Re: [asterisk-users] How strip +1 from caller id on inbound call
On 8/14/07, James Collier [EMAIL PROTECTED] wrote:
What if it is an international call? Then your callerID won't work.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de C F
Enviado el: lunes, 13 de agosto de 2007 3:21
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] How strip +1 from caller id on inbound call
After rereading this post, I belive that this could also be
acomplished doing this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than 10 digits grab the last 10 digits of the CIDNUM
exten = _X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):-10})
;this grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
On 8/12/07, C F [EMAIL PROTECTED] wrote:
you can do like this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than grab the last 10 digits of the CIDNUM
exten =
_X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):$[${LEN(${CALLERID(num)})}-
10]});this
grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
Hope this helps.
On 8/12/07, voiplist [EMAIL PROTECTED] wrote:
From some of our telecom providers we get the caller-id as:
NXXNXX
From others we get:
+1NXXNXX
We are trying to standardize the way our caller-id comes in so we
would like to strip off the +1 from the inbound caller id.
Can anyone offer any suggestions?
I have tried:
;exten = +18664918575,1,Set(CALLERID(all)=${CALLERIDNAME}
${CALLERIDNUM:2})
but it just yacks..
Thanks in advance for any help.
Regards,
Todd R.
--
Prestige Messaging
Live Answering Services
SIP or Toll-Free Connectivity
Light Accounts From $14.95/mo
http://www.PrestigeMessaging.com
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Yeah, that's a problem.. I guess I may be stuck with the +1 because it
more right than wrong. Maybe I can just add a +1 to the others which
will be much easier and make it all standard..
Thanks for that, I am sure I would have run into it eventually but
it's always nice to not just run into things :-)
Regards,
Todd R.
--
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Live Answering Services
SIP or Toll-Free Connectivity
Light Accounts From $14.95/mo
http://www.PrestigeMessaging.com
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Re: [asterisk-users] How strip +1 from caller id on inbound call
Am Sonntag, den 12.08.2007, 21:16 -0400 schrieb C F:
you can do like this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than grab the last 10 digits of the CIDNUM
exten =
_X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):$[${LEN(${CALLERID(num)})}-10]});this
grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
Argh! You do not ever get international calls, do you? (Well, Canada
does not count here for obvious reasons)
The clean solution to the question
I get some calls with a leading +1. If that is the case, how do I
strip that off?
is of course
If the CALLERID(num) starts +1, re-set it to the same value, offset 2:
...
exten = _X.,n,GoSubIf($[${CALLERID(num):0:2} = +1]?strip1)
...
exten = _X.,n(strip1),Set(CALLERID(num)=${CALLERID(num):2})
exten = _X.,n,Return()
Which leaves international calls for themselves. Of course you still
could replace the leading + for all other numbers by 011, if you
like.
Your code would probably handle
+12125551212
correctly, would work OK with
+495924236
(which might or might not be one of the old, short numbers still present
in some places in Germany), leaving it intact, but not with
+4916177554224
which would be remapped to a Boston MA number (actually a Cingular cell
phone number) instead of mapping it to a german mobile phone.
Variable handling (offset et al) is documented on
http://www.voip-info.org/wiki/view/Asterisk+variables
BR
Anselm
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Re: [asterisk-users] Dialplan loop
I do something like this. But I am using Realtime and 1.4
context185 1 Set caller_num=${CALLERID(num)}
context185 2 SetMusicOnHold default
context185 3 Playbacksilence2
context185 4 AGI context1.php
context185 5 Set __FWCOUNT=0${FWCOUNT}
context185 6 Set __FWCOUNT=$[${FWCOUNT}+1]
context185 7 GotoIf $[${FWCOUNT}10]?10
context185 8 DialSIP/ext1SIP/ext2|30|m
context185 9 Hangup
context185 10 Congestion
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Anselm
Martin Hoffmeister
Enviado el: domingo, 12 de agosto de 2007 12:35
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] Dialplan loop
Am Donnerstag, den 09.08.2007, 20:12 -0500 schrieb David Bandel:
Folks,
I'm trying to implement a simple loop in a dialplan. The object is to
set a counter, run through some IVR options, increment the counter,
return to the start, then finally fall through to an operator or
voicemail.
exten = s,n,Set(loop = 0)
...
exten = s,n,Set(loop = $[${loop} + 1])
The above loop increment doesn't work. The error message is:
WARNING[14490]: ast_expr2.fl:398 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '+', expecting $end; Input:
+ 1
^
WARNING[14490]: ast_expr2.fl:402 ast_yyerror: If you have questions,
please refer to doc/channelvariables.txt in the asterisk source.
Try removing extra space characters around the =. Very similar example
from my dialplan
exten = _2XX,n,Set(I=1)
...
exten = _2XX,n,Set(EXTR=$[${I} + 1])
Works fine. Also assigning a variable a new value based on the old value
works OK here (although not calculated, but concatenated):
exten = _2XX,n,Set(D=${D}SIP/sip501)
I am using Asterisk 1.2 here, but I remember similar errors with stray
characters.
