similar to: [OT] What's changing /etc/hosts?

Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

Is there a Linux tool that will splice several gsm sound clips together into one clip? In my agi script, I would like to use 'get_data' with one clip instead of multiple 'stream_file' so the user doesn't have to listen to the entire spiel before responding. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com

I finally got my ASUSCOM (Cologne chip) ISDNLink card working here in the US. When a call arrives, I get "Unknown IE 42 (Unknown Information Element)" and "Unknown IE 21 (Unknown Information Element)". IE 21 (0x15) is defined as Q931_CALL_STATE_SUSPEND_REQUEST. IE 42 (0x2a) is not defined in the code and I couldn't google it. Is this something perculiar with ISDN in

I need to connect a bunch of analog telephone sets. Does anyone have any comments about this channel bank? Disconnect supervision? Echo? ADSI problems? The price is right @ $995 new and $695 refurbished. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

Has anyone implemented HOBIC SMDR output from *? Can someone point me to the Bell HOBIC specification? Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

I'm having problems with the voicemail prompts. The beginning of each prompt is garbled. Prompts "edian mail" and "assword" can be recognized, but when it starts on the short phrases it is completely garbled. It is almost like the subsequent phrase starts before the current phrase finishes. I have isolated the LAN to the * box and the phones, so LAN congestion is

I need to reopen this discussion because it's impossible to run spandsp (and VoIP) under these circumstances. With zaptel unloaded, I see the following "vmstat 1" output: no swapping, an occasional disk output, +/- 1003 interrupts/sec., less than 10 context switches/sec., CPU idle 100%. A very quiet system. I load modules zaptel and wcfxo, and the system utilization stays the

I'm debugging a TxFax problem whereby the fax transmission fails. I suspect interrupt latency--some interrupt routine is holding its interrupt too long. I have all unnecessary services switched off and X is not running when I perform these tests. Some transmission are successful while others fail at random points. I've noticed that after I boot Linux, load zaptel, wcfxo, and wcfxs,

I've seen other posts about this problem, but I haven't found a solution. I'm dumping eight call files into the "outgoing" directory at one time. Three of the calls are successful while the other five are lost. Here is the call file: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: <filename>.tiff|caller Note: All

I'm having an intermittent problem with fax transmissions, something I call the streaking mascara syndrome. TxFax will be part way through a transmission when something fails--on the fax printer, part of the page will have correctly printed, but the rest of the page is "streaked" to the bottom. I have several successful multi-page transmissions interspersed with streaked

I went on a service call yesterday to find an asterisk system with a T100P card on a Qwest PRI and a TDM40B card connected to fax machines. The TDM40B LEDs were not lit, and the system did not respond to keyboard input. However, calls were being processed for the PRI and 7960 phones. I replaced the TDM40B card with a new one, and the system now seems to be ok. But I'm wondering, why

I realize why leaving email addresses in the archives would definitely _not_ be a good thing. However, is there a way to reply off list to an archived message? I would like to contact 'ulexus' about his asterisk/hdlc setup. Would anyone have his email address? Sorry for the interruption, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575

I'm trying to use ztcfg with the following zaptel.conf: span=1,1,0,esf,b8zs bchan=1-4 nethdlc 5-23 dchan=24 and I get "ZT_CHANCONFIG failed on channel 5: Function not implemented (38). This is on Debian 2.4.26. I have hdlc compiled as a module. Should hdlc be compiled into the kernel? Does anyone see a problem here? Thanks, -- Michael Welter Introspect Telephony Corp. Denver,

Does anyone have a recommendation for a 48 port LAN switch for a new * system? I'm not happy with NetGear's reliability. -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

I believe it is not fiber, but I am not sure. I am going to take one of them home tonight and hook it to my POTS line there, which for sure is not fiber. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Michael Welter Sent:

Just when I thought I couldn't be wrong, I was wrong. We have woodpeckers that drill into the arial telephone cables, and water seeps through the holes and partially grounds the tip and/or ring wires causing hum. I thought the hum/buz on my lines was a telco problem. The Qwest HQ noise team assures me that my lines are within spec. Sure enough, when I listen on the test set the lines

I have an * system backending a Fujitsu 9600 system. The interface is a P2P T1 circuit using E&M wink. At 6PM every evening the T1 circuit goes down. I don't have any cron or 'at' jobs that fire at this time. Does anyone have any insight? Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

I've received my Aastra 390 phone. I got the unlock procedure from the vendor, and the services button now shows all four entrys as "<available>". When I give the phone "ADSIProg()" the phone displays: Asterisk PBX download refused Conflict with: <available> The asterisk.adsi hasn't been changed: DESCRIPTION "Asterisk PBX" VERSION 0x02 SECURITY

Any clues as to what this is? Google isn't much help. Both cards have the power connector plugged in, and both randomly give me static instead of a dialtone, or an outside line. Cards are 1 4 Port TDM400 FXO, and 1 4 Port TDM400 FXS. Computer is a Compaq Proliant ML330 The full error is as follows: Ouch, part reset, quickly restoring reality (0) Power alarm on module 1, resetting!

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show