similar to: On multiple dial phones continue ringing after picked up

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

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 -------------- An HTML attachment was scrubbed... URL:

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 SNOM phone.

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

Well I dont want it to operate as a group. I understand that it is *8 on most PBX systems but I would like it to work as dialing *8+extension-thats-ringing to have it go over to my extension. I am having a lot of trouble finding more information from the wiki and google. Thanks! -James -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 10000-20000. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side,

I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco spa8800, all them are internal lines. 1.- spa921, 401 ext 2.- spa921, 402 ext 3.- normal phone connected to spa8800 404 ext. It had a very strange behavior when I was configuring call transfer and call pickup. These are steps to repeat it: 1.- from 401 call to 404 2.- from 404 don't answer it. 3.- from 402 press *8

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

Dear all I have asterisk with TDM808B FXO port with i call in asterisk and i promt IVR then user dial extention for user then my SIP phone rining but i disconnect or hangup my mobile phone but still SIP phone rining and stop rining after timeout is there any PSTN problme or FXO signalling problme i have configuraed singalling=fxs_ks ----PGP Signature-- Satish

Hi. Again one big mysterious problem I hope some good guy can help me solve. I'm trying to connect some Siemens C450 SIP IP Dect phones to asterisk (1.2.13) (I have actually 3 handsets + 3 ip base). After configuring them 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

Dear all I have TDM808B 8 port FXO it is configure perfectly but i got some problem of incomming phone Hangup and callerid display problem i am going to explain you the issue i have install asterisk 1.4 and i have 100 of SIP phone now everything is fine but problem is when i incoming call on FXO and dial sip extention SIP phone is rining but when i disconnect my incoming

Hi, It seems to me that DECT Gigasets do not support MWI when connected to a Polycom/Kirk DECT base station : when a new message is dropped into user's mailbox, I can see a NOTIFY message sent by Asterisk to Polycom/Kirk DECT base station (here a KWS300) but Gigaset's handset MWI remains unlit. At the same time, a Polycom/Kirk handset subscribed to the same base station would turn its MWI

Hi list, Before purchasing a number of Siemens DECT SIP phones, the Gigaset S675 IP, I read that the problems with MWI had been fixed with the latest firmware version (see http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm not so sure that's the case. After setting up a network mailbox for one of these phones, as well as an Asterisk voicemail account (ext.

hi list. after my unsuccessfull experiences with mISDN i tried again to implement a zaphfc based solution. problem is: sound on calls via capi is stuttering/broken and therefore unuseable. my conf: - cel 1300, 256mb ram - avm b1 via capi connected to my outgoing ISDN - acer surf pci via zaphafc and crosslink cable with termination as internal bus - siemens gigaset 3035 as internal phone -

Hi, Here is the configuration of the server that I currently have extension 100 (SIP) =>(SIP)asterisk server 1.8.18(IAX trunk) <===>(IAX trunk)asterisk server 1.4.32(SIP) ===> SIP Providers The issue is while dialing out from extension 100(sip) if the providers sends back 180 Rining the SIP extension(100) won't hear the ringback tone, where as if the providers send 183

Hi, Does anyone have any advice on customising firmware of an SPA921 so that it can be locked to a sip provider and display logos on the config pages. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f6aa2510/attachment.htm

When you configure the Siemens gigaset handsets (I have S685IP), there is a single option for all handsets to use either the POTS interface or VOIP as the default outbound destination - you then need to add a dial suffix if you want to use an alternate outbound route. Does anyone have any suggestions as to how to make just *one* of the DECT handsets only use the POTS but others default to

Hi, We have SPA921 handsets which apparently support Paging, however i can't find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm having no end of trouble with some Siemens Gigaset phones and GS HT286s. Gigaset 100 and 3010 phones work perfectly, but a 4010 only rings once then it goes off and then just flashes it's LEDs and displays "incoming call" on the LCD with no further ringing. According to the manual it is CTR37 but the only setting on the GSs is CTR21, I've tried different cables but some

I've just ordered a Siemens Gigaset C450 IP cordless IP/DECT phone, given that it's supported by asterisk http://www.voipuser.org/review_41.html However, I see that a slightly better Gigaset S450 IP is available for only a slight price premium. Are there any user experiences with the S450 IP? -- Eugen* Leitl <a href="http://leitl.org">leitl</a> http://leitl.org