BR
Anselm
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Re: [asterisk-users] How to use OpenVPN with Asterisk
AB == Alan Bunch [EMAIL PROTECTED] writes: AB Just another OpenVPN data point, and not Asterisk related but here AB goes. I run 15 users over a DSL link on one end and a Internet T1 AB on the other with OpenVPN and it just rocks. The road warrior AB setup is down to running one script to create a file, email file AB to user and the user run one script. Which OS do you use at the client end? /Benny ___ --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] IVR and MySQL
Hi Does somebody know if I can save the answers made by the caller person on the IVR menu in a MySQL Database? If yes, can I save the CallerID as well? Thanks, Fabio Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ --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] Faulty voicemail
Hi All,
I was made aware today that some of my calls coming in are not going to
voicemail... Below are some logs, and the macro that should run on the
incoming_pstn context for that extension. I can see that theres a
non-zero exit before it gets to voicemail, but I've no idea why. In
this case theres 2 SIP clients to sim-call. On other occasions it works
fine. In the CDR logs, I can see NO ANSWER and ANSWERED - what
would be there if voicemail answers?
Asterisk: 1.2.23
[macro-ext-group-home]
; ${ARG1} - Virtual Extension (e.g. 2005)
exten =
s,1,ExecIF($[${RECORDSIP}=TRUE],Monitor,wav|${TIMESTAMP}-${CALLERID(
num)}-${MACRO_EXTEN}-${UNIQUEID}.WAV)
exten =
s,2,Dial(SIP/2${ARG1:-2}SIP/4${ARG1:-2}SIP/6${ARG1:-2},${OFFICE_TIMEOU
T},rw)
exten = s,3,Voicemail(u${ARG1})
exten = s,103,Voicemail(u${ARG1})
The call logs show:
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-1
,SIP/600-08e0b990,Dial,SIP/200SIP/400SIP/600|15|rw,2007-08-14
08:49:16,,2007-08-14 08:49:18,2,0,NO ANSWER,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-2
,SIP/600-08e19d58,Dial,SIP/200SIP/400SIP/600|15|rw,2007-08-14
08:49:46,,2007-08-14 08:49:56,10,0,NO ANSWER,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-1
,SIP/600-08e0b990,VoiceMail,u2000,2007-08-14
08:50:37,2007-08-14 08:50:52,2007-08-14
08:51:00,23,8,ANSWERED,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-2
,SIP/600-08e19d58,Dial,SIP/200SIP/400SIP/600|15|rw,2007-08-14
08:51:35,,2007-08-14 08:51:45,10,0,NO ANSWER,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-1
,SIP/600-08e0b990,VoiceMail,u2000,2007-08-14
08:52:19,2007-08-14 08:52:34,2007-08-14
08:52:38,19,4,ANSWERED,DOCUMENTATION
And my messages log for that time (for one failed call) shows:
ubiphone*CLI
-- Accepting AUTHENTICATED call from 193.111.200.135:
requested format = alaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw|alaw),
priority = mine
ubiphone*CLI
-- Executing Macro(IAX2/ubigradin-2, ext-group-home|2000) in new
stack
-- Executing ExecIf(IAX2/ubigradin-2,
0|Monitor|wav|20070814-085135-07xx-2000-1187077895.3392.WAV)
in new stack
-- Executing Dial(IAX2/ubigradin-2,
SIP/200SIP/400SIP/600|15|rw) in new stack
ubiphone*CLI
-- Called 200
Aug 14 08:51:35 NOTICE[30952]: app_dial.c:1076 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
-- Called 600
ubiphone*CLI
-- SIP/600-08e19d58 is ringing
ubiphone*CLI
-- SIP/200-08e0b990 is ringing
ubiphone*CLI
== Spawn extension (macro-ext-group-home, s, 2) exited non-zero on
'IAX2/ubigradin-2' in macro 'ext-group-home'
== Spawn extension (macro-ext-group-home, s, 2) exited non-zero on
'IAX2/ubigradin-2'
-- Hungup 'IAX2/ubigradin-2'
Adrian Marsh
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Re: [asterisk-users] Asterisk Manager to Record Greetings
Anselm Martin Hoffmeister wrote:
I did something similar using multiple records in a row.
Something like
exten = 931,1,Answer()
exten = 931,2,Wait(2)
exten = 931,3,Set(E=1000)
exten = 931,4,Playback(beep)
exten = 931,5,Set(E=$[${E} + 1])
exten = 931,6,Record(/tmp/asterisk-recording-${E:1})
exten = 931,7,Playback(/tmp/asterisk-recording-${E:1})
exten = 931,8,Wait(2)
exten = 931,9,Goto(4)
This will loop: beep, record until # pressed, replay, wait, beep...
The files will be written with ascending numbers starting 001. Move
them to another place before doing the next recording session.
Couldn't you use %d instead of settup up variable E?
ie.
exten = 931,1,Answer()
exten = 931,2,Wait(2)
exten = 931,3,Playback(beep)
exten = 931,4,Record(/tmp/asterisk-recording-%d.wav)
exten = 931,5,Playback(${RECORDED_FILE})
exten = 931,6,Goto(2)
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Re: [asterisk-users] IVR and MySQL
Hi Fabio, of course that you can. One way to do it is working with app MYSQL(), where you will put your sql as argumment. read more in http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL good luck, Thiago Maluf Resende. 2007/8/14, Fabio Ardeola [EMAIL PROTECTED]: Hi Does somebody know if I can save the answers made by the caller person on the IVR menu in a MySQL Database? If yes, can I save the CallerID as well? Thanks, Fabio Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ --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 -- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] ___ --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] How strip +1 from caller id on inbound call
If you want remove in CALLERID.
you can remove it this way:
exten= _X./_+1X.,1, Set()
ok?
good luck!
Thiago Maluf.
2007/8/14, Anselm Martin Hoffmeister [EMAIL PROTECTED]:
Am Sonntag, den 12.08.2007, 21:16 -0400 schrieb C F:
you can do like this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than grab the last 10 digits of the CIDNUM
exten =
_X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):$[${LEN(${CALLERID(num)})}-10]});this
grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
Argh! You do not ever get international calls, do you? (Well, Canada
does not count here for obvious reasons)
The clean solution to the question
I get some calls with a leading +1. If that is the case, how do I
strip that off?
is of course
If the CALLERID(num) starts +1, re-set it to the same value, offset 2:
...
exten = _X.,n,GoSubIf($[${CALLERID(num):0:2} = +1]?strip1)
...
exten = _X.,n(strip1),Set(CALLERID(num)=${CALLERID(num):2})
exten = _X.,n,Return()
Which leaves international calls for themselves. Of course you still
could replace the leading + for all other numbers by 011, if you
like.
Your code would probably handle
+12125551212
correctly, would work OK with
+495924236
(which might or might not be one of the old, short numbers still present
in some places in Germany), leaving it intact, but not with
+4916177554224
which would be remapped to a Boston MA number (actually a Cingular cell
phone number) instead of mapping it to a german mobile phone.
Variable handling (offset et al) is documented on
http://www.voip-info.org/wiki/view/Asterisk+variables
BR
Anselm
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--
THIAGO MALUF RESENDE
Consultor Voip e Programador WEB (Voip Developer and Web Developer)
Tel: +55 21 86042100
e-mail: [EMAIL PROTECTED]
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Re: [asterisk-users] IVR and MySQL
On 8/14/07, Thiago Maluf [EMAIL PROTECTED] wrote: Hi Fabio, of course that you can. One way to do it is working with app MYSQL(), where you will put your sql as argumment. read more in http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL That's possible, but i wouldn't recommend on large production system. Using MySQL you would need to connect and disconnect all the time, and it takes resources.. I would suggest to append that info to CDR userfield (if you are storing your CDR in MySQL), and run periodically some script that extracts them. Of course it's more complex, but that would be my way. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ --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] BLF with Aastra
I have a 536i expansion module attached to a 57i-CT. The BLF lights on the 536i will light up and work fine for a while... however after a bit they seem to loose their ability to see if someone is on a phone. They still work to dial, if I try to dial, however, they don't light up when someone makes a call, or if their phone rings. If I reboot the phone, the lights start working again (for a while). Any ideas? Asterisk problem? Switch problem? Aastra problem? ___ --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] IVR and MySQL
On 8/14/07, Atis [EMAIL PROTECTED] wrote: That's possible, but i wouldn't recommend on large production system. Using MySQL you would need to connect and disconnect all the time, and it takes resources.. I would suggest to append that info to CDR userfield (if you are storing your CDR in MySQL), and run periodically some script that extracts them. Of course it's more complex, but that would be my way. If the data you wish to store is more complex than stuffing in the CDR userfield would allow, you can always call out to an AGI which can write the data to whatever file format you want for later loading into a database. If you used FastAGI and a pre-forking AGI server model, you could even take the database connection hit when the AGI server starts. The per-call cost would then be the cost to establish the socket connection to the AGI server from Asterisk, the cost to perform the SQL inserts over an established database connection, plus whatever other calculation or transformation you needed to do before doing the insert. That architecture would hold up under a fairly large load. Perl's Asterisk::FastAGI framework lets you specify the number of pre-forked children to launch, plus you can tell each child to exit (spawning a replacement for the pool) after processing a certain number of transactions. It's very similar to the Apache prefork model. -- 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
Re: [asterisk-users] BLF with Aastra
On 8/14/07, Matt [EMAIL PROTECTED] wrote: I have a 536i expansion module attached to a 57i-CT. The BLF lights on the 536i will light up and work fine for a while... however after a bit they seem to loose their ability to see if someone is on a phone. They still work to dial, if I try to dial, however, they don't light up when someone makes a call, or if their phone rings. If I reboot the phone, the lights start working again (for a while). Does 'sip show subscriptions' indicate that the 57i is still subscribed to the extension for updates? If not, you might have to do a test with 'sip debug peer aastraname' to confirm that the subscription is being made properly on phone startup and not being removed by the phone in response to some state change. A quick glance at chan_sip.c indicates that if a user agent tries to subscribe with an expiry time greater than 'maxexpiry' from sip.conf(default 3600 seconds), the subscription expiry in Asterisk will be silently changed to whatever the allowed maximum is. So if the Aastra is trying to subscribe for say 3 hours and Asterisk doesn't allow subscriptions greater than one hour, then notify messages will stop being sent after one hour until the Aatra re-subscribes. I haven't delved in very deep, so I can't tell if the response to the UA indicates the actual expiry Asterisk used, but even so you'd have to be certain that the Aastra respects an expiry in the response that differs from what it asked for. When you're doing the debug (hopefully on a quiet system), watch the phone boot, then use 'sip show subscriptions' to get the call-id of the subscription. Then watch for console messages indicating that the call has been destroyed (which should come at the 1 hour mark or whatever time the Aastra used for it's subscription length. -- 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
[asterisk-users] Maximum retries for seqno 102 when re-inviting.
We have an interesting issue: One of our providers has two softswitches. Calls coming from the first one are handled fine by asterisk, calls coming from the second one and going through the first one are euhm... dropped half a second into the RTP stream. I have opened a ticket at Digium for it: http://bugs.digium.com/view.php?id=10449 The output of sip debug is funny from line 366, where it is transmitting and re-transmitting a lot of re-invites back to the softswitch with CSeq 102. Has somebody else seen this behaviour, and know how to resolve it? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog/ ___ --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] PRI Question
You can eliminate the set CallerID line. This will just set the
variable back to itself. Asterisk will pass the callerid from one
span to the next.
You can use a GotoIF to set the callerid to something else if it is
blank or marked as Private:
exten = s,1,GoToIf($[${CALLERID(num)} = ]?2:3)
exten = s,2,Set(CALLERID(num)=00)
If the CallerID number is blank go to 2 else go to 3.
I wonder if asterisk or the norstar system is holdng on to that last
callerid number on the channel?
The only time you may want to set callerid is when your Norstar dials
out through Asterisk:
[norstar]
; This context is where all incoming calls from the norstar are placed
; Basically take the call from the norstar and bridge it over to the first
; available line on the bottom of the T1 going to TimeWarner.
exten = _1900XXX,1,Playback(cannot-complete-as-dialed)
exten = _1900XXX,2,Hangup()
exten = _1X.,1,GoToIf($[${CALLERID(num)} = ]?2:3)
exten = _1X.,2,Set(CALLERID(num)=511212)
exten = _1X.,3,NoOp(${CALLERID(num)})
exten = _1X.,4,Dial(${PRITRUNK}/${EXTEN},300,tD())
exten = _1X.,5,Hangup()
exten = _X.,1,GoToIf($[${CALLERID(num)} = ]?2:3)
exten = _X.,2,Set(CALLERID(num)=511212)
exten = _X.,3,NoOp(${CALLERID(num)})
exten = _X.,4,Dial(${PRITRUNK}/${EXTEN},300,)
exten = _X.,5,Hangup()
exten = i,1,Answer()
exten = i,n,Wait(1)
exten = i,n,Playback(cannot-complete-as-dialed)
exten = i,n,Playback(please-contact-tech-supt)
exten = i,n,Hangup()
On 8/9/07, Mike Lynchfield [EMAIL PROTECTED] wrote:
hmm from what i have seen this is not supposed to be.. the info is still
there but should not be used in case of privacy..
zap show channels always show last info till a span refresh.. but the
privacy should indeed replace those with Privacy.
Maybe it could be a bug ,
On 8/9/07, Jeremy Mann [EMAIL PROTECTED] wrote:
I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco,
Span 2 sends to my existing phone system(Nortel).
My Span1 gets sent to the context from-pri, detailed here:
[from-pri]
exten = _49XX,1,Set(CALLERID(all)=${CALLERID(all)})
exten = _49XX,2,Dial(Zap/g2/${EXTEN},,twk)
exten = _49XX,3,Congestion()
exten = _49XX,4,Set(CALLERID(all)=)
exten = _49XX,5,Hangup()
exten = _49XX,103,Congestion()
exten = _49XX,104,Set(CALLERID(all)=)
exten = _49XX,105,Hangup()
exten = h,1,Set(CALLERID(all)=)
exten = h,2,Hangup()
I'm receiving caller ID fine, and setting it on the outgoing channel the
same I received it, is my logic above wrong? Will Asterisk natively pass
through the caller ID, or is there a better way to set it?
The reason I ask, is that calls that are not coming in with CLID(blocked
or private) are showing up as the same number that was previously answered
on that channel.
Thanks.
Using Asterisk 1.4 FYI.
This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of federal
or state law and regulations. If you have received this information in
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Re: [asterisk-users] CDR-CSV Processing
There is an example in the asterisk gui (trunk/1.4/asterisknow) that has cdr-csv parsing. You could check that, and even use the javascript to generate reports, integrate it into a little bit of php and ezPDF generation and bam, you have some reports. The cdr viewer in the gui is very useful also. -bk - Original Message - From: Paul Hales [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 13, 2007 10:31:34 PM (GMT-0800) America/Tijuana Subject: Re: [asterisk-users] CDR-CSV Processing http://areski.net/asterisk-stat-v2/about.php ??? PaulH On Mon, 2007-08-13 at 08:49 -0500, Jeremy Mann wrote: Does anyone have any tools to process CDR-CSV files into reports? I don’t have anything specific in mind, I’d just like some reporting examples so I don’t have to reinvent the wheel. __ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 ___ --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] PRI Question
Good idea! It's working great. I also like your local vs LD logic, much
simpler to do than NXXNXX or 1NXXNXX.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck
Sent: Tuesday, August 14, 2007 8:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Question
You can eliminate the set CallerID line. This will just set the
variable back to itself. Asterisk will pass the callerid from one
span to the next.
You can use a GotoIF to set the callerid to something else if it is
blank or marked as Private:
exten = s,1,GoToIf($[${CALLERID(num)} = ]?2:3)
exten = s,2,Set(CALLERID(num)=00)
If the CallerID number is blank go to 2 else go to 3.
I wonder if asterisk or the norstar system is holdng on to that last
callerid number on the channel?
