Re: [asterisk-users] Method to use SOX inside a Dialplan

Steve Edwards wrote: On Sat, 10 Oct 2009, Bart Fisher wrote: 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,

[asterisk-users] DTMF Issues

I have a block of DID's that I ported to Vitelity about 7 days ago. The problem is if a POTS caller dials into the system, his dtmf is not heard at READ() or Background() while a prompt is played. After the prompt is finished, then dtmf is heard. I've been working with their support, but it

[asterisk-users] DTMF problems during a message play

I'm using the latest asterisk-1.4.26.2 and no zaptel trunks used, all SIP. I have one user that is having problems once he connects to asterisk. He's dialing from his home phone (pstn) to a Vitelity DID (SIP Trunk) which goes to my asterisk IVR. If he presses a dtmf during any message, the

[asterisk-users] different verbose level for full log than to console?

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 attachment: bhfisher.vcf___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] ACR Anonymous Call Rejection

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 attachment: bhfisher.vcf___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon

[asterisk-users] MySQL question

Here's my dialplan: [initialize-log] exten = _X.,1,Noop(Initialize CallLog ${CallersDT} ${CallersTel} ${LOGCONFIRM}) exten = _X.,n,MYSQL(Connect connid ${HOST} ${USER} ${PASSWORD} ${DATABASE}) exten = _X.,n,MYSQL(QUERY resultid ${connid} INSERT\ INTO\ tbl_calllog\ SET\

Re: [asterisk-users] Play Fake ring in phpagi

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

Re: [asterisk-users] RBS T1 DID issue

you need to port you zaptel.conf zapata.conf (might be channel-additional.conf in trixbox) Bart - Original Message - From: Jeff LaCoursiere j...@jeff.net To: asterisk-users@lists.digium.com Sent: Monday, February 02, 2009 6:24 PM Subject: [asterisk-users] RBS T1 DID issue Howdy,

Re: [asterisk-users] Pay Phone Controller Project

Very Cool! But then does anyone still use payphones ? :) Good job Bart - Original Message - From: Stephen Rodgers hws...@rodgers.sdcoxmail.com To: asterisk-users@lists.digium.com Sent: Saturday, January 10, 2009 10:13 AM Subject: [asterisk-users] Pay Phone Controller Project I

[asterisk-users] A method to determine PSTN Call Provider?

I'm looking for a solution to determine if a PSTN call to a zaptel channel was originated from a VoIP provider or not in real time. I'd like to use the callerid(num) to reverse match to the provider. Does anyone have a clue how I could do this? TIA

Re: [asterisk-users] A method to determine PSTN Call Provider?

, 21 Dec 2008, Barton Fisher wrote: I'm looking for a solution to determine if a PSTN call to a zaptel channel was originated from a VoIP provider or not in real time. I'd like to use the callerid(num) to reverse match to the provider. Does anyone have a clue how I could do this? What country

Re: [asterisk-users] Record CMD

Exactly! but sadly these variables don't seem to exists as far as I can tell The point is that you're the first person to make this request. If nobody had the idea to do it before you, that is precisely the reason it never got done. Now that it has been requested, it is in queue for

Re: [asterisk-users] Record CMD

:05 Barton Fisher wrote: I don't see a method to detect the success or failure for the Record CMD. I'd like to know the reason why the recording ended Am I wrong? exten = recordmsg,1,Noop() exten = recordmsg,n,Record(${NEWPHRASEID}:ulaw|4|180) So you'd be looking for a RECORD_STATUS

[asterisk-users] Record CMD

I don't see a method to detect the success or failure for the Record CMD. I'd like to know the reason why the recording ended Am I wrong? exten = recordmsg,1,Noop() exten = recordmsg,n,Record(${NEWPHRASEID}:ulaw|4|180) Bart___ -- Bandwidth and

Re: [asterisk-users] top posting again [was: Re: CDR Design]

Bottom posting only works if you trim the post to the parts you are answering - nobody does this! So we end up reading and re-reading the same old post over and over - Bottom posting make NO sense due to this. Bottom posting is stupid and out of date - likely applied more when people used,

