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, 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 add
the temp message to the primary message making it one larger message.
Would you mind showing me an example of how to run SOX inside the
dialplan?
The system() dialplan application will allow you to run any executable.
If you plan on concatenating more than 32 input files you'll have to make
sure you have sox v14.3.0 or later.
The dialplan snippet would look something like:
exten = s,n,system(/path-to-sox/sox a.wav b.wav c.wav)
This would copy a.wav followed by b.wav to a new file, c.wav.
I would code the entire feature up as an AGI so you can hide all the ugly
details like creating files with unique file names, maybe running
normalize on the pieces before concatenation, error handling, maybe even
trimming the leading and trailing silence off each file so the gaps are
consistent, allowing the caller to listen to the new file and accept or
re-record the suffix, cleaning up in case the caller hangs up, etc, etc,
etc.
hmm, no luck. Here's what I have:
exten = append,1,Noop(${PHRASEID}) ; this is the full path to original
message without extension .wav
exten = append,n,Playback(custom/dax/record)
exten =
append,n,Set(TEMPMESSAGE=/var/lib/asterisk/sounds/custom/${IVR-EXTEN}/temp);this
is the full path to temp message without extension .wav
exten = append,n,Record(${TEMPMESSAGE}:wav|4|369) ; Begin recording new
message
exten = append,n,System(sox ${PHRASEID}.wav ${TEMPMESSAGE}.wav
${PHRASEID}.wav)
exten = append,n,system(rm ${TEMPMESSAGE}.wav)
exten = append,n,Goto(record,${IVR-EXTEN},confirm); go back and play
entire message
-- Goto (record,append,1)
-- Executing [app...@record:1] NoOp(SIP/8001-0a0227c8,
/var/lib/asterisk/sounds/custom/7146762004/1_17692) in new stack
-- Executing [app...@record:2] Playback(SIP/8001-0a0227c8,
custom/dax/record) in new stack
-- SIP/8001-0a0227c8 Playing 'custom/dax/record' (language 'en')
-- Executing [app...@record:3] Set(SIP/8001-0a0227c8,
TEMPMESSAGE=/var/lib/asterisk/sounds/custom/7146762004/temp) in new stack
-- Executing [app...@record:4] Record(SIP/8001-0a0227c8,
/var/lib/asterisk/sounds/custom/7146762004/temp:wav|4|369) in new stack
-- SIP/8001-0a0227c8 Playing 'beep' (language 'en')
-- Executing [app...@record:5] System(SIP/8001-0a0227c8, sox
/var/lib/asterisk/sounds/custom/7146762004/1_17692.wav
/var/lib/asterisk/sounds/custom/7146762004/temp.wav
/var/lib/asterisk/sounds/custom/7146762004/1_17692.wav) in new stack
-- Executing [app...@record:6] System(SIP/8001-0a0227c8, rm
/var/lib/asterisk/sounds/custom/7146762004/temp.wav) in new stack
-- Executing [app...@record:7] Goto(SIP/8001-0a0227c8,
record|7146762004|confirm) in new stack
-- Goto (record,7146762004,39)
Anything wrong?
Bart
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[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 still not resolved. SIP callers are not effected. Yesterday, I purchased a DID from Flowroute. The setting are the same as Vitelity. And amazingly, this DID works perfectly. This to me would indicate the problem whatever it is, is Vitelity or its upstream provider. - Am I right here?? Also, I have other DID blocks that have been with Vitelity for 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 attachment: bhfisher.vcf___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 press is ignored unless the press was a #, 0 or *. Otherwise, he needs to wait for the message to stop before the press 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 attachment: bhfisher.vcf___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ${connid} INSERT\ INTO\
tbl_calllog\ SET\ log_start=\'${CallersDT}\'\,\
log_phone=\'${CallersTel}\'\,\ log_confirmation=\'${LOGCONFIRM}\')
exten = _X.,n,Set(ERROR=1042)
exten = _X.,n,GotoIf($["${MYSQL_STATUS}" =
"-1"]?error,${IVR-EXTEN},1)
exten = _X.,n,Noop(Successful MYSQL STAT:[${MYSQL_STATUS}]
CONNID:[${connid}] RESULTS:[${resultid}])
exten = _X.,n,MYSQL(Clear ${resultid})
exten = _X.,n,MYSQL(Disconnect ${connid})
exten = _X.,n,Return
The problem is I don't know if the record was inserted successfully for
certain? I've used this code before and it appears to work. But on this
project the record fails to insert. Me confused...
CLI:
[2009-09-09 09:02:17] VERBOSE[32573] logger.c: ==
[3388-SIP/8001-08d77ab8] [INITIALIZE LOG]: Initialize CallLog
2009-09-09 11:02:178001 1252512137.1
[2009-09-09 09:02:17] VERBOSE[32573] logger.c: -- Executing
[3...@initialize-log:2] MYSQL("SIP/8001-08d77ab8", "Connect
connid xx.xx.xx.xx dbuser pass mysqldb") in new
stack
[2009-09-09 09:02:17] VERBOSE[32573] logger.c: -- Executing
[3...@initialize-log:3] Set("SIP/8001-08d77ab8", "ERROR=1041") in
new stack
[2009-09-09 09:02:17] VERBOSE[32573] logger.c: -- Executing
[3...@initialize-log:4] GotoIf("SIP/8001-08d77ab8",
"0?error|3388|1") in new stack
[2009-09-09 09:02:17] VERBOSE[32573] logger.c: -- Executing
[3...@initialize-log:5] MYSQL("SIP/8001-08d77ab8", "QUERY
resultid 1 INSERT INTO tbl_calllog SET log_start='2009-09-09 11:02:17',
log_phone='8001', log_confirmation='1252512137.1'") in new stack
[2009-09-09 09:02:17] VERBOSE[32573] logger.c: -- Executing
[3...@initialize-log:6] Set("SIP/8001-08d77ab8", "ERROR=1042") in
new stack
[2009-09-09 09:02:17] VERBOSE[32573] logger.c: -- Executing
[3...@initialize-log:7] GotoIf("SIP/8001-08d77ab8",
"0?error|3388|1") in new stack
[2009-09-09 09:02:17] VERBOSE[32573] logger.c: -- Executing
[3...@initialize-log:8] Verbose("SIP/8001-08d77ab8", "1| ==
[3388-SIP/8001-08d77ab8] [INITIALIZE LOG]: Initialize CallLog was
successful MYSQL STAT:[0] CONNID:[1] RESULTS:[]") in new stack
[2009-09-09 09:02:17] VERBOSE[32573] logger.c: ==
[3388-SIP/8001-08d77ab8] [INITIALIZE LOG]: Initialize CallLog was
successful MYSQL STAT:[0] CONNID:[1] RESULTS:[]
[2009-09-09 09:02:17] VERBOSE[32573] logger.c: -- Executing
[3...@initialize-log:9] MYSQL("SIP/8001-08d77ab8", "Clear ") in
new stack
[2009-09-09 09:02:17] WARNING[32573] app_addon_sql_mysql.c: Identifier
0, identifier_type 2 not found in identifier list
[2009-09-09 09:02:17] WARNING[32573] app_addon_sql_mysql.c: Invalid
result identifier 0 passed in aMYSQL_clear
Bart
