similar to: Gigaset R630H and Asterisk

Hi everyone, Does anyone have the documentation for OutCall? http://code.google.com/p/outcall/ The link isn't working. Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150

Hi, Does anyone have an up to date guide for setting up fax 2 email with asterisk? Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150 per month, please contact Kesher

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

a phone system, and plug it into a SIP Adapter like the PAP2T. Never done it myself, so I can't recommend a suitable intercom. Hopefully s= omeone else can. Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<h= ttp://www.keshercommunications.com/hostedpbx.html>

Hello, I was wondering if anyone has any insights on the best way to automatically monitor an asterisk box to check it is constantly available and processing calls. Many thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more

Hi, I'm trying to work out an issue with call queues. I need the calls that are in a queue to be kicked out if all members are unavailable (for example if all SIP members are having network problems). I tried leavewhenempty = yes but that only seems works when all queue members specifically log out of a queue. I've looked at autopause, but we need it to automatically un-pause once it

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help........................ On 4/8/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at

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.

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

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

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 > >> >

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

Hi, Searching this list archives, I couldn't find a definitive answer to my question : how to send SMS to Gigaset phones ? My goal is to send Alert SMS such as "This phone system will be stopped in 5mn for maintenance" to every terminal (SIP phones and Gigaset DECT phones). (So at the moment, I'm not looking for way to send SMS from handsets). I could successfully send 1 short

Hi, I know this is a little OT but there are many Asterisk users of the excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is probably newsworthy for them. One of the biggest bug bears has been no mute function on the handset. When I woke up this morning, the handset told me there was a firmware update. I updated and then visited the web site to find out what had been

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:

Hy all of you, Is anybody has a tutorial for integrate a siemens gigaset as180 and connect it to Asterisk. I've searched a lot and didn't found something concluding. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110611/17522d7a/attachment.htm>

Hi all, I am having a problem with ChanIsAvail. It always returns the same result, regardless of whether an extension is available or not. It always returns 0 Unknown Status. This is my dialplan. exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s) exten => _2XX,2,Verbose(0, ${AVAILSTATUS}) exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5) exten =>

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: