similar to: OT - S450ip and R-key transfer

2008/10/5 robert.boardman at gmail.com <robert.boardman at gmail.com> > Kevin P. Fleming wrote: > > Olivier wrote: > > > > > >> 2. R Hook-flash key is now available to transfer calls. > >> In s450IP web management server, its defaults settings are : > >> Application-type: dtmf-relay > >> Application-signal: 16 > >> >

Hi, My Gigaset S450ip allows 2 simulatneous calls when each incoming call are targeted to different phones. When both calls target the same extension, the second one is forwarded to voicemail. I couldn't check yet SIP messages but has anyone met this limitation (one simultaneous call per phone) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Leg/Transaction Does Not Exist" and obviously not taken into account as endpoint GUI remains unchanged. Looking deeper into this here are : NOTIFY message accepted by S450IP NOTIFY sip:7531 at 192.168.100.197:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: "asterisk" <sip:asterisk at asterisk>;tag=as4ea953db To: <sip:sip:7531 at

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, With my Gigaset C470IP (with latest 02223 firmware), I can't find a way to access SMS settings from web configuration app or using a handset. Has someone been more successful without using auto-configuration mode ? (For instance, manual says an SMS entry is showing on handset screen but as I plugged my base station into a private LAN, I skipped the whole auto-configuration process ).

Hi there, I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream ATA's. The problem is that with both of these devices the Unattended call transfer process seems to be just like Attended but instead you hang up as soon as you have dialled the number of the party your are transferring to. The call transfer all works fine BUT as you complete your side of the transfer

Hello, We have an application where operators will sometimes take an incoming call from a queue, then contact an outside line, do a consultation, and finally do a SIP attended transfer to join the two parties together. We'd like to record the incoming caller's conversation with the operator and the attended part of the outgoing call, but not the unattended part, after the transfer has

Hello all, I am running into some problems with a snom 105 phone trying to do a attended transfer . Snom phones are connected to Asterisk. This does not work, it will only do a unattended transfer. I have downloaded the manual from snom and followed the instructions. Has anyone experienced the same problem ? any ideas how to solve the problem. thanks, Arne.

Hello list, whenever I press the #-key I hear a voice saying 'transfer'. How can I use the #-key without this voice-message or without having it the function of unattended transfer ?! The T or t option is not set in my Dial()-command so I don't know where this transfer is coming from in the first place. Kind regards, Jonas. -------------- next part -------------- An HTML

Hello, when a call comes in and is answered by colleague A, this colleague A sees the CallerID of the external calling number. When colleague A transfers the call to colleague B, attended or unattended, then colleague B sees the number of colleague A on his screen while talking to the external calling number. I expect here that colleague B would see the external calling number on the screen

hello, I would to define and unattended process to configure devices which are http-server-enabled, use DHCP but do not use TFTP-DCHP to configure themselves during boot. Has anyone worked on such subject ? I was thinking of something like : populating configuration file from device web pages (rendering this as generic and flexible as possible) writing a script which reads this file and set

Hi there, I just upgraded a relatively old Asterisk installation (1.2) in our office to a relatively new version (1.6svn from last wednesday) which runs a Junghans QuadBRI card [1]. To get this flying I got dahdi-linux, dahdi-tools and libpri from SVN as well. After a while of juggling it "works". What doesn't work: connected ISDN devices (Gigaset phones connected to QuadBRI

Hello using Asterisk 1.8.32.3. What is the best way of knowing a call is being transfered (attended and unattended) ? And also knowing whereto (sip user) the call is being transfered and who is the transferer ? So I can log this information. Kind regards. J. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ok, I've seen discussion before on doing transfers (attended and unattended), there seems to be much confusion over it. As things sit, I've been trying (unsuccessfully) to do transfers with a zap channel (analog phone attached to TDM400). I have no idea what I'm missing. My current understanding is that I need to have transfer=yes in zapata.conf, do a flash hook, dial the 2nd

Hi @all, i have a foreman-proxy server, build from scratch, works fine and i can build unattended hosts. I don''t want to configure all my foreman-proxys manually, so i build them in puppet, and only setup the OS (SL) and basic puppet config manually. I can run the puppet configuration sucsessfully, my config is exactly what i want, but i am unable to build unattended hosts anymore,

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

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.

Hello, some problems with transfer and zap... one hfc-card in NT mode and one fritz isdn-card in server. there is one gigaset SX353 isdn phone on the hfc-card. anybody calls from external via capi and the call is bridged to the zap-device. if you want to transfer the call via R-button on the isdn-phone the caller get the music-on-hold. you get a dialtone and dial - if the called person gets on

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 -