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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 /var/lib/asterisk/agi-bin/includes/phpagi.php on line 326 Warning: Missing argument 2 for AGI::exec(), called in /var/lib/asterisk/agi-bin/dax-ivr.agi on line 156 and defined in /var/lib/asterisk/agi-bin/includes/phpagi.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 begin:vcard fn:Barton Fisher n:Fisher;Barton org:Innovative Communivations adr:;;7439 La Palma Ave Ste. 255;Buena Park;CA;90620;US email;internet:bhfis...@icpage.com tel;work:714-228-5499 x-mozilla-html:TRUE url:http://icpage.com version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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, New installation, trying to connect an RBS T1 with AMI/D4 framing and EM Wink. Using a Sangoma A102d and asterisk 1.4.22-2 on Centos5 (Trixbox 2.6.2.1). Outbound calls work fine, but inbound calls fail to read the DID information, and with debug set to 10 I get the following: [Feb 2 19:40:23] DEBUG[25184] chan_zap.c: Monitor doohicky got event Wink/Flash on channel 3 [Feb 2 19:40:23] VERBOSE[25273] logger.c: -- Starting simple switch on 'Zap/3-1' [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Exception on 14, channel 3 [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Got event Wink/Flash(3) on channel 3 (index 0) [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Got wink in weird state 4 on channel 3 [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Exception on 14, channel 3 [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Got event Hook Transition Complete(12) on channel 3 (index 0) [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Exception on 14, channel 3 [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Got event Wink/Flash(3) on channel 3 (index 0) [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Got wink in weird state 4 on channel 3 [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Exception on 14, channel 3 [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Got event Wink/Flash(3) on channel 3 (index 0) [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Got wink in weird state 4 on channel 3 [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Exception on 14, channel 3 [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Got event Wink/Flash(3) on channel 3 (index 0) [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Got wink in weird state 4 on channel 3 [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Exception on 14, channel 3 [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Got event Wink/Flash(3) on channel 3 (index 0) [Feb 2 19:40:24] DEBUG[25273] chan_zap.c: Got wink in weird state 4 on channel 3 [snip more of the same] [Feb 2 19:40:26] DEBUG[25273] chan_zap.c: Got event Ring/Answered(2) on channel 3 (index 0) [Feb 2 19:40:26] DEBUG[25273] chan_zap.c: Ring detected [Feb 2 19:40:26] WARNING[25273] channel.c: Unexpected control subclass '2' From here a bunch of SIP debug begins and the call progresses as if no DID was sent. Any ideas? Thanks! j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 finally documented my Payphone Controller project used to control Western Electric Fortress pay phones using Asterisk and some external hardware. The link to the project info is here: http://qrvc.com/gpl-projects/payphone. Steve. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Bart___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A method to determine PSTN Call Provider?
- Original Message - From: Gordon Henderson gordon+aster...@drogon.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 21, 2008 11:30 AM Subject: Re: [asterisk-users] A method to determine PSTN Call Provider? On Sun, 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 are you in/want to tell? In the UK it's realtively easy, but it's a big database, and it won't tell you anything about ported numbers eg. from a PSTN to VoIP operator) That's assuming the caller doesn't withold... Gordon Sorry, I'm in the US Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 trunk and will be in the next 1.6 to be branched (probably 1.6.2). -- Tilghman Could it be back-ported to 1.4? Really not ready for 1.6 Thanks, Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record CMD
- Original Message -
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 16, 2008 10:40 AM
Subject: Re: [asterisk-users] Record CMD
On Monday 15 December 2008 18:37:05 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, perhaps of SILENCE, MAXLENGTH,
or POUNDKEY, right? That sounds like a reasonable request.
--
Tilghman
Exactly! but sadly these variables don't seem to exists as far as I can tell
Bart
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[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___
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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, telnet, teletype or CRT terminals for email and what not. I say, end it Bart - Original Message - From: Ira To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 05, 2008 5:03 PM Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] At 09:11 AM 12/5/2008, you wrote: I realize that not everyone sees it that way, but maybe it throws a different perspective in the mix. Personally I rarely read bottom posted messages. I've already read the rest of it and after so many times scrolling to the bottom to see I agree, I rarely bother any more. My loss I know, but my time is worth something too. Ira -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.176 / Virus Database: 270.9.14/1832 - Release Date: 12/5/2008 9:57 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 result
identifier 0 passed in aMYSQL_clear
[Dec 1 12:26:31] WARNING[20425] app_addon_sql_mysql.c: Identifier 4,
identifier_type 2 not found in identifier list
[Dec 1 12:26:31] WARNING[20425] app_addon_sql_mysql.c: Invalid result
identifier 4 passed in aMYSQL_clear
DialPlan:
exten = s,1,Noop()
exten = s,n,MYSQL(Connect connid localhost userid password database)
exten = s,n,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ cnam\ WHERE\
ani=\'${CALLERID(number)}\')
exten = s,n,MYSQL(Fetch fetchid ${resultid} name)
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
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Re: [asterisk-users] MySQL Error Message
Barton Fisher wrote:
Can some tell me what this warnings means?