[asterisk-users] MySQL Error Message

Can some tell me what this warnings means? The dialplan works, but I get these warnings every once in a while: Log: [Dec 1 12:26:31] WARNING[20425] app_addon_sql_mysql.c: Identifier 0, identifier_type 2 not found in identifier list [Dec 1 12:26:31] WARNING[20425] app_addon_sql_mysql.c: Invalid

Re: [asterisk-users] MySQL Error Message

Barton Fisher wrote: Can some tell me what this warnings means? The dialplan works, but I get these warnings every once in a while: I'm guessing that some times the caller-id is blank. I got tired of those errors and did the following before the query: exten = s,1,GotoIf($[${CALLERID

[asterisk-users] DTMF

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___ -- Bandwidth and Colocation Provided by

[asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?

Hi, It has IOS is 12.3(14)T7 - I'm wondering if this router can be made SIP aware. Apparently, this current firmware/programming is not, one way audio problems. Is there a version that support VoIP directly for this router? Thanks, Bart___ --

[asterisk-users] Can you verify this bug?

I'm stuck on 1.2 until I can pass DTMF from a SIP Trunk (Vitelity Virtual PRI) call towards a ZAP (TE410P using em wink) port. The call connects OK, I can hear DTMF with DNIS ANI inband from asterisk to the external IVR, Voice is OK, but if any DTMF is required after the bridge has been

Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP

along with changing the frequencies at the same time? Thanks, Steve T On Sat, Jun 21, 2008 at 3:29 PM, Barton Fisher [EMAIL PROTECTED] wrote: OK, tried changing DTMF tone as described on URL and no difference Bart Steve, I fooled with dtmf mode and it was 2833 - However, got stranger

Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP

to acknowledge your issue. What features do you need in 1.4 anyways? Maybe if the DTMF bugs you opened get resolved then 1.4.X could be revisited. Thanks, Steve T On Sun, Jun 22, 2008 at 11:30 AM, Barton Fisher [EMAIL PROTECTED] wrote: Yep - tried both and combination thereof - The bad effect of inband

[asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP

I place SIP DID call towards ZAP (TE410P). ZAP uses em signaling to an external IVR system. I can hear the asterisk sending the DTMFs properly toward ZAP at call setup. After the call connects, any further DTMF digits from SIP is very short sounding or distorted (barely audible) on the ZAP and

Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP

guess I could switch back if it turns out to be a bad idea Bart On Sat, Jun 21, 2008 at 12:11 PM, Barton Fisher [EMAIL PROTECTED] wrote: I place SIP DID call towards ZAP (TE410P). ZAP uses em signaling to an external IVR system. I can hear the asterisk sending the DTMFs properly toward ZAP

Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP

what's it doing, but wonder what else would be effected afterwards - I guess I could switch back if it turns out to be a bad idea Bart On Sat, Jun 21, 2008 at 12:11 PM, Barton Fisher [EMAIL PROTECTED] wrote: I place SIP DID call towards ZAP (TE410P). ZAP uses em signaling to an external IVR

[asterisk-users] Telco intercept prompts

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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

[asterisk-users] ADIT TDM T1 Asterisk MGCP

I have this idea to use an old ADIT 600 with a CMG card to convert two T1 TDM circuits to MGCP towards asterisk. The basics I've found on the net, but there is not much available. I've cut and pasted the mgcp.conf details I could find, but there not much as far as CMG setup. I was hoping

Re: [asterisk-users] DTMF dropping digits

In article [EMAIL PROTECTED], Barton Fisher [EMAIL PROTECTED] wrote: 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

[asterisk-users] DTMF dropping digits

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

Re: [asterisk-users] Which cause less CPU usage: GSM or wav??

recording new messages in ulaw instead of converting them all to ulaw at once. So it's possible to have two prompts with both file extension at a time Bart Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Barton Fisher wrote: Thanks, OK, a bit confused The cards

Re: [asterisk-users] Which cause less CPU usage: GSM or wav??

Thanks, again. That did the trick! Bart Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Barton Fisher wrote: Thanks, OK, a bit confused The cards are TE410P. I really don't see how the set a codec for this, other than it might default to something in code like ulaw

[asterisk-users] Which cause less CPU usage: GSM or wav??

I have 4 TDM T1's 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

Re: [asterisk-users] Which cause less CPU usage: GSM or wav??