The only time you may want to set callerid is when your Norstar dials
out through Asterisk:
[norstar]
; This context is where all incoming calls from the norstar are placed
; Basically take the call from the norstar and bridge it over to the first
; available line on the bottom of the T1 going to TimeWarner.
exten = _1900XXX,1,Playback(cannot-complete-as-dialed)
exten = _1900XXX,2,Hangup()
exten = _1X.,1,GoToIf($[${CALLERID(num)} = ]?2:3)
exten = _1X.,2,Set(CALLERID(num)=511212)
exten = _1X.,3,NoOp(${CALLERID(num)})
exten = _1X.,4,Dial(${PRITRUNK}/${EXTEN},300,tD())
exten = _1X.,5,Hangup()
exten = _X.,1,GoToIf($[${CALLERID(num)} = ]?2:3)
exten = _X.,2,Set(CALLERID(num)=511212)
exten = _X.,3,NoOp(${CALLERID(num)})
exten = _X.,4,Dial(${PRITRUNK}/${EXTEN},300,)
exten = _X.,5,Hangup()
exten = i,1,Answer()
exten = i,n,Wait(1)
exten = i,n,Playback(cannot-complete-as-dialed)
exten = i,n,Playback(please-contact-tech-supt)
exten = i,n,Hangup()
On 8/9/07, Mike Lynchfield [EMAIL PROTECTED] wrote:
hmm from what i have seen this is not supposed to be.. the info is still
there but should not be used in case of privacy..
zap show channels always show last info till a span refresh.. but the
privacy should indeed replace those with Privacy.
Maybe it could be a bug ,
On 8/9/07, Jeremy Mann [EMAIL PROTECTED] wrote:
I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco,
Span 2 sends to my existing phone system(Nortel).
My Span1 gets sent to the context from-pri, detailed here:
[from-pri]
exten = _49XX,1,Set(CALLERID(all)=${CALLERID(all)})
exten = _49XX,2,Dial(Zap/g2/${EXTEN},,twk)
exten = _49XX,3,Congestion()
exten = _49XX,4,Set(CALLERID(all)=)
exten = _49XX,5,Hangup()
exten = _49XX,103,Congestion()
exten = _49XX,104,Set(CALLERID(all)=)
exten = _49XX,105,Hangup()
exten = h,1,Set(CALLERID(all)=)
exten = h,2,Hangup()
I'm receiving caller ID fine, and setting it on the outgoing channel the
same I received it, is my logic above wrong? Will Asterisk natively pass
through the caller ID, or is there a better way to set it?
The reason I ask, is that calls that are not coming in with CLID(blocked
or private) are showing up as the same number that was previously answered
on that channel.
Thanks.
Using Asterisk 1.4 FYI.
This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
affiliates hereby claim all applicable privileges related to this
information.
--
This message has been scanned for viruses and
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believed to be clean.
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Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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IAXTEL: 17002871718
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Re: [asterisk-users] How strip +1 from caller id on inbound call
I just use
exten = +12564286115,1,Goto(${EXTEN:1})
exten = 12564286115,1,noop(It worked.)
I believe that should work
-bk
- Original Message -
From: Thiago Maluf [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, August 14, 2007 4:25:57 AM (GMT-0800) America/Tijuana
Subject: Re: [asterisk-users] How strip +1 from caller id on inbound call
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Re: [asterisk-users] AGI answering the channel even though I neverasked it to
On Aug 13, 2007, at 4:37 PM, Martin Smith wrote: See http://www.asterisk.org/doxygen/1.4/ res__agi_8c.html#c631d48f46d51d4b057 b31807baa1f10 The AGI application will answer the channel if it isn't already answered. You probably need to do whatever you want to do in the dialplan, and keep using DeadAGI. Excellent information. That's what I spent an hour or so unsuccessfully looking for ;) Thank you very much. Now I just have to figure out how to do a database lookup without answering the channel, as that seems to indicate that the AGI is going to answer regardless of whether a play progress tones or not from the AGI. Daryl ___ --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] IVR and MySQL
James / Atis / Thiago Let say that the user entry during the call is a reference number of a house to rent. Would be possible to check if the reference number is a valid entry on the MySQL database and then base on its answer define the next menu item on the IVR menu. Thanks, Fabio --- James FitzGibbon [EMAIL PROTECTED] wrote: On 8/14/07, Atis [EMAIL PROTECTED] wrote: That's possible, but i wouldn't recommend on large production system. Using MySQL you would need to connect and disconnect all the time, and it takes resources.. I would suggest to append that info to CDR userfield (if you are storing your CDR in MySQL), and run periodically some script that extracts them. Of course it's more complex, but that would be my way. If the data you wish to store is more complex than stuffing in the CDR userfield would allow, you can always call out to an AGI which can write the data to whatever file format you want for later loading into a database. If you used FastAGI and a pre-forking AGI server model, you could even take the database connection hit when the AGI server starts. The per-call cost would then be the cost to establish the socket connection to the AGI server from Asterisk, the cost to perform the SQL inserts over an established database connection, plus whatever other calculation or transformation you needed to do before doing the insert. That architecture would hold up under a fairly large load. Perl's Asterisk::FastAGI framework lets you specify the number of pre-forked children to launch, plus you can tell each child to exit (spawning a replacement for the pool) after processing a certain number of transactions. It's very similar to the Apache prefork model. -- 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 Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center. http://autos.yahoo.com/green_center/ ___ --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] Faulty voicemail
Adrian Marsh wrote:
Hi All,
I was made aware today that some of my calls coming in are not going to
voicemail... Below are some logs, and the macro that should run on the
incoming_pstn context for that extension. I can see that theres a
non-zero exit before it gets to voicemail, but I've no idea why. In
this case theres 2 SIP clients to sim-call. On other occasions it works
fine. In the CDR logs, I can see NO ANSWER and ANSWERED - what
would be there if voicemail answers?