The dialplan works, but I get these warnings every once in a while:
I'm guessing that some times the caller-id is blank. I got tired of
those errors and did the following before the query:
exten = s,1,GotoIf($[${CALLERID(num)} = ]?2:3)
exten = s,n,Set(CALLERID(all)=Restricted 0)
exten = s,1,Noop()
exten = s,n,MYSQL(Connect connid localhost userid password database)
Doug
Sorry, I removed those lines before I posted dialplan - I do check to see if
there is some value before I write
Here's another case:
[Dec 1 13:48:15] WARNING[14017] app_addon_sql_mysql.c: Identifier 0,
identifier_type 2 not found in identifier list
[Dec 1 13:48:15] WARNING[14017] app_addon_sql_mysql.c: Invalid result
identifier 0 passed in aMYSQL_clear
[Dec 1 13:48:15] VERBOSE[14017] logger.c: [5412254900-inn] [8006848429]
[update-cnam] NAME Updated=[8006848429 800 Service ]
[Dec 1 13:48:15] WARNING[14017] app_addon_sql_mysql.c: Identifier 4,
identifier_type 2 not found in identifier list
[Dec 1 13:48:15] WARNING[14017] app_addon_sql_mysql.c: Invalid result
identifier 4 passed in aMYSQL_clear
[Dec 1 13:48:15] VERBOSE[14017] logger.c: [5412254900-inn] [8006848429]
Database Marketing [5412254900] [8006848429] [800 Service ]
any ideas?
Bart
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[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 http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 made, they are muted. I posted on http://bugs.digium.com/view.php?id=12913 but I have got much notice. I was wondering if you could test this scenario to see if it in fact fails and post your results in bugs? Thanks, Bart___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP
Yep - tried both and combination thereof - The bad effect of inband mode was audio went one way after first press My test app reads back the ANI DNIS at answer (which works), then prompts for more digits. It's suppose to read back whatever is heard. I can see it reading back something, back I don't hear anything. One note: if I press say '111' fast, it might hear '11', but not all digits sadly I'm sure this is a 'bug' as it work perfectly on 1.2, but so far there is no acknowledgement from Developers yet. Not sure how long it should take :( Bart -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Sunday, June 22, 2008 7:36 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP Bart, Did you try the method of inband 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 results with inband, seems it would take digits, but audio goes to 1 way afterwards first push. As far as changing the code per the URL, I think I get 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 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 ignored. ZAP to ZAP connections work perfect. Just so you know, with 1.2 this is not an issue and this issue is keeping me from moving to 1.4. I have a test system setup with a SIP DID to a test IVR system to demonstrate this problem. I can provide full access to these systems for testing. I've placed on Digium bugs but have not received any responses yet. There is nothing in the logs or cli that provides anything meaningful - Below is a call where I press '*' and it is ignored. Hello, here are a few pointers that might help. Are you using RFC2833, inband, info? My guess is 2833, you might want to give inband a try unless you are using a lossy codec. This is pretty interesting and might solve your issue. It seems that by doing this, Asterisk just passes the DTMF as regular audio instead of trying to interpret it. Bookmarked for when I run into this same issue. http://astrecipes.net/index.php?n=248 Linked from this page on the wiki that has more info on your issue. http://www.voip-info.org/wiki/view/Asterisk+DTMF Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP
Yeah, it gets a bit confusing with all the scenario possible - Regardless, you are right I should stay on 1.2 until 1.4 is ready for prime time, but now that 1.6 is out, I'm sure I'm in for a long wait. I reposted my bug again, since I think I may have listed it wrong - it's now http://bugs.digium.com/view.php?id=12913 - Maybe now someone might notice :) Thanks, Steve for your inputs Bart Asterisk has never been good at catching DTMF in rapid succession. I have read in many places that asterisk 1.4 had many changes to DTMF. You contradict yourself below. The bad effect of inband mode was audio went one way after first press and One note: if I press say '111' fast, it might hear '11', but not all digits sadly I suppose that you were using different methods. Try pressing the keys a little slower. Personally, I would just go back to 1.2.X if you cannot get anyone 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 mode was audio went one way after first press My test app reads back the ANI DNIS at answer (which works), then prompts for more digits. It's suppose to read back whatever is heard. I can see it reading back something, back I don't hear anything. One note: if I press say '111' fast, it might hear '11', but not all digits sadly I'm sure this is a 'bug' as it work perfectly on 1.2, but so far there is no acknowledgement from Developers yet. Not sure how long it should take :( Bart -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Sunday, June 22, 2008 7:36 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP Bart, Did you try the method of inband 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 results with inband, seems it would take digits, but audio goes to 1 way afterwards first push. As far as changing the code per the URL, I think I get 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 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 ignored. ZAP to ZAP connections work perfect. Just so you know, with 1.2 this is not an issue and this issue is keeping me from moving to 1.4. I have a test system setup with a SIP DID to a test IVR system to demonstrate this problem. I can provide full access to these systems for testing. I've placed on Digium bugs but have not received any responses yet. There is nothing in the logs or cli that provides anything meaningful - Below is a call where I press '*' and it is ignored. Hello, here are a few pointers that might help. Are you using RFC2833, inband, info? My guess is 2833, you might want to give inband a try unless you are using a lossy codec. This is pretty interesting and might solve your issue. It seems that by doing this, Asterisk just passes the DTMF as regular audio instead of trying to interpret it. Bookmarked for when I run into this same issue. http://astrecipes.net/index.php?n=248 Linked from this page on the wiki that has more info on your issue. http://www.voip-info.org/wiki/view/Asterisk+DTMF Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http