Sep 07, Barton Fisher wrote: I have 4 TDM T1's 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 depends on what

[asterisk-users] Channels in use?

this. Any ideas? Bart -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com begin:vcard fn:Barton Fisher n:Fisher;Barton org:Innovative Communications adr:;;7439 La Palma Ave # 255;Buena Park;CA;90620;USA email;internet:[EMAIL PROTECTED] tel;work:714-228-5410 url:http

[asterisk-users] ADIT 600 CMG = Asterisk question

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 begin:vcard fn:Barton Fisher n:Fisher;Barton org:Innovative Communications

[asterisk-users] Append Extension number sounds to Voice Mail Message?

the voice message when played. Bart -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com begin:vcard fn:Barton Fisher n:Fisher;Barton org:Innovative Communications adr:;;7439 La Palma Ave # 255;Buena Park;CA;90620;USA email;internet:[EMAIL PROTECTED] tel;work

Re: [asterisk-users] ADIT 600 CMG = Asterisk question

, only DI. I use them all the time to breakout FXS/FXO's for incoming and outgoing analog lines, but they have a tendency to introduce lots of echo.I've had to use HWEC every time I use the 600. -D From: [EMAIL PROTECTED] on behalf of Barton Fisher Sent

[asterisk-users] Telco Testing locks up asterisk

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

Re: [asterisk-users] Unable to install Asterisk Now Beta 6

I don't believe AsteriskNow will install on a dual processor system. I had this same problem - installing on single process MB went OK I don't know how to fix, so went with elastx.org and adminsparadise.com packages, both seemed to be OK - can't decide which one to keep - the last choice, maybe

Re: [asterisk-users] Console duplicate output problem

Eric ManxPower Wieling wrote: This is really strange. Every message to the (VGA) console is written twice to the screen, but not on the SSH connection. Any clues how to stop this behavior? Stop running in graphics mode. OK, that's a great clue, but can you tell me how to disable now?

Re: [asterisk-users] Console duplicate output problem

Anybody have an answer? TIA This is really strange. Every message to the (VGA) console is written twice to the screen, but not on the SSH connection. Any clues how to stop this behavior? -- Executing BackGround(Zap/216-1, custom/3566/91_|m|) in new stack -- Executing

[asterisk-users] Console duplicate output problem

This is really strange. Every message to the (VGA) console is written twice to the screen, but not on the SSH connection. Any clues how to stop this behavior? -- Executing BackGround(Zap/216-1, custom/3566/91_|m|) in new stack -- Executing BackGround(Zap/216-1,

Re: [asterisk-users] ${ANSWEREDTIME} Broken on 1.2.13?

stuff... exten = _X.,n,Set(ETIME=${EPOCH}) ; save the end time exten = _X.,n,Set(DUR=$[${ETIME}-${STIME}]) ; set DUR to difference (seconds) Bart Joshua Colp wrote: Barton Fisher wrote: No matter what I do, ${ANSWEREDTIME} is always 0, even on the most simplest dial plan such as: Using

[asterisk-users] ${ANSWEREDTIME} Broken on 1.2.13?

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, )

Re: [asterisk-users] dialplan / problem with extension-length 1

Try moving 2 digit extensions before single digit. I believe asterisk matches the first found extension which is always the single digit extensions the way you have it Bart Michael Kamleitner wrote: hi community, I'm new to this list asterisk in general, so let me first say thx to

Re: [asterisk-users] EM Wink start problem

We use EM wink here: Basically we have asterisk talking to dialogic cards, but shouldn't be much different to a PBX zaptel.conf # Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 AMI/D4 RED span=1,0,0,d4,ami em=1-24 zapata.conf signalling =em_w context=from-pstn group = 1 channel = 1-24 for

[asterisk-users] MySQL Update from exten

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

Re: [asterisk-users] 3rd T1 of quad card won't change signaling

Looks like: amaflags=billing switchtype=national is being carry-over from prior PRI.. (All PRI stuff) Try moving below before the first PRI? ; NEW FAX t1 group=3 signaling=em_w context=from-internal channel = 49-72 Bart Jay Wilton wrote: Hello, I'm trying to set the 3rd span of a new

[asterisk-users] Stuck on MySQL UPDATE

What I'm retrying to do is update mysql field with the new message ID that was just recorded. Ideally, I'd like to specify the field to update using a variable ${BINID} and use ${NEWPHRASENAME} for the value - I'm not sure asterisk will allow using a variable for the field name and if not, I'll

[asterisk-users] MySQL query from extensions?