Asterisk: 1.2.23
[macro-ext-group-home]
; ${ARG1} - Virtual Extension (e.g. 2005)
exten =
s,1,ExecIF($[${RECORDSIP}=TRUE],Monitor,wav|${TIMESTAMP}-${CALLERID(
num)}-${MACRO_EXTEN}-${UNIQUEID}.WAV)
exten =
s,2,Dial(SIP/2${ARG1:-2}SIP/4${ARG1:-2}SIP/6${ARG1:-2},${OFFICE_TIMEOU
T},rw)
exten = s,3,Voicemail(u${ARG1})
exten = s,103,Voicemail(u${ARG1})
The call logs show:
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-1
,SIP/600-08e0b990,Dial,SIP/200SIP/400SIP/600|15|rw,2007-08-14
08:49:16,,2007-08-14 08:49:18,2,0,NO ANSWER,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-2
,SIP/600-08e19d58,Dial,SIP/200SIP/400SIP/600|15|rw,2007-08-14
08:49:46,,2007-08-14 08:49:56,10,0,NO ANSWER,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-1
,SIP/600-08e0b990,VoiceMail,u2000,2007-08-14
08:50:37,2007-08-14 08:50:52,2007-08-14
08:51:00,23,8,ANSWERED,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-2
,SIP/600-08e19d58,Dial,SIP/200SIP/400SIP/600|15|rw,2007-08-14
08:51:35,,2007-08-14 08:51:45,10,0,NO ANSWER,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-1
,SIP/600-08e0b990,VoiceMail,u2000,2007-08-14
08:52:19,2007-08-14 08:52:34,2007-08-14
08:52:38,19,4,ANSWERED,DOCUMENTATION
And my messages log for that time (for one failed call) shows:
ubiphone*CLI
-- Accepting AUTHENTICATED call from 193.111.200.135:
requested format = alaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw|alaw),
priority = mine
ubiphone*CLI
-- Executing Macro(IAX2/ubigradin-2, ext-group-home|2000) in new
stack
-- Executing ExecIf(IAX2/ubigradin-2,
0|Monitor|wav|20070814-085135-07xx-2000-1187077895.3392.WAV)
in new stack
-- Executing Dial(IAX2/ubigradin-2,
SIP/200SIP/400SIP/600|15|rw) in new stack
ubiphone*CLI
-- Called 200
Aug 14 08:51:35 NOTICE[30952]: app_dial.c:1076 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
-- Called 600
ubiphone*CLI
-- SIP/600-08e19d58 is ringing
ubiphone*CLI
-- SIP/200-08e0b990 is ringing
ubiphone*CLI
== Spawn extension (macro-ext-group-home, s, 2) exited non-zero on
'IAX2/ubigradin-2' in macro 'ext-group-home'
== Spawn extension (macro-ext-group-home, s, 2) exited non-zero on
'IAX2/ubigradin-2'
-- Hungup 'IAX2/ubigradin-2'
Adrian Marsh
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Exited non-zero here looks like the caller hungup before going to
voicemail, the caller is the one thing you can't control.
Anthony
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[asterisk-users] DTMF on Bridged ZAP call
Should asterisk be intercepting DTMF on a bridged ZAP call? If so, how do I disable it recognizing #, as it's hanging up my users when they try to enter #. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] IVR and MySQL
On 8/14/07, Fabio Ardeola [EMAIL PROTECTED] wrote: Let say that the user entry during the call is a reference number of a house to rent. Would be possible to check if the reference number is a valid entry on the MySQL database and then base on its answer define the next menu item on the IVR menu. If you want to do something like that, you can either use the MYSQL function (with the attendant issues of connecting/reconnecting/etc.) or put all of the functionality in an AGI script. Since AGI can both receive information from and send commands to Asterisk, you can do pretty much anything you can code. There are programming frameworks for AGI for Perl, PHP, Java, and you could even do it in shellscript if you want. The communication channel between Asterisk and the script is stdin/stdout, so you're not restricted at all. Using AGI does make the the integrity of your system depend on an external component (i.e. if you're using FastAGI and the agi server goes down, your calls will just return immediately to the dialplan), but when you need to do something that doesn't fit intuitively into the Asterisk dialplan, I find it's the way to go. -- 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
Re: [asterisk-users] Patent issues, what features we can't use?
Shouldn't you ask your attorney these questions? Any answers you receive here will not legally protect you. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan Zakaria Sent: Monday, August 13, 2007 7:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Patent issues, what features we can't use? Hi everybody, As the Asterisk community is getting larger and larger, I was wondering that the features which are provided in Asterisk and are programmed by the open source community under GPL, or GUIs like FreePBX which also come loaded with wonderful features and uses same Asterisk, are they anywhere violating any patent laws? Most of the features work the same way as Nortel, Avaya and other PBX systems. Is there anyone who owns these features and will come one day to claim his royalties? When I deploy an asterisk soultion for a customer, is there any violation of any patent or copyright laws anywhere? Of if I use my own Asterisk server to provide services to some customers, am I violating any patent laws by not paying the royalties to some patent owners? I heard people saying that IVR technology is patented and google search for patents also say so. But we all are using IVR for ourselves and our customers without paying royalties to anyone. But when it comes to using g729, all of a sudden royalty issue comes in. So what is right to use and what is not? -- Zeeshan A Zakaria ___ --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] Faulty voicemail
Hmm... He swears he heard a voice saying he'd dialed the number
incorrectly.. But that's no-where in the dialplan, and I do see the
incoming calls correctly for the times he's saying..
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: 14 August 2007 15:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Faulty voicemail
Adrian Marsh wrote:
Hi All,
I was made aware today that some of my calls coming in are not going
to
voicemail... Below are some logs, and the macro that should run on
the
incoming_pstn context for that extension. I can see that theres a
non-zero exit before it gets to voicemail, but I've no idea why. In
this case theres 2 SIP clients to sim-call. On other occasions it
works
fine. In the CDR logs, I can see NO ANSWER and ANSWERED - what
would be there if voicemail answers?