[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 ignored. ZAP to ZAP connections work perfect. Just so you know, with 1.2 this is not an issue and this issue is keeping me from moving to 1.4. I have a test system setup with a SIP DID to a test IVR system to demonstrate this problem. I can provide full access to these systems for testing. I've placed on Digium bugs but have not received any responses yet. There is nothing in the logs or cli that provides anything meaningful - Below is a call where I press '*' and it is ignored. [7147832205-inn] ROUTING TO: CUST 03 [*7142318000*7147832205*] -- Executing [EMAIL PROTECTED]:12] Dial(SIP/innov-09a73f78, Zap/g5/*7142318000*2205*|10|r) in new stack [Jun 19 15:26:15] DEBUG[12160]: chan_zap.c:1949 zt_call: Dialing '*7142318000*2205*' [Jun 19 15:26:15] DEBUG[12160]: chan_zap.c:2025 zt_call: Deferring dialing... -- Called g5/*7142318000*2205* [Jun 19 15:26:15] DEBUG[12160]: chan_zap.c:4378 zt_handle_event: Ignoring wink on channel 97 [Jun 19 15:26:16] DEBUG[12160]: chan_zap.c:4441 zt_handle_event: Sent deferred digit string: T*7142318000*2205 [Jun 19 15:26:19] DEBUG[12160]: chan_zap.c:1452 zt_train_ec: Engaged echo training on channel 97 [Jun 19 15:26:21] DEBUG[12160]: chan_zap.c:1415 zt_enable_ec: Echo cancellation already on -- Zap/97-1 answered SIP/innov-09a73f78 [Jun 19 15:26:30] DTMF[12160]: channel.c:2204 __ast_read: DTMF begin '*' received on SIP/innov-09a73f78 [Jun 19 15:26:30] DTMF[12160]: channel.c:2215 __ast_read: DTMF begin passthrough '*' on SIP/innov-09a73f78 [Jun 19 15:26:30] DEBUG[12160]: chan_zap.c:1050 zt_digit_begin: Started VLDTMF digit '*' [Jun 19 15:26:30] DTMF[12160]: channel.c:2129 __ast_read: DTMF end '*' received on SIP/innov-09a73f78, duration 100 ms [Jun 19 15:26:30] DTMF[12160]: channel.c:2176 __ast_read: DTMF end accepted with begin '*' on SIP/innov-09a73f78 [Jun 19 15:26:30] DTMF[12160]: channel.c:2192 __ast_read: DTMF end passthrough '*' on SIP/innov-09a73f78 [Jun 19 15:26:30] DEBUG[12160]: chan_zap.c:1085 zt_digit_end: Ending VLDTMF digit '*' I'm using: Asterisk Source Version : 1.4.21 Zaptel Source Version : 1.4.11 Libpri Source Version : 1.4.4 Addons Source Version : 1.4.7 Please help, I'm stuck on 1.2 until resolved - Thanks Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP
Steve, I fooled with dtmf mode and it was 2833 - However, got stranger results with inband, seems it would take digits, but audio goes to 1 way afterwards first push. As far as changing the code per the URL, I think I get 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 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 ignored. ZAP to ZAP connections work perfect. Just so you know, with 1.2 this is not an issue and this issue is keeping me from moving to 1.4. I have a test system setup with a SIP DID to a test IVR system to demonstrate this problem. I can provide full access to these systems for testing. I've placed on Digium bugs but have not received any responses yet. There is nothing in the logs or cli that provides anything meaningful - Below is a call where I press '*' and it is ignored. Hello, here are a few pointers that might help. Are you using RFC2833, inband, info? My guess is 2833, you might want to give inband a try unless you are using a lossy codec. This is pretty interesting and might solve your issue. It seems that by doing this, Asterisk just passes the DTMF as regular audio instead of trying to interpret it. Bookmarked for when I run into this same issue. http://astrecipes.net/index.php?n=248 Linked from this page on the wiki that has more info on your issue. http://www.voip-info.org/wiki/view/Asterisk+DTMF Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP
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 results with inband, seems it would take digits, but audio goes to 1 way afterwards first push. As far as changing the code per the URL, I think I get 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 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 ignored. ZAP to ZAP connections work perfect. Just so you know, with 1.2 this is not an issue and this issue is keeping me from moving to 1.4. I have a test system setup with a SIP DID to a test IVR system to demonstrate this problem. I can provide full access to these systems for testing. I've placed on Digium bugs but have not received any responses yet. There is nothing in the logs or cli that provides anything meaningful - Below is a call where I press '*' and it is ignored. Hello, here are a few pointers that might help. Are you using RFC2833, inband, info? My guess is 2833, you might want to give inband a try unless you are using a lossy codec. This is pretty interesting and might solve your issue. It seems that by doing this, Asterisk just passes the DTMF as regular audio instead of trying to interpret it. Bookmarked for when I run into this same issue. http://astrecipes.net/index.php?n=248 Linked from this page on the wiki that has more info on your issue. http://www.voip-info.org/wiki/view/Asterisk+DTMF Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 I could hook-up with someone that's tried this so I could pick your brain about the finer details. Thanks, Bart ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF dropping digits
Hmm, this seems to describe my problem - Thanks, Bart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Tuesday, September 25, 2007 6:41 AM To: asterisk-users@lists.digium.com Subject: 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 = 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 using Asterisk 1.2 and Server is 2.8 Dual Xeon SuperMicro with 2 GB RAM. Any clues what I can do to fix this? Try applying the patch at http://bugs.digium.com/view.php?id=10535 Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 the traffic arrives into a simple IVR script where a message is played. We are using Asterisk 1.2 and Server is 2.8 Dual Xeon SuperMicro with 2 GB RAM. Any clues what I can do to fix this? Bart ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
Thanks Guys... ulaw it is. One more question if you don't mind. If a phase recorded as both .wav and .ulaw in the same folder, which will asterisk pick using Playback(), Read() and Background() since you can't specify the file extension in the command? I thought I change my script to begin 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 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. Any clue on how to verify codec in use during a call? Basically its going to be g711.ulaw for T1 (USA) and g711.alaw for E1 (rest of world) 99.9% of the time. Unless you have something strange or different, I'd record in ulaw for T1. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG5MI9DQNt8rg0Kp4RAgRWAKCL2l8egvLV2Xu3T754KJMzGXrKnQCfboCx aFwrtGNKZ0EbZr176MDZUkY= =HvDo -END PGP SIGNATURE- __ NOD32 2517 (20070910) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com -- 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://icpage.com version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. Any clue on how to verify codec in use during a call? Basically its going to be g711.ulaw for T1 (USA) and g711.alaw for E1 (rest of world) 99.9% of the time. Unless you have something strange or different, I'd record in ulaw for T1. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG5MI9DQNt8rg0Kp4RAgRWAKCL2l8egvLV2Xu3T754KJMzGXrKnQCfboCx aFwrtGNKZ0EbZr176MDZUkY= =HvDo -END PGP SIGNATURE- __ NOD32 2517 (20070910) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com -- 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://icpage.com version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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://icpage.com version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
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. Any clue on how to verify codec in use during a call? Bart Steve Totaro wrote: Michiel van Baak wrote: On 10:28, Sun 09 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 codec the T1 is using. Best to transcode the ivr sounds to the same codec to prevent on-the-fly transcoding by asterisk. The answer is going to ulaw or alaw depending where you live. T1 should most likely be using ulaw so make everything ulaw, end to end. Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 2516 (20070909) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com -- 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://icpage.com version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channels in use?
