similar to: How to hookup to cell phone for outbound calls?

I'd like to add a device to my Asterisk server to leverage my cellular account. Does anyone on-list have experience with hardware gateways vs using cah_bluetooth and an old cell phone? I'm considering something like http://www.mobigater.com/index.php?p=5 Thanks, Michael -- Michael Graves mgraves<at>mstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:mjgraves at

Some light relief .... SNOM say "Please note that you will not be able to reach us by phone." http://www.theregister.co.uk/2008/03/13/dont_call_us/ regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com

For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP range in the U.S. I'm particularly interested in the Gigaset S685 IP. Since it's DECT 6.0, and there's an English (UK) version, I'm thinking it should work just fine, after dealing with the walwart issue (and maybe caller ID signalling). Anyone imported one from the UK and using it in the US? for how

Hi, Our operator frequently gets requests to transfer a call directly to voicemail in order for the caller to leave a message without disturbing the callee. Basicly, I'm looking for a blindxfer to vm. My first thought was to prepend a digit (eg 7) to the extension but this does not fit well with our dialplan. According to an article on voip-info.org Asterisk@Home appears to implement

Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937]

I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like

Hi, We have a few Aastra 480ci phones and we've noticed that in order to get the phone to receive a call, qualify must be = no. Apparently the Aastras do not respond to the qualify message (or respond in a way Asterisk doesn't understand) and Asterisk thinks the phone is unreachable. However, this now prevents MWI from working properly on the phones. Does anyone know how to get MWI

Hello, 1. Is it possible to install several SIP softphones on the same PC, have them registered to the same Asterisk server and attribute to each softphone a specific extension, ringtones or call forwarding rules ? 2. Is possible to do the same with SIP hardphones ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was

Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where the phone did not send rtp keepalives when on mute (resulting in disconnect from tech support hold and concalls) A side effect seems to be that Asterisk pops the following warning on the console... Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short Grandstream say they are not sure what it is but

Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes

Hi, I don't understand why a SIP hardphone can't provision itself using tftp. I'm very suspicious about my tftp daemon but I lack basic knowledge of Linux CLI to pinpoint what's going wrong and separate what belongs to SIP phone configuration from what comes from tftp server. What I would like to do is to add a given file in current /srv/tftp directory and test by hand that tftpd

Hello everybody Anyone, to know a gateway that works with nextel simm cards? I'm looking for them, in internet, but I did'n look. Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080129/0527ba52/attachment.htm

Hi I have a analog phone connected to my Gateway Handytone and registered to Asterisk 1.4 I have configured my HandyTone 488 (in the section FXS Port) for make and receive calls, however I can not transfer a call when it come via PSTN. But, when a call come from via IP I can transfer it. [phoneanalog] type=friend secret=XXXXXXX context=local nat=no qualify=yes host=dynamic dtmfmode=rfc2833

Is anyone having problems with Cisco's 2960/3560 LAN switch? Problems causing "retries exceeded" in Asterisk? Thanks

Given a choice, and a green-field site, would you a) Have a separate network (switches etc) for your data and phone b) Use the same network, but use VLAN's ?? What are the pro's and con's of each ? TIA Julian

1 of our customers records all phone calls and needs to be able to be played back via a searchable web app. I tried ARI but it is very limited. Anyone have any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070518/40621109/attachment.htm

On Fri, 2007-07-20 at 02:08 -0400, BSumrall wrote: > My dial plan of issues?.. > exten => s,1,Answer(60) > exten => s,2,Background(otherwise-press) > exten => s,1,Playback(digits/1) > exten => s,2,Goto(default,s,1) > exten => s,1,Playback(digits/2) > exten => s,2,Goto(default,s,1) I'm not sure why you have three different sets of priorities one and two

Hello, I feel like I understand how the dial plan works pretty well with one exception. It seems like queues are using the stdexen macro to ring the agents/extensions. Is this normal? Is there anyway to configure this differently? I realize this is a newbie question, but I have searched google/archives and haven't been able to find the answer. Thanks, Elliot --------------

Is the Cisco UC 500 able to integrate with Asterisk? Specifically does it work via SIP? Just curious, as the Cold Call Cisco sales rep had no clue what SIP even was, and this device looks interesting. ________________________________ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is