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 =

Re: [asterisk-users] MySQL query from extensions?

,n,return Bart Alex Balashov wrote: On Fri, 13 Apr 2007, Barton Fisher said something to this effect: What wrong with this: Well... what is wrong with it? :-) I'm not trying to be funny, but, what are the symptoms that it doesn't work? Error output on Asterisk console? Logs? Anything

Re: [asterisk-users] MySQL query from extensions?

}) exten = s,n,returnpes On 4/14/07, *Barton Fisher* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sorry, From the logs I see: Apr 13 13:32:06 WARNING[19854] app_addon_sql_mysql.c: Identifier 0, identifier_type 2 not found in identifier list Apr 13 13:32:06 WARNING[19854

[asterisk-users] Noob question regarding PCI 2.x TDM400P Card

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:

Re: [Asterisk-Users] SYMBOL NETVISION II NP-3010

Thanks Andy... I decided not to purchase for the reason you stated. Bart - Original Message - From: Andy Hamilton [EMAIL PROTECTED] To: Barton Fisher [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 17, 2005

[Asterisk-Users] SYMBOL NETVISION II NP-3010

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 ___ Asterisk-Users mailing list

[Asterisk-Users] CONSOLE/dsp

I'm trying to create an extension that will connect caller to asterisk sound card. I've followed the example at http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+console. With no luck. What I get is: Jul 13 09:56:45 VERBOSE[1315]: -- Executing Dial("SIP/300-3bd6", "CONSOLE/dsp")

[Asterisk-Users] Dial ZAP Problem

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, EM Wink Start. - At ZAP/g2 wink, asterisk should send DTMF

Re: [Asterisk-Users] SpanDSP - Squished Faxes

Which T1 card? Had same problem with TE410P. Things I did: 1. Move card to higher priority IRQ fixed problem (IRQ10). 2. Make sure IRQ is not shared. 3. Disable everything in CMOS that's not needed or using - COM, LPT, USB, Hyper-Threading, and the likes. 4. Use the latestZAPTEL Drivers.

[Asterisk-Users] Zaptel and Zapata Conf's

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding,

Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?

Wow! I never learn so much! Thanks Guys So if I understand correctly, a full T1 should be 1.5Mbps full duplex. And it should support 22 SIP Users at once - Right? Bart - Original Message - From: Wiley Siler [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial

[Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?

I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choosea T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loopsor 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart ___

[Asterisk-Users] TE410P Drops Calls after many touch tones from caller

Ihave a TE410P card with two Telco T1's and two external IVR systems attached. Calls from Telco are routed to proper IVR system based on DNIS (DID) received from Telco using a native bridge. T1's are D4 AMI SF Some IVR applications requires the caller to enter digits using their touch

Re: [Asterisk-Users] TE410P Drops Calls after many touch tones fromcaller

well here is an example dial: exten = 3732,3,Dial(ZAP/g2}/*${CALLERID}*${EXTEN}*) But the logs really only show call hangup Bart - Original Message - From: Tim Connolly To: 'Barton Fisher' ; 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent

[Asterisk-Users] How to make Span Port Selection in Round Robin fashion?

I have span in a group (ZAP/g1) - How can I make this group sequentially select ports on the span, instead always selecting port 1? TIA Bart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] How to make Span Port Selection in Round Robinfashion? - [SP]

Robinfashion? - [SP] Barton Fisher wrote: I have span in a group (ZAP/g1) - How can I make this group sequentially select ports on the span, instead always selecting port 1? Amazingly, a quick search on the wiki turned up this page: http://www.voip-info.org/wiki-Asterisk+ZAP+channels

[Asterisk-Users] Repeated Notice:

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

[Asterisk-Users] Nwebie Config Problem

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 workingthis far! Most everything is working. However, I'm having a few problems outlined below: Using XLite: - Working inside the LAN Ican