I'm using version 1.2 and need a method to detect the number of channels in use from inside the dial plan. 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 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://icpage.com version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ADIT 600 CMG = Asterisk question
I've searched but can't find an answer as to how many MGCP paths can a single ADIT/CMG card support? It appears it's only 24 ports, maybe 48. What I'd like to do is install 6 Telco T1's into a single (or more) Adit 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 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://icpage.com version:2.1 end:vcard ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Append Extension number sounds to Voice Mail Message?
Could some provide me a sample code to append the extension number in voice to the beginning of a voice mail message wav file before or after the message is saved? The idea is if the voice mail message wav file arrives from several sources, the listener will hear the 4 digit extension inside 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:714-228-5410 url:http://icpage.com version:2.1 end:vcard ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ADIT 600 CMG = Asterisk question
Actually, these are old D4 SF (non-PRI) circuits - Could your echo be caused by FXO/FXS termination? I wonder if CMG would suffer as much as I believe it would stay 4 wire towards asterisk ? Bart Darren Wright wrote: Are you talking about PRI's? The ADIT's can't handle termination of PRI's, 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: Mon 9/3/2007 11:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ADIT 600 CMG = Asterisk question I've searched but can't find an answer as to how many MGCP paths can a single ADIT/CMG card support? It appears it's only 24 ports, maybe 48. What I'd like to do is install 6 Telco T1's into a single (or more) Adit 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 __ NOD32 2500 (20070903) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com -- 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://icpage.com version:2.1 end:vcard ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 company. Afterwards, they do several circuit tests to the NIU and then to the CSU. At the point they do the pattern test to the CSU, Asterisk will lock up after about 5 seconds of patterns. The only thing left to do is reboot. The lock up can be reproduce with a T1 test set and it happens every time. I'm wondering if anyone else has seen this behavior? And if there is a fix to keep it from happening? I'm using asterisk-1.2.18 zaptel-1.2.17.1 TIA Bart ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 should be first choice is trixbox - it's the best
supported package out there for the newbie - but does not support
Hylafax and asterisk 1.4 (yet) like the other two. They say it's coming :)
Bart
mtest001 wrote:
Hi everybody !
I'm desperately trying to install AsteriskNow Beta 6. I
downloaded the iso file (version x86 32 bits) and burned it,
then I tried on three different computers (from an old Pentium
4 to a brand new HP DL380 2xDual Core) and each time I got the
same error...
Shortly after the installation begins, after the probing of
hardware component, the installer stops with the following
message :
Quote:
Running Anaconda [...]
file /usr/bin/anaconda, line 316, in ?
if (os.path.exists('isys')):
AttributeError: 'module' object has no attribute 'path'
...and then ask to reboot.
Am I the only one to have this error ? I burned two CDs and
tried on three computers ... no luck. It seems to me that
there's something wrong with this iso... Sad
Appreciate your help !
Btw I've got a question... I'm new to Asterisk and until now I
only configured it by editing the text files. I like to have
in my dialplan a macro that sends the caller to the voicemail
if the extension called is not available or does not answer in
15 seconds.
Is it possible to configure such a rule with the GUI of
Asterisk Now ? Is it possible to make it generic for each and
every extension ?
Thank you for your help.
Créez votre adresse électronique [EMAIL PROTECTED]
1 Go d'espace de stockage, anti-spam et anti-virus intégrés.
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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? Bart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 BackGround(Zap/216-1, custom/3566/91_|m|) in new stack Bart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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, custom/3566/91_|m|) in new stack Bart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ${ANSWEREDTIME} Broken on 1.2.13?
Yep, I found about that a few hours after I posted - I guess it's only
use for outbound calls - sad
However, I was able to do some math in the set command using EPOCH to
solve the problem
For those that might seek a solution:
exten = _X.,n,Set(STIME=${EPOCH}) ; save the start time
...do some 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, ${ANSWEREDTIME} is always 0, even on the most
simplest dial plan such as:
Using Asterisk 1.2.13
exten = 77,1,Answer
exten = 77,2,Playback(custom/dax/S300) ; one minute file
exten = 77,3,Noop(${ANSWEREDTIME})
exten = 77,4,Hangup
-- Executing Answer(SIP/5402-b7b45f58, ) in new stack
-- Executing Playback(SIP/5402-b7b45f58, custom/dax/S300) in
new stack
-- Playing 'custom/dax/S300' (language 'en')
-- Executing NoOp(SIP/5402-b7b45f58, ) in new stack
-- Executing Hangup(SIP/5402-b7b45f58, ) in new stack
What gives on this simple thing?
Bart
Slight correction: It is NULL, not 0. Something can't be broken that
was never expected to work or coded to work... ANSWEREDTIME only gets
set by app_dial when you dial something else and it is answered or not
answered.
Joshua Colp
Software Developer
Digium, Inc.
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[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, ) in new stack
-- Executing Playback(SIP/5402-b7b45f58, custom/dax/S300) in new
stack
-- Playing 'custom/dax/S300' (language 'en')
-- Executing NoOp(SIP/5402-b7b45f58, ) in new stack
-- Executing Hangup(SIP/5402-b7b45f58, ) in new stack
What gives on this simple thing?
