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...t 600 and route inbound calls towards asterisk. Can I have more than one CMG in a single chassis? Or maybe you know of a better way to connect T1's to asterisk without zaptel cards using SIP Trunks? Thanks Bart -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com -------------- next part -------------- A non-text attachment was scrubbed... Name: bart.vcf Type: text/x-vcard Size: 231 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070903/1261d185/attachment.vcf

I'm trying create a feature that allows a callers to add more speech to his recording. I think this can be done inside a dialplan, but I can't find an example of how to do this. Basically,after he records the primary message, a menu would play asking if he wants to append to this message. If yes, then he would record a temp file with the additional message and when done, I want SOX to

I was looking at these SYMBOL NETVISION II NP-3010 VoIP TCP/IP WIRELESS PHONES - I know they have been discontinued. Am I asking for trouble to buy some of these for use on Asterisk? TIA Bart

...lan. I'd like to count total channels system-wide, but even better if I can determine for a selected extension also. I've searched the wiki, and don't see such a function that does this. Any ideas? Bart -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com -------------- next part -------------- A non-text attachment was scrubbed... Name: bart.vcf Type: text/x-vcard Size: 231 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070907/f4320f9e/attachment.vcf

...going in to a IVR system. The IVR messages are recorded .wav format - The system appears to crap out at about 40 calls - Would using GSM or some other format help save CPU cycles? Using 1.2, Dual Xeon and 2GB ram TIA -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com -------------- next part -------------- A non-text attachment was scrubbed... Name: bart.vcf Type: text/x-vcard Size: 231 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070909/60585f14/attachment.vcf

We have a Te410P with 3 Telco T1's (D4 SF ) with DID's (non-PRI). ANI & DNIS is received in-band DTMF in a format such as *7145551212*8002* What happens when there are 30 or more calls, asterisk might see is DNIS = 802 or ANI = 4551212 for examples, where parts of the numbers are dropped. All the traffic arrives into a simple IVR script where a message is played. We are

Does anyone have all the Telco intercept prompts (numbers and such) with voice inflections to simulate number referrals and disconnects I could download? TIA, Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080522/08b001ce/attachment.htm

How can I know for sure if SIP Trunk Provider is sending DTMF 'inband' or 'rfc2833'? And more importantly if they could be sending both? If I specify 'inband' should they honor that? Thanks, Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081002/3b34d38d/attachment.htm

> I'm going blind searching - maybe you know? > > During the execution of a script I want to play fake ring to caller. > Both of these examples complain of missing option: > > $agi->exec("Ringing"); > $agi->exec("Playtones ring"); > > Notice: Undefined variable: options in > /var/lib/asterisk/agi-bin/includes/phpagi.php on line 326

Is it possible to have a different verbose level full log than to console output? I'd like to keep console verbose at 1, but now full log is at 1 also. Bart -------------- next part -------------- A non-text attachment was scrubbed... Name: bhfisher.vcf Type: text/x-vcard Size: 253 bytes Desc: not available Url :

I have some old PC's I want to build as a test box - It's up and running OK now. Now I installed a TDM400P and there is nothing I can do to get the card to come up. My guess is the box is not PCI 2.2 compliant or does it need to be to see the card? Thanks, Bart Here's what I know: Processors 1 Model Pentium III (Katmai) CPU Speed 551.37 MHz Cache Size 512 KB System Bogomips

I currently have asterisk-1.4.26.2 installed and working. It was sugguested I try asterisk-1.4.25 to see if it fixes my SIP dtmf problems. What is the method to downgrade? Do I just do in the asterisk-1.4.25 folder: make clean ./configure make install Or do I need to 'make clean' in the asterisk-1.4.26.2 first then move to the asterisk-1.4.25 folder and do ./configure & make

I've been reading the Google searches trying to understand how to tie together Adit 600 to Asterisk to provide 2 way service. I'm about blind from reading. I assume, the answer is using MGCP between the boxes. However, the examples I found don't really explain fully enough to know how to modify examples to work for me. I'll have in the ADIT with T1's. There is a CMG and

Can anyone clue me in about these T400 T1 cards I see advertised? I hear they are Digium Clones. Is there some reason to avoid these? How do they compare to TE410P's for example. Bart

I just started poking around with writing a python module to interface to the Manager API, and it suddenly hit me... how the heck are you supposed to program this thing? All the events seem to be dumped to all the open connections. If I send a command, such as a login, there seems to me to be no way to determine which response are intended for me, and which may be intended for another open

I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

I purchased the DigitNetworks VoIP Starter Kit Full (FXO Card & GrandStream BudgeTone-100 IP Phone) To tell the truth, I can't believe I've got it working this far! Most everything is working. However, I'm having a few problems outlined below: Using XLite: - Working inside the LAN I can dial and use all the options in the demo IVR I can dial to an outside line telephone

I see repeated over and over the following messages: NOTICE[1142106560]: chan_sip.c:4988 handle_response: Peer '1001' is now REACHABLE then 5 minutes later: NOTICE[1142106560]: chan_sip.c:5958 sip_poke_noanswer: Peer '1001' is now UNREACHABLE both messages repeated over and over Any clue what I can do to fix this? Is there any where I can look up these Notices to find

I'm trying to get this zap dial to work. I want to send DNIS and ANI to other system (ZAP/g2) at answer, while the caller hears ring (RBT). I hear the RBT, but callerid and exten is not sent to other T1 - The ZAP/g2 T1 is standard D4, SF, E&M Wink Start. - At ZAP/g2 wink, asterisk should send DTMF "*ANI*DNIS*" exten => _XXXX,1,NoOp,${CALLERID} exten =>

I have a TE410P with two real Telco T1's and the other 2 ports terminate into an in-house voice mail/IVR system. Calls arrive from Telco are routed to the appropriate in-house system based on the DID Digits. This part works perfectly. Now I what to allow the in-house VMS to dial though asterisk to the Telco T1's. The VMS complains there is no dial tone on its T1 and drops the attempt.