similar to: off topic - DECT phones with FSK VMWI in the UK

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

May be I have fiddled too much with my sipura settings but I can't get it to give the stutter tone when there is a new voice mail waiting on the asterisk box. I can either get a stutter tone all the time or not at all. Anyone got this working. Thanks Chris

The TDM410P doesn't support 'hvac', only the obsolete TDM400P supports that option was for the old phones that have a neon light (or equivalent LED+ZENER ciruit). Are other phones off the TDM410P (other than the VTECH) working, or is the Vtech the only model with VMWI available to you. I'm not able to check at the moment, I have copied the asterisk-users list, someone else may

Anyone else got a Sipura 3000 in the UK? Apart from CID not working it also seems to not notice any of the line state changes on the PSTN when the remote party terminates the call. It only recognises the offhook signal which gets sent much later. Chris

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

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After purchasing the Uniden TRU9480 and then the Panansonic 5672, both of which do not have "phone company" compatible FSK/stutter MWI, I finally got smart and found out just which Panasonic phones have this feature. Only the following 5.8G models in their current line have FXO compatible MWI. I purchased the 5771 unit and one remote. I have confimed it does in fact work with Asterisk

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 >

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/7e2c9deb/attachment.htm -------------- next part -------------- Hello I have Scitec 5S-c Analog Phone connected to Asterisk PBX. This phone has Message Waiting Indicator but I don't see it lighted whenever any voice mail comes. From phone documentation I came to know that this phone

Ive just installed a TDM4000P with 4 fxos. The zaptel config is fine, zttest comes back with configured. If i call a line when zttest it shows on the display,and then goes when the line drops. In * when a call comes in, it follows my dialplan and answers the call according to the log, but IT DOESN'T actually pick up the call, i.e. it continues ringing. I'm using KS signalling, and

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

Basically, pretty much all the 2 line cordless systems I've seen come with a built in digital answering system that I'll never use, the main problem with this is that these units don't support VMWI (visual message waiting indicator) with telco supplied voicemail. This is a problem because I'll be setup an Asterisk system in the next month or so to handle my 2 analog and 1

Anyone written an extension that will take a 12 digit number, convert it to an IP address so that you can make a sip call to it. Chris

see below I tried to call 98013356 from the following phone (from mgcp.conf) [iptlf03] host = 192.168.33.3 context = default inbanddtmf = 1 callerid = 22545062 line => aaln/1 Console output: == Spawn extension (capiring, 9988001133335566, 1) exited non-zero on 'MGCP/aaln/1@iptlf03-1' -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 -- Delete connection 4

Hi list, I face a problem with voice SMSs. In some countries, if you send an SMS to a landline number, the mobile operator will record the message and then call this number. When picking up the phone you hear "You get an SMS from <phone number>, press 1 to listen the message, 2 to repeat the sender phone number". If you press 1 you hear the message and after it you have the

Hi all, Hopefully you can help me. I want to turn off the audio "Call Waiting" beep that plays during a call. I have found the line in the indications.conf for Call Waiting but apart from setting the frequency to zero or the length to zero is there a proper way to disable this functionality. thanks very much alex This email and any attached files are confidential and

I've got the following setup: IAXy -> Dect Base Station. When you dial from a SIP phone (cisco 7960), the rings with very little delay. However, if you hangup it takes 3-4 rings after hanging up before the dect base station phone stops ringing. The same applies when an incoming call is directed from PSTN FXO -> Dect Base. Is there a fix to this I've looked about on voip-info but

Hello, I was hoping that this would work: http://blogs.sun.com/zhangfan/entry/how_to_turn_a_mirror I have 4x(1TB) disks, one of which is filled with 800GB of data (that I cant delete/backup somewhere else) > root at FSK-Backup:~# zpool create -f ambry raidz1 c4t0d0 c5t0d0 c5t1d0 > /dev/lofi/1 > root at FSK-Backup:~# zpool list > NAME SIZE USED AVAIL CAP HEALTH ALTROOT