Bart
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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 everybody involved in providing such great tools ressources!! I'm currently trying to implement a simple voicebox-system. for demonstration purposes, I've successfully connected my cellphone via bluetooth using the current chan_cellphone-patch on the current SVN-version of asterisk. everything seems to work fine so far (great patch!) what I want to achieve: * incoming call arrives * asterisk/cellphone answers * caller is greeted (playback of my-intro) * caller enters an extension * caller is directly forwarded to the voicemail of entered extension here's my dialplan for this scenario: [demo] exten = s,1,Answer ; Answer the line exten = s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds exten = s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds exten = s,n(restart),BackGround(my-intro); Play a congratulatory message exten = s,n,WaitExten(5) ; Wait for an extension to be dialed. exten = 1,1,Voicemail(1001,u) exten = 2,1,Voicemail(1002,u) ... exten = 9,1,Voicemail(1009,u) exten = 10,1,Voicemail(1010,u) exten = 11,1,Voicemail(1011,u) now basically this seems to work - when I'm calling in, I can press 1-9 and am connected to the right mailbox. however, apparantly this is only working for extension with a length of 1! when I try to enter f.e. 11, asterisk seems to get only the first digit and forwards mit to extension 1. somehow it seems only the first digit is processed correctly... I've no idea if this is a basic misunderstanding of the concept (sorry, newbie...), or maybe just a particular problem I'm having with the cellphone (however I tried both nokia 6630 nokia n73 with the same results). any help greatly appreciated, thx again! michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 2217 (20070425) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 examples Bart Timothy McKee wrote: Attempting to talk to an Eagle Telephonics switch at a disaster exercise. Didn't think a plain old EM wink start T1 would be this much of an issue. We finally got the Eagle to accept a call from *, but whilst I can hear the person on the Eagle, they can't hear me. When they initiate a dial out I only get the first 2 digits from their switch... Does anyone have decent sample EM Wink start configs for the Digium cards and * ? Any suggestions on the Eagle side? Has anyone = Timothy McKee VP, Network Services SDN Global +1-704-587-4829 work +1-704-587-4830 NOCC __ NOD32 2216 (20070424) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 2216 (20070424) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ${connid} UPDATE\ caller\ SET\
lastcall=${LASTCALL}\,totalcalls=totalcalls+1\,currentcalls=currentcalls+1\
WHERE\ dnis=\'${IVR-Exten}\'\ AND\ ani=\'${CALLERID(number)}\')
exten = update,n,MYSQL(Clear ${resultid})
exten = update,n,MYSQL(Disconnect ${connid})
Asterisk logs says:
Apr 19 15:50:05 VERBOSE[19740] logger.c: -- Executing
MYSQL(SIP/5400-b7bbfaf0, QUERY resultid 201 UPDATE caller SET
lastcall= 04/18/07 11:12:55, totalcalls= totalcalls+1, currentcalls=
currentcalls+1 WHERE dnis= '7690' AND ani= '5400') in new stack
Apr 19 15:50:05 WARNING[18333] app_addon_sql_mysql.c: Identifier 200,
identifier_type 2 not found in identifier list
Apr 19 15:50:05 WARNING[18333] app_addon_sql_mysql.c: Invalid result
identifier 200 passed in aMYSQL_clear
I understand what the warning message is really saying
Bart
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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 digium quad card as a EM T1 for Faxes to a Hylafax server. The 1st and 2nd spans are working as PRIs. When I start asterisk, the logs show a signaling error and chan_zap.c dies. I also get an error that it can't read the gains but they are the standard shown below. 2.6 kernel, Debian Stable, * 1.2 svn from feb 2007 my procedure: make changes to zaptel.conf zapata.conf rmmod wct4xxp modprobe wct4xxp ztcfg -vv #shows 1+2 span as PRI, 3rd span as EM asterisk -vvc ###Error log logger.c: -- Registered channel 47, PRI Signalling signalling chan_zap.c: Signalling requested on channel 49 is PRI Signalling but line is in E M Immediate signalling chan_zap.c: Unable to register channel '49-72' loader.c: chan_zap.so: load_module failed, returning -1 --ZAPTEL.CONF--- span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #span=2,2,1,esf,b8zs #have tried this way as well span=2,1,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs em=49-72 ---ZAPATA.CONF [channels] language=en usecallerid=yes callerid=asreceived callwaiting=no relaxdtmf=no group=0 callgroup=0 faxdetect=no rxgain=0 txgain=0 echocancel=yes echocancelwhenbridged=yes echotraining=600 jitterbuffers=6 amaflags=billing context=from-pstn switchtype=national signalling=pri_cpe channel = 1-23 group=1 channel = 25-47 ; NEW FAX t1 group=3 signaling=em_w context=from-internal channel = 49-72 Thanks for any tips or glaring oversights on my part. JJ __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 attempt to create
an exten for each bin to update.
Here the method I'd like to use:
exten = sav,n,MYSQL(Connect connid localhost root password dax)
exten = sav,n,MYSQL(QUERY resultid ${connid}UPDATE\ dnislookup\ SET\
${BINID}\ =\ ${NEWPHRASENAME}\ WHERE\ dnis\ =\ ${IVR-Exten})
But I've tried this too:
exten = sav,n,MYSQL(Connect connid localhost root password dax)
exten = sav,n,MYSQL(QUERY resultid ${connid}UPDATE\ dnislookup\ SET\
bin2\ =\ ${NEWPHRASENAME}\ WHERE\ dnis\ =\ ${IVR-Exten})
However, neither one of these saves to new value into the bin2 (or
${BINID}) field.
From the logs:
Apr 16 12:40:05 VERBOSE[13718] logger.c: == Where Field Name = bin2 and
value to update is 2_4643
Apr 16 12:40:05 DEBUG[13718] pbx.c: Launching 'MYSQL'
Apr 16 12:40:05 DEBUG[13718] pbx.c: Launching 'MYSQL'
Apr 16 12:40:05 WARNING[13718] app_addon_sql_mysql.c: aMYSQL_query:
missing some arguments
Apr 16 12:40:05 DEBUG[13718] pbx.c: Launching 'MYSQL'
Apr 16 12:40:05 WARNING[13718] app_addon_sql_mysql.c: Identifier 160,
identifier_type 2 not found in identifier list
Apr 16 12:40:05 WARNING[13718] app_addon_sql_mysql.c: Invalid result
identifier 160 passed in aMYSQL_clear
Apr 16 12:40:05 DEBUG[13718] pbx.c: Launching 'MYSQL'
Can you suggest something?
Bart
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[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 = s,n,MYSQL(Fetch fetchid ${password} password)
exten = s,n,Verbose( == Password found was
[${password}]-[${connid}]-[${fetchid}]-[${resultid}] )
exten = s,n,MYSQL(Clear ${password})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,return
I found no less than 15 ways to write this query on the net searches
(with and without escapes and quotes), now I give up and ask here. I
can do this query from the command and get the correct results.
any ideas?
Thanks, Bart
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Re: [asterisk-users] MySQL query from extensions?
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] app_addon_sql_mysql.c: aMYSQL_fetch:
Invalid result identifier 0 passed
Using this:
exten = s,1,Noop()
exten = s,n,MYSQL(Connect connid localhost root passw0rd 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,MYSQL(Clear ${password})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,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 you
can provide would be helpful.
-- Alex
--
Alex Balashov [EMAIL PROTECTED]
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Re: [asterisk-users] MySQL query from extensions?
Sorry, me again..
I'm at a loss as to why your example worked and mine didn't - I was
using one of the last examples I found during my searches.
Can you tell me when/why I need to use the escape or quotes? Is there
some basic rule to follow? I'm asking because there is
a confusing mix of examples on google search and I'm not sure how to know.
Also, if I wish to expand the query to return additional fields (for
example online owner) How would I add these to query and populate the
variables?
