirk gear. Anyone has any experience with them ? The website states they are recognized as Cisco 7970 in CCM. Does chan-sccp handel those Kirk emulated devices ? Is there any other solution like this out there that works with asterisk ? Thanks for your input, -- Michiel van Baak http://michiel.vanbaak.info michiel@vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?"

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998

I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian

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 "Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence."

Hi, when I use useradd of adduser in CentOS 3 or 4, the default shell a user gets /bin/bash. I want this to be changed to /bin/false for security reasons... How can I accomplish this? Thanks in advance for any help. -- Michiel

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

Hello ! I would like to get working a Fritz PCI card using chan_misdn operating in ptp mode. Afer compiling mISDN into the kernel and building chan_misdn Asterisk stops loading with : [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables

Hi, Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support roaming and handover and are these functions transparent for handset (then, these functions are implemented in DECT base stations) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080110/4254f602/attachment.htm

I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial

Does there exist a text file with all the 'standard' Asterisk voice messages? I'm planning to get them recorded in dutch, but need to know the exact text of each prompt... Michiel

...nning asterisk 1.0.3 on Debian Sid (official debian package). We have 2 fritz ISDN cards. All is working great. Till I called the bank. It rings one time and then gives me the congestion tone. Here is what I see on the CLI (phone nr obfuscated for privacy reasons): -- Executing Dial("SCCP/michiel-00000004", "Modem/g1:xxxxxxxxxx|50|Ttr") in new stack -- Called g1:0342426530 -- Asked to indicate '3' (Dialing) condition on channel SCCP/michiel-00000004 -- Current tone (36) is equiv to wanted tone (36). Ignoring. -- Modem[i4l]/ttyI3 is busy -- Hungup...

...e * 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 point me in the right direction ? Thanks -- Michiel van Baak http://lunteren.vanbaak.info michiel@vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence."