search for: c450ip

Hi list, I am faced by a situation where I am trying to make a softphone and a Siemens C450IP talk to each other. Both are hooked up directly to the same asterisk, in the same IP net. - a softphone runs on 192.168.14.3 - the C450IP is at 192.168.14.30 - asterisk runs on the machine known as 192.168.14.1 I am running Asterisk 1.4.11, backported to Debian Etch by Xorcom. If I set ca...

Hi, I'm using Asterisk 1.2.18 on a Debian Etch box. I've noticed a very strange fact which causes a bad prob. When I get an inbound call, I make 4 phones ring at the same time, one is a Snom while others are Gigaset C450IP with _latest firmware_. When I get a call and answer with the Gigaset, a second call going to voicemail makes the first call received on the gigaset C450IP stop audio (it seems the call does not drop). This does not happen if I answer the first call with the snom or with Siemens using older firm...

...and rebooting, all of them register properly on asterisk, then, after the first call, they appear no more registered as registered in asterisk, and on the handset the display shows "SIP registration failed". Has anyone got the same problem? Has anyone ever tried to operate more than one C450IP in the same open space? fyi: - there are no lan/ethernet problems - I've tried with different "qualify" values (0, 2000, 5000) - no nat involved - everything is running on a private lan - phones reply to ping even when qualify times out - any other sip phone (wired to the lan) works...

Hi all. We have Snom phones which do have a defined key in order to drop incoming call WITHOUT answering. Pressing that key, a "SIP/2.0 486 Busy Here" message is sent back. We have other phones (I.E. DECT Siemens C450IP, or ATCOM 320 or other) which DO NOT have any key to do that (or the key does not work, as is with Siemens C450 IP ): you have to answer and immediatly after hangup the call. Acting on feature.conf we succed in defining keys for blind transfer or attended transfer: the last thing we need is the ab...

Hello, My goal is to provision C450IP or S450IP models. Has anyone a hint to provision them from configuration files ? Usually, we use dedicated menu embedded inside Gigaset handset. An http server also exists but I couldn't find any dhcp-tftp combination to configure them. Any clue ? Regards -------------- next part -----------...

...I use this format : Dial(SIP/4029&SIP/4030,15,tTr) As soon as one of the phone is picked up, all the others should just stop rining. But the fact is that they continue to ring for several seconds (4-5s), and this is quite annoying as all phones are in the same room. Phones are Siemens Gigaset C450IP and Linksys SPA921. Do you have any idea what I could do to fix this ? Regards, Yves.

Hi, I'm using Asterisk 1.2.18 on a Debian Etch box. I have two phones a Gigaset C450IP and a Snom 360. Suppose someone is calling the Gigaset phone and a second call comes and is redirected to the voicemail: if the new caller hangs up during voicemail announcement, Asterisk drops the first call. This does not happen if the first called party answers the incoming call using the SN...

I'm going to be rolling out asterisk at a small office and one requested feature was the ability to have a phone that can be configured so that ringtones can be configured according to the callerid of the caller. Does anyone have Asterisk experience with such a phone? Any suggestions would be greatly appreciated. Thanks in advance!!!

Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and doesn't cost $300? There seem to be a plethora of decent and affordable corded phones (like from Grandstream) but the search for a cordless unit seems elusive. I purchased a vtech 8100 online only to discover after receiving it that it is locked to vonage service. Thank you.