Thanks
Bart
Yossi Ben Hagai wrote:
That's the correct syntax:
exten = s,1,Noop()
exten = s,n,MYSQL(Connect connid localhost root passw0rd dax)
exten = s,n,MYSQL(Query resultid ${connid} SELECT\ password\ FROM\
dnislookup\ WHERE\ dnis=${IVR-Exten})
exten = s,n,MYSQL(Fetch fetchid ${resultid} password)
exten = s,n,MYSQL(Clear ${password})
exten = s,n,MYSQL(Disconnect ${connid})
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] app_addon_sql_mysql.c: aMYSQL_fetch:
Invalid result identifier 0 passed
Using this:
exten = s,1,Noop()
exten = s,n,MYSQL(Connect connid localhost root passw0rd 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,MYSQL(Clear ${password})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,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 you
can provide would be helpful.
-- Alex
--
Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
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[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: Processors 1 Model Pentium III (Katmai) CPU Speed 551.37 MHz Cache Size 512 KB System Bogomips 1103.57 PCI Devices - Bridge: Intel Corporation 82371AB/EB/MB PIIX4 ACPI - Ethernet controller: Intel Corporation 82557/8/9 [Ethernet Pro 100] - Host bridge: Intel Corporation 440BX/ZX/DX - 82443BX/ZX/DX Host bridge - IDE interface: Intel Corporation 82371AB/EB/MB PIIX4 IDE - ISA bridge: Intel Corporation 82371AB/EB/MB PIIX4 ISA - PCI bridge: Intel Corporation 440BX/ZX/DX - 82443BX/ZX/DX AGP bridge - USB Controller: Intel Corporation 82371AB/EB/MB PIIX4 USB - VGA compatible controller: Chips and Technologies F69000 HiQVideo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 7:13 PM Subject: Re: [Asterisk-Users] SYMBOL NETVISION II NP-3010 On 7/15/05, Barton Fisher [EMAIL PROTECTED] wrote: 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 Bart: I purchased some of these a while back for about $30 US and than never got motivated enough, so I can't give any pointers to configuration, except for the actual phone. They seem to be sneaky little devils on the phone for keypad configuration; one of Symbol's cable may be required (it can't be readily made: serial on one end and custom connector that no distributor seems to carry on the other). They are relatively cheap, though. They also only do H.323, so be prepared to play around with that for a bit. -Andy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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") in new stackJul 13
09:56:45 WARNING[1315]: No channel type registered for 'CONSOLE'Jul 13
09:56:45 NOTICE[1315]: Unable to create channel of type 'CONSOLE'Jul 13
09:56:45 VERBOSE[1315]: ==
Everyone is busy/congested at this time
Jul 13 09:56:45 VERBOSE[1315]: -- Executing
Dial("SIP/300-3bd6", "CONSOLE/dsp") in new stackJul 13 09:56:45
WARNING[1315]: No channel type registered for 'CONSOLE'Jul 13 09:56:45
NOTICE[1315]: Unable to create channel of type 'CONSOLE'Jul 13 09:56:45
VERBOSE[1315]: == Everyone is
busy/congested at this timeJul 13 09:56:45 DEBUG[1315]: Exiting with
DIALSTATUS=CHANUNAVAIL.Jul 13 09:56:45 VERBOSE[1315]: -- Executing
Hangup("SIP/300-3bd6", "") in new stackJul 13 09:56:45 VERBOSE[1315]: == Spawn extension (from-internal,
111, 2) exited non-zero on 'SIP/300-3bd6'Jul 13 09:56:45 VERBOSE[1315]: -- Executing
Macro("SIP/300-3bd6", "hangupcall") in new stackJul 13 09:56:45
DEBUG[1315]: Exiting with DIALSTATUS=CHANUNAVAIL.Jul 13 09:56:45
VERBOSE[1315]: --
Executing Hangup("SIP/300-3bd6", "") in new stackJul 13 09:56:45
VERBOSE[1315]: == Spawn
extension (from-internal, 111, 2) exited non-zero on 'SIP/300-3bd6'Jul 13
09:56:45 VERBOSE[1315]:
-- Executing Macro("SIP/300-3bd6", "hangupcall") in new stack
However, I'm not 100% sure the sound card drivers are working.
My question is how can I test the sound card separately from Asterisk using only
the command line? For example, play a file to sound card.
If I know the
sound card is working, I should be able find the reason why I can not connect to
CONSOLE/dsp
TIA
Bart
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[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 *ANI*DNIS*
exten = _,1,NoOp,${CALLERID}
exten = _,2,NoOp,${EXTEN}
exten = _,3,SetVar(CALLFILENAME=${CALLERID}-${TIMESTAMP})
exten = _,4,Monitor(wav,${CALLFILENAME},m)
; problem starts here:
exten = 0099,5,Dial(ZAP/g2,20,r}/*${CALLERID}*${EXTEN}*) ; Plays RBT, but
no ANI DNIS delivery
;exten = 0099,5,Dial(ZAP/g2}/*${CALLERID}*${EXTEN}*) ; works for
ANI DNIS delivery, need RBT added
what am I doing wrong
Bart
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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. 5. Use Telco for timing source in zaptel.conf. Only set Telco as source.4 ports cards only need one source Bart - Original Message - From: Brian West To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, June 24, 2005 10:23 AM Subject: Re: [Asterisk-Users] SpanDSP - Squished Faxes Check the clocking on your T1's if you're using a TDM board GIVE UP NOW those don't do faxing well due to frame slips. Squished faxes are the number one sign of clocking issues on your boards. /b On Jun 23, 2005, at 2:44 PM, Richard Cook wrote: Hello, Has anyone had issues with faxes showing up squished in theTIFF file? Any ideas what could be causing it? -- Richard Cook [EMAIL PROTECTED] T: 705-497-9320 ext 2010 Blank Bkgrd.gif___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was check with eTrust Antivirus [undefined] and found virus free. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure how to show channels numbers for the FXS Ports. This was working properly before I upgraded to TDM400P (was X100P). But now the TE410P LED's are flashing RED Please Help Bart Here is my current Zaptel.conf: # Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 AMI/D4 span=1,0,0,d4,ami em=1-24 # Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 AMI/D4 RED span=2,0,0,d4,ami em=25-48 # Span 3: TE4/0/3 TE410P (PCI) Card 0 Span 3 AMI/D4 RED span=3,0,0,d4,ami em=49-72 # Span 4 TE4/0/4 TE410P (PCI) Card 0 Span 4 AMI/D4 RED span=4,0,0,d4,ami em=73-96 # Span 5: WCTDM/0 Wildcard TDM400P REV E/F Board 1 fxoks=97 fxoks=98 # channel 3, WCTDM, inactive. # channel 4, WCTDM, FXO fxsks=99 And Current Zapata.conf: ; Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 ; This T1 is attached to in-house CUST 3 System ; language=en rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 immediate=no busydetect=no busycount=15 callprogress=no ;relaxdtmf=yes ;callerid=asreceived faxdetect=incoming signalling =em_w group = 2 channel = 1-24 ; Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 ; This T1 is attached to inhouse CUST 10 System ; language=en rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 