similar to: clever ways to "share" an extension between sip and fxs

Hello All, I have a need for a wireless solution and have been looking at the Aastra 57i CT phone that have the wireless handset with them. Aastra says they will cover "up to 300,000 square feet". I am finding this hard to accept. I was also wondering about the "secure WDCT cordless technology" Could this be a form of DECT? Any one using these that can shed some lite?

I upgraded from a perfectly working 1.6.2 asterisk installation (including fax via app_fax_digium) to 1.8.0 this evening. All my custom modules (including swift <thanks darren!>) are working fine except for fax. When a caller connects, asterisk switches to the fax context and hangs up the call. i've captured with: core set verbose 10 core set debug 10 fax set debug on sip

After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP calls are being destroyed after 1 minute and 20 seconds (80 seconds). Asterisk is sending a BYE message - I have no idea why. http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug. can anyone suggest how i can further deal with this? -- Jeremy Kister http://jeremy.kister.net./

just thought this was cute enough to pass along, https://www.youtube.com/watch?feature=player_detailpage&v=GLwct15X_3g#t=135 -- Jeremy Kister http://jeremy.kister.net./

Running Asterisk 1.6.2.11 on debian 5.0.6 with mpg123 1.4.3 when i start asterisk, i immediately see two mpg123 processes spawned which sit there forever. I can't imagine it's normal behavior, but if it is, please explain :) # /etc/init.d/asterisk stop stopping asterisk. #[...] # /etc/init.d/asterisk start starting asterisk. # psg aster root 14573 1 0 16:29 pts/2 00:00:00

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

Hi, I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a remote peer and an fxs phone gets connected and the remote peer hangsup, then asterisk sends the "Idle" state to notify the watcher before you hangup the fxs phone! Such a way if the user forgets to hangup the fxs phone (which is a cordless for example) then the operators will keep sending calls to him

Folks, I have a TDM400 with an FXS module. I'm having some trouble ringing some phones attached to the device. I have a real el cheapo handset that cost me about 9UKP. It works fine, I plug it into the Wildcard and call the channel and it rings. I have tried plugging in my BT Diverse 3016 cordless phone but it will not ring, although I can call out from it. I have also tried my GPO 332L

I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to a standard analog handset with caller id display (US caller ID). Although it appears that caller id information is coming into asterisk (it shows up in voicemail), I can not get it to display on the analog handset. Is there anything special I need to do to send the caller id info out the FXS port? I've tried a few

Hi Folks, Looking for feedback on the cordless phones with the Aastra 480i CT handset. Is voice quality comparable to standard consumer residential 2.4GHz cordless phones in the US$30 - $50 price range, or better/worse? How about handset and speakerphone quality for the main phone? Seems like there have been various big issues with firmware in past, but is it pretty stable now? Thanks,

I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is

With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} is [un]set in an odd way. for example consider: 999,1,Swift(some long message that you dont want to wait for|5000|5) 999,n,NoOp(DTMF: ${SWIFT_DTMF}) if while I am listening to the playback, i interrupt and dial: - "12345", SWIFT_DTMF is set to

I just bought an Aastra 480i CT for a client who needed cordless capabilities in their office. I'm trying to set up the base station and cordless handset in my office first. I'm able to connect the phone to my Asterisk box and make outgoing calls from either the base station or the handset - to extensions within my office as well as numbers outside the network. But I can't

I have a dilemma. I am trying to setup an asterisk setup for about 10 people, but with the ability to expand to 100s. We are looking into hardphones to use for our systems. Here are the requirments I am looking for: SIP Not POE Full duplex Speakerphone $100-$300 We would like to have the cordless and corded option if possible (at least for 3 of the 10 people). So here are the

using asterisk 11.6.0-rc1 i just converted my "nat=yes" to "nat=auto_force_rport,auto_comedia" I have my asterisk box on the same subnet as a cisco 1760 (vgw1). a few times per day, Asterisk thinks vgw1 is dead (by qualify/options). A 'sip reload' always fixes the problem. i left 'sip set debug peer vgw1' on the console. but i dont see what's

I have a Cisco 1760V with FXO ports hooked to POTS lines talking SIP to asterisk 1.8.15.0. imagining in extensions.conf: exten => 1,1,Dial(SIP/121) exten => 2,1,Dial(SIP/121&SIP/122) When a caller dials extension 2 /and/ I have trustrpid=yes generaterpid=yes sendrpid=yes in sip.conf and I use the pickup exten, the caller is disconnected. see:

I'm trying to figure out how calls are ending up in my default context (which should never happen). I've got a Cisco 1760V with a VIC-2FXO-M1/VIC-4FXS and 5 Cisco sip phones. When I make a call from one of the FXS ports on the 1760, the call goes into asterisk's default context instead of where i think i'm directing it. Can someone tell me what I have misconfigured? 1760

Hello! I'd like to connect regular analog phones (Cordless Panasonic, for instance) to Asterisk. What's the cheapest FXS device on the market that can be used with Asterisk? Where can I find it? It has to be the cheapest to avoid huge import taxes, since I'm planning to buy in bulk. I'm also looking for BRI ISDN cards (cheap and used, preferable) that can be used in NT mode so I

Are the cordless phones on the 480i CT from Aastra registered independently in Asterisk? Such that if I have 5 of the cordless phones hooked up, each one is it's own "extension?" ________________________________ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended

I recently upgraded to asterisk 11 from 1.8. I had VXML working via AGI in 1.8 - from extensions.conf: [VXML] exten => s,1,Answer exten => s,n,Set(ENCODED=${URIENCODE(${ARG1})}) exten => s,n,AGI(agi://localhost/url=${ENCODED}) exten => s,n,Hangup Using asterisk 11 on the same host with the same config in extensions.conf: -- Executing [s at VXML:1]