similar to: UPDATE - Analog Phones with FSK/Stutter MWI

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

Hi, Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] & same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry

Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after

Off topic but I am after a DECT phone to connect to my sipura 3000 that has a FSK VMWI light or flashing envelope on the LCD screen. Any ideas Chris

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3050 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060309/33760a15/smime.bin

Hello I've been googling for a couple of days now, but still can't figure out what to put in zapata.conf to get it to report CID. Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202 as CID FSK Standard: http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg Does Zaptel support those on Digium

Hi, I'm currently having problems detecting FSK BT (UK) caller id in our API (Pika boards). I have a recording to test on but it is giving me checksum errors. I'm wondering if someone from UK using BT lines could send me a recording with the FSK signal so I can have more data to work on? If you have something, please send it to paulo.astuser at gmail.com. I appreciate all help about

hi: in our country callerid is sent with fsk. but it will sent DTAS(dual tone alerting signal) first, then fsk callerid, then first ring. I search google, but didn't find the configuration method or patch for this. any good suggestion for asterisk to detect this? I have trid asterisk 1.4 and 1.6.2. thanks for help!! Regards, tbskyd

Hi Isamar maybe I think disclose your code to CVS is best and fast :-) mack > > Hi folks, > > I'm trying now to play with fsk_modem.c and callerid.c > to get the Japanese callerid working and I already got to make some > steps.. > I don't know if anybody accomplished that already... but > Since two or more minds think better than one, send private messages >

Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug

Last week I asked about MWI indicators on wireless phones that would work with Asterisk. I sent a message off to Panasonic asking them about it because in their ads they specifically stated that the indicator works with and requires phone company voicemail subscription. The is the model TG5631. Specs here... http://www.amazon.com/Panasonic-KX-TG5631S-GigaRange-Cordless-Answering/dp/B000F4C2CA

Hi, This question isn't specifically asterisk related, but perhaps someone here can shed some light or offer some insight. Is anyone else here running VoIP over Alvarion wireless? If yes, do you have any suggestions for what you've done to make it "work"? It seems that no amount of traffic shaping, checking installs for error rates, lowering error rates, or setting

Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden stuff, no dice. Any ideas? Thanks... -Ken unidencom.txt: OverwriteLocalSettings

Anyone used any of the Uniden VoIP gear, any comments one how easy it was to integrate with asterisk? Also quality of the gear in general? This unit _looks_ pretty nice on paper http://www.uniden.com/productpop/00_productpop.cfm?prd_code=UIP1868 -- Harondel J. Sibble Sibble Computer Consulting Creating solutions for the small business and home computer user. help@pdscc.com (use pgp keyid

I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect'

hello, yesterday the uniden uip200 phone was recommended to someone. i am looking for an alternative to grandstream bt-100 because i can not do a supervised tranfer with it. here my questions: 1) does the uip200 support supervised transfers? 2) can i buy the phones in europe, especially in germany? thanks in advance, jan goericke

I got a Uniden UIP200 and started to configure it and I am lost.... I have a tftp server setup on my * server and have the files unidencom.txt and uniden<mac>.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I

Hi, I'm new to asterisk, and have a uniden UIP 200 that I got off of ebay. I'm having trouble getting the phone to register with asterisk. I've tried a few different settings. I'd be extremely grateful if someone with a similar setting could give me the sip.conf block for the UIP and the settings you're using in uniden.txt. Here's what I have currently: IP of phone

Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at

Hi, I've read through the archives, and wanted to get an updated opinion on the Uniden UIP200 phone. Seems like there were a lot of opinions that it was a good phone, but there were a few items that people were waiting for firmware updates for, but that was in 2004. I'm going to be using them in an office, 12 phones, on a LAN connected to an asterisk box. Thanks for any advice or