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...gi.php on line 323 > I changed to $agi->exec("Playtones ring",""); - no error message but not sure that correct > Any ideas what to put for missing option? > > TIA > > Bart -------------- next part -------------- A non-text attachment was scrubbed... Name: bhfisher.vcf Type: text/x-vcard Size: 253 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090818/a12537eb/attachment.vcf

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 : http://lists.digium.com/pipermail/asterisk-users/attachments/20090920/99149120/attachment.vcf

Does any have or can point me to /ACR/ Anonymous Call Rejection message I can download? The one I found was not not too clear. Thanks, Bart -------------- next part -------------- A non-text attachment was scrubbed... Name: bhfisher.vcf Type: text/x-vcard Size: 253 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090916/c3b682d5/attachment.vcf

...s is hear. I've tried all the suggestions found searching the wiki, so I ask here if there is something else I can try. The Vitelity trunk is set up as: dtmfmode=rfc2833 disallow=all allow=ulaw Thanks, Bart -------------- next part -------------- A non-text attachment was scrubbed... Name: bhfisher.vcf Type: text/x-vcard Size: 249 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091001/5f514489/attachment.vcf

...or a while that do not have this issues. I'm told the upline carrier is the same. So I'm reaching out to you guys to give me ideas of what could be causing the problem and how I can resolve Thanks, Bart -------------- next part -------------- A non-text attachment was scrubbed... Name: bhfisher.vcf Type: text/x-vcard Size: 249 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091007/2f1edbff/attachment.vcf

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'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

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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

Redhat has a 'Hardware Discovery Utility' called Kudzu. When I change cards, kudzu pops up and ask to remove/config the card. Most of the time kudzu has trouble recognizing the Digium Zaptel cards and calls them something wrong, like calling the TDM card a network card. I'm having a devil of a time getting 3 TE410P cards to come up with all green lights. For example one or two cards

Here my script: exten => 230,1,Answer exten => 230,2,NoOp exten => 230,3,Voicemail(u${EXTEN}) exten => 230,4,NoOp(Need results from VoiceMail() above - should be non-zero and provided to ARG3) exten => 230,5,NoOp exten => 230,6,GoToIf($[${ARG3} = 0]?s|8) exten => 230,7,system(/var/lib/asterisk/agi-bin/230.php|${EXTEN}) ; exten => 230,8,Hangup I need to know

I'm trying to connect an Asterisk T1 port to a Dialogic card. The Dialogic side is an external VMS. I setup for ISDN-PRI between systems and have green lights on both card/ports. Zttool shows connection is good also. However, when I try attempt terminate or originate a call to either system, nothing appears of CLI. How can I monitor the "D" channel? zaptel.conf: # Span 8:

I'm a bit confused about how to handle this. I have Asterisk sitting in the middle between a Qwest Long Distance T1 (Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic D/240SC-T1 card. The Qwest T1 originally was connected to the Dialogic card directly. The signaling was set to E&M Wink Start because Dialogic used this as its default settings, so it just worked

I'm looking for a method to signal an inside extension (asterisk extension with external dialing appl.) with a DTMF "A" tone to indicate when Asterisk has completed dialing and the voice path has been cut-though on a ZAP T1 Trunk. If this can be done, I'd also like to know if there is a method to know when the called party has answered from the "A" and "B"

Maybe one of you can help me with this: We have T1's that come from both MCI and Global Crossing as uses channelized (24 Ports per T) with inband (DTMF) ANI and DNIS delivery (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk Server. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail

I just downloaded and installed asterisk-1.2.13 Reading http://www.voip-info.org/wiki/view/Asterisk+AEL2 says I should be using AEL2, but what's downloaded is pbx_ael not pbx_ael2 as expected. Where and how do I get current release of AEL2 - Is there some 'How To' somewhere? TIA Bart

What wrong with this: [get-dnisinfo] ; sub-routine to get owner's password exten => s,1,Verbose( == ) exten => s,n,MYSQL(Connect connid localhost root password dax) exten => s,n,MYSQL(Query resultid ${connid} SELECT\ password\ FROM\ dnislookup\ WHERE\ dnis=\'${IVR-Exten}\') exten => s,n,MYSQL(Fetch fetchid ${password} password) exten => s,n,Verbose( == Password found

I've tried every combination I could find on the net and so far there is no joy The thing is I can do this update from the command line: Maybe some new eyes might find the answer? exten => update,1,MYSQL(Connect connid localhost root password dax) exten => update,n,MYSQL(QUERY resultid ${connid} UPDATE\ caller\ SET\

No matter what I do, ${ANSWEREDTIME} is always 0, even on the most simplest dial plan such as: Using Asterisk 1.2.13 exten => 77,1,Answer exten => 77,2,Playback(custom/dax/S300) ; one minute file exten => 77,3,Noop(${ANSWEREDTIME}) exten => 77,4,Hangup -- Executing Answer("SIP/5402-b7b45f58", "") in new stack -- Executing

Over the last week we've been having issues on our Telco provided TDM T1 with the circuit bouncing for several seconds and restoring itself back into service. The T1 is using a TE410P. On the CLI, I see message that span 1 is yellow alarm, then restoring. I reported this problem to the phone company. Afterwards, they do several circuit tests to the NIU and then to the CSU. At the point