similar to: Asterisk Maintenance

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

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

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'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 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

Occasionally my /etc/hosts file gets corrupted. The IP address and the host name switch positions with the host name to the left. What this happens, my 7940 phones won't register. Fixing /etc/hosts allows the phones to register. Do any of you Linux gurus know who is corrupting the hosts file? Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575

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

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

Dear gods of the PBX (flattery works here, right?) - I've been pulling my hair out trying to get Asterisk to receive and decode a fax using spandsp and app_rxfax. It seems like it should be working. The fax machine on the other end connects and Asterisk reports a fax coming in. But when it's done all I have is a 2 or 3 KB TIF (see attachment). The console activity looks completely

I'm losing call files in /var/spool/asterisk/outgoing because * isn't able to detect the busy signal. The call file looks like this: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: <filename>.tiff|caller Using the "|caller" parameter, TxFax injects the fax tone (CNG) onto the line. With the CNG tone, asterisk is unable

Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would

Hello I have an office with a T1 that provides 4 (out of 8) analog PSTN lines thru an adtran board. I want to add 4 more analog lines. Currently I have a Digium TDM40B. I'm wondering what the best upgrade path is, where I define 'best' as the solution that is most likely to work without problems (like interupt conflicts) and work with my current echo tuning . I see my purchase

Message: 16 Date: Thu, 19 May 2005 00:16:34 -0600 Michael, Do both! As for Sip Termination: ----------------------- Contact Kristi Eggers @ Txlink.net for month to month Originating/Termination VoIP Toll Free or Local USA DID #s. Yes they do both Sip and IAX. You must have seperate accounts for either Sip or IAX and fund your account with a minimum of $100. This is what I did. Once I get

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 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,