immediate=no busydetect=no busycount=15 callprogress=no ;relaxdtmf=yes ;callerid=asreceived faxdetect=incoming signalling =em_w group = 3 channel = 25-48 ; Span 3: TE4/0/3 TE410P (PCI) Card 0 Span 3 ; This T1 is attached to WorldCom Local 714 DID's language=en context=from-localt1 ; = rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 immediate=no busydetect=no busycount=15 callprogress=no ;relaxdtmf=yes ;callerid=asreceived faxdetect=incoming signalling =em_w group = 4 channel = 49-72 ; Span 4 TE4/0/4 TE410P (PCI) Card 0 Span 4 ; GBX inbound outbound T1 language=en context=from-tollfree ; = rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 immediate=no busydetect=no busycount=15 callprogress=no ;relaxdtmf=yes ;callerid=asreceived faxdetect=incoming signalling =em_w group = 5 channel = 73-96 ; Span 5: WCTDM/0 Wildcard TDM400P REV E/F Board 1 ; ; Note: this is an extension. Create a ZAP extension in AMP ; for Channel 1 ; signalling=fxo_ks context=from-internal group=1 channel = 97 ; ; Note: this is an extension. Create a ZAP extension in AMP ; for Channel 2 ; signalling=fxo_ks context=from-internal group=1 channel = 98 ; ; channel 3, WCTDM, inactive. ; ; Note: this is a trunk. Create a ZAP trunk in AMP ; for Channel 4 ; signalling=fxs_ks context=from-pstn group=0 channel = 99 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 14, 2005 1:07 PM Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Telecom has had the world by the short hairs for decades so being overcharged for technology that is 100 years old (excluding T1 and other newer stuff of course) is comepletely old hat for most people... That's why we are all here using VoIP right! Besides, they CAN charge it so they WILL charge it. Bummer but it is what it is... Now if I could just get better than $600ish for a PRI in AZ 8) W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Huddleston, Robert Sent: Tuesday, June 14, 2005 12:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Anyone paying over $450 for a T1 is being ripped off... If you are in VA,MD,DC,PA,DE,NJ you can get an integrated VoIP T1 for $300 - $400 and a flat internet t1 for about $400. The integrated VoIP T1 is great because it's handed off as an ethernet - no need for a csu/dsu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Tuesday, June 14, 2005 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Which then presumably leads to higher overselling in the home market since use is presumed lower. Also there are often restriction on the line like no Ips given for servers and no servers allowed. I doubt they really care if we can afford it persay... I think it is just a matter of what pricepoint to what feature set. W There's also the fact that a lot of companies charge LESS for home access than for a business, under the assumption that the business will utilize it more, and/or can afford the higher price. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.7.3/15 - Release Date: 6/14/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was check with eTrust Antivirus [undefined] and found virus free. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 $561 a month + Loops. Is this a good deal? Thanks Bart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P Drops Calls after many touch tones from caller
Ihave a TE410P card with two Telco T1's and two external IVR systems attached. Calls from Telco are routed to proper IVR system based on DNIS (DID) received from Telco using a native bridge. T1's are D4 AMI SF Some IVR applications requires the caller to enter digits using their touch tone phone such as phone number. Not every time, but enough to be annoying asterisk drops the call after about 6 - 10 digits. I've adjusted to busy count to 15 and turn off busy detect, but it still happens. I removed asterisk and attach directly to IVR, it works. What I'd like to do is have asterisk only monitor the port for hang-up, not listen for touch tone if that's what causing the problem Any ideas to try would be appreciated Bart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P Drops Calls after many touch tones fromcaller
well here is an example dial:
exten =
3732,3,Dial(ZAP/g2}/*${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: Wednesday, May 04, 2005 9:18
PM
Subject: RE: [Asterisk-Users] TE410P
Drops Calls after many touch tones fromcaller - [SP]
Do you have dial
command in there with option t or T? Whats the log say right before a call is
dropped ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Barton FisherSent: Wednesday, May 04, 2005 11:13
PMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] TE410P Drops
Calls after many touch tones fromcaller
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 tone phone such as phone number. Not
every time, but enough to be annoying asterisk drops the call after about 6 -
10 digits. I've adjusted to busy count to 15 and turn off busy detect,
but it still happens. I removed asterisk and attach directly to IVR, it
works.
What I'd like to do is have
asterisk only monitor the port for hang-up, not listen for touch tone if
that's what causing the problem
Any ideas to try would be
appreciated
Bart
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[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 http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make Span Port Selection in Round Robinfashion? - [SP]
You Da Man! - Thanks - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 5:01 PM Subject: Re: [Asterisk-Users] How to make Span Port Selection in Round 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was check with eTrust EZ Antivirus v6 6.1.7.0 and found virus free. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 clue what I can do to fix this? Is there any where I can look up these Notices to find meaning? Thanks Bart
[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 dial and use all the options in the demo IVR Icandial to an outside line telephone number Using XLite: - Workingoutside the LAN from WAN Icandial to an outside line telephone number - but disconnects after 5 seconds Using IP Phone: I can dial into the demo IVR Application.However, once connected,asterisk seems to ignore any button presses. I candial though the FXO card to any telephone number and talk both ways - Touch tones can be heard from IP Phone. Phone Line: If I dial the number associated with the FXO Card - it does answer and I can use the demo - all functions work On the Console: Repeated message: NOTICE[1125329600]: chan_sip.c:5405 handle_request: Registration from "sip:[EMAIL PROTECTED]' failed for '192.168.1.197' DEBUG[1125329600]: chan_sip.c:574 __sip_ack: Stopping retransmission on 'and some long numbers' Thanks Bart