Asterisk: 1.2.23
[macro-ext-group-home]
; ${ARG1} - Virtual Extension (e.g. 2005)
exten =
s,1,ExecIF($[${RECORDSIP}=TRUE],Monitor,wav|${TIMESTAMP}-${CALLERID(
num)}-${MACRO_EXTEN}-${UNIQUEID}.WAV)
exten =
s,2,Dial(SIP/2${ARG1:-2}SIP/4${ARG1:-2}SIP/6${ARG1:-2},${OFFICE_TIMEOU
T},rw)
exten = s,3,Voicemail(u${ARG1})
exten = s,103,Voicemail(u${ARG1})
The call logs show:
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-1
,SIP/600-08e0b990,Dial,SIP/200SIP/400SIP/600|15|rw,2007-08-14
08:49:16,,2007-08-14 08:49:18,2,0,NO ANSWER,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-2
,SIP/600-08e19d58,Dial,SIP/200SIP/400SIP/600|15|rw,2007-08-14
08:49:46,,2007-08-14 08:49:56,10,0,NO ANSWER,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-1
,SIP/600-08e0b990,VoiceMail,u2000,2007-08-14
08:50:37,2007-08-14 08:50:52,2007-08-14
08:51:00,23,8,ANSWERED,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-2
,SIP/600-08e19d58,Dial,SIP/200SIP/400SIP/600|15|rw,2007-08-14
08:51:35,,2007-08-14 08:51:45,10,0,NO ANSWER,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-1
,SIP/600-08e0b990,VoiceMail,u2000,2007-08-14
08:52:19,2007-08-14 08:52:34,2007-08-14
08:52:38,19,4,ANSWERED,DOCUMENTATION
And my messages log for that time (for one failed call) shows:
ubiphone*CLI
-- Accepting AUTHENTICATED call from 193.111.200.135:
requested format = alaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw|alaw),
priority = mine
ubiphone*CLI
-- Executing Macro(IAX2/ubigradin-2, ext-group-home|2000) in
new
stack
-- Executing ExecIf(IAX2/ubigradin-2,
0|Monitor|wav|20070814-085135-07xx-2000-1187077895.3392.WAV)
in new stack
-- Executing Dial(IAX2/ubigradin-2,
SIP/200SIP/400SIP/600|15|rw) in new stack
ubiphone*CLI
-- Called 200
Aug 14 08:51:35 NOTICE[30952]: app_dial.c:1076 dial_exec_full: Unable
to
create channel of type 'SIP' (cause 3 - No route to destination)
-- Called 600
ubiphone*CLI
-- SIP/600-08e19d58 is ringing
ubiphone*CLI
-- SIP/200-08e0b990 is ringing
ubiphone*CLI
== Spawn extension (macro-ext-group-home, s, 2) exited non-zero on
'IAX2/ubigradin-2' in macro 'ext-group-home'
== Spawn extension (macro-ext-group-home, s, 2) exited non-zero on
'IAX2/ubigradin-2'
-- Hungup 'IAX2/ubigradin-2'
Adrian Marsh
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Exited non-zero here looks like the caller hungup before going to
voicemail, the caller is the one thing you can't control.
Anthony
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Re: [asterisk-users] DTMF on Bridged ZAP call
Check features.conf Jeremy Mann wrote: Should asterisk be intercepting DTMF on a bridged ZAP call? If so, how do I disable it recognizing #, as it’s hanging up my users when they try to enter #. ___ --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] Patent issues, what features we can't use?
Ah! you must be American! :-) Perhaps Zeeshan is looking for an understanding of the issues before seeking legal advice (it's a lot cheaper that way). Or perhaps it is a topic worthy of public discussion? I, for one, would be interested in any known issues. regards, Drew Eric Chamberlain wrote: Shouldn't you ask your attorney these questions? Any answers you receive here will not legally protect you. -- Eric Chamberlain , CISSP Chief Technical Officer Voxilla - http://voxilla.com/ * From: * [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Zeeshan Zakaria *Sent:* Monday, August 13, 2007 7:07 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Patent issues, what features we can't use? Hi everybody, As the Asterisk community is getting larger and larger, I was wondering that the features which are provided in Asterisk and are programmed by the open source community under GPL, or GUIs like FreePBX which also come loaded with wonderful features and uses same Asterisk, are they anywhere violating any patent laws? Most of the features work the same way as Nortel, Avaya and other PBX systems. Is there anyone who owns these features and will come one day to claim his royalties? When I deploy an asterisk soultion for a customer, is there any violation of any patent or copyright laws anywhere? Of if I use my own Asterisk server to provide services to some customers, am I violating any patent laws by not paying the royalties to some patent owners? I heard people saying that IVR technology is patented and google search for patents also say so. But we all are using IVR for ourselves and our customers without paying royalties to anyone. But when it comes to using g729, all of a sudden royalty issue comes in. So what is right to use and what is not? -- Zeeshan A Zakaria ___ --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 -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.com ___ --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] BLF with Aastra
Does 'sip show subscriptions' indicate that the 57i is still subscribed to the extension for updates? If not, you might have to do a test with 'sip Yes it does: EMSPBX*CLI sip show subscriptions Peer UserCall ID ExtensionLast state Type 10.30.17.120 120 72cc9be69ae 200 Idle dialog-info+xml 10.30.17.120 120 8bbbe608bf0 111 Idle dialog-info+xml 10.30.17.120 120 00399b8111a 110 Idle dialog-info+xml 10.30.17.120 120 34558bd778f 109 Idle dialog-info+xml 10.30.17.120 120 9fe8d2cc534 108 InUse dialog-info+xml 10.30.17.120 120 fc88630f51d 107 InUse dialog-info+xml 10.30.17.120 120 b867631abbc 106 Idle dialog-info+xml 10.30.17.120 120 4da7ecc52bf 105 Idle dialog-info+xml 10.30.17.120 120 99814a95b5c 104 Idle dialog-info+xml 10.30.17.120 120 52d5093eb40 103 Idle dialog-info+xml 10.30.17.120 120 ee47cbb7a6e 102 Idle dialog-info+xml 10.30.17.120 120 d7a3b47df8f 101 Idle dialog-info+xml debug peer aastraname' to confirm that the subscription is being made properly on phone startup and not being removed by the phone in response to some state change. A quick glance at chan_sip.c indicates that if a user agent tries to subscribe with an expiry time greater than 'maxexpiry' from sip.conf (default 3600 seconds), the subscription expiry in Asterisk will be silently changed to whatever the allowed maximum is. So if the Aastra is trying to subscribe for say 3 hours and Asterisk doesn't allow subscriptions greater than one hour, then notify messages will stop being sent after one hour until the Aatra re-subscribes. I haven't delved in very deep, so I can't tell if the response to the UA indicates the actual expiry Asterisk used, but even so you'd have to be certain that the Aastra respects an expiry in the response that differs from what it asked for. When you're doing the debug (hopefully on a quiet system), watch the phone boot, then use 'sip show subscriptions' to get the call-id of the subscription. Then watch for console messages indicating that the call has been destroyed (which should come at the 1 hour mark or whatever time the Aastra used for it's subscription length. -- 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] Patent issues, what features we can't use?
Eric Chamberlain wrote: Shouldn't you ask your attorney these questions? Any answers you receive here will not legally protect you. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ Nor will any answers from an attorney. Ever try and get a straight answer from one? Especially a patent and trademark one? I have, and the best to hope for is maybe Also keep in mind that, at least in the US, ANY asshole can sue anyone for the filing fee! Too many, such as Verizon, gain patents on things that really shouldn't be patented, thanks to the US PTO and their ignorance of technical matters, then others have to attempt to defend themselves in front of a jury of their peers who are as least as ignorant, and suffer the results. Refer back to the Vonage case. There is no guarantee of no risk. Your mother lied to you when she said everything would be alright John Novack -- Dog is my co-pilot ___ --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] Recognize 800 number
Is there a way to recognize if someone called our PRI using an 800 number? The DID is showing my 4 digit primary line, not anything obvious signifying that an 800 number is called? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Patent issues, what features we can't use?
John Novack [EMAIL PROTECTED] writes: There is no guarantee of no risk. Your mother lied to you when she said everything would be alright Maybe we can convince Digium to have an indemnification program for people who purchase the business edition! :) -- Kyle Sexton ___ --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] Using CURL
Is your dynamic page returning a newline after, like SIP/12345-1\n?
Moj
Mike wrote:
Hi,
Here is my first step (call it a proof of concept) in using the hint
priority with dynamic values.
Background - this works
exten = 12345,hint,SIP/12345-1
To make this a little dynamic, I used a web page to return to me the
value of the sip registration. In other words,
http://www.somepage.com/test.html returns the following (without
quotes): SIP/12345-1
I should therefore be getting the same result by using the following:
exten = 12345,hint,${CURL(http://www.somepage.com/test.html)}
BUTno. I get the following in the Asterisk CLI when reloading the
config:
Aug 8 18:24:27 NOTICE[26765]: pbx.c:1508
pbx_substitute_variables_helper_full: Error in extension logic (missing '}')
Now, I'm paying close attention to my Asterisk code up there, and I
don't see a missing '}' .
Anybody has an explanation for me? Is there some deeper meaning to this
notice I am getting?
Mike
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Re: [asterisk-users] Recognize 800 number
If your 800 number is setup in the same way I understand them to be, the 800 numbers are just forwards to a did (or your main number in the instance). You'll need to get a specific did setup just for your 800 number to use then you can just recognize the specific DID. Best Regards, William J McCloskey Information Technology Manager 503-827-8141 www.timbercon.com http://www.timbercon.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Tuesday, August 14, 2007 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Recognize 800 number Is there a way to recognize if someone called our PRI using an 800 number? The DID is showing my 4 digit primary line, not anything obvious signifying that an 800 number is called? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner http://www.mailscanner.info/ , and is believed to be clean. ___ --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] asterisk 1.2.24 installation
is there a new way to install asterisk? im using centos 4.5 and trying to install asterisk. when i do make clean and make install i get this error. # make clean --snip-- make[1]: Leaving directory `/usr/src/asterisk-1.2.24/apps' make: *** codecs: No such file or directory. Stop. make: *** [clean] Error 1 --snip-- # make --snip-- make[1]: Leaving directory `/usr/src/asterisk-1.2.24/apps' make: *** codecs: No such file or directory. Stop. make: *** [depend] Error 1 --snip-- ___ --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] Recognize 800 number
Are these POTS lines or a PRI? If you could get RDNIS from the carrier, then you could tell. My LD T1 only handles toll free numbers. It is called dedicated as opposed to switched. I may get some local DIDs from a VoIP provider just because some providers will reject calls with a toll free ANI. Thanks, Steve William McCloskey wrote: If your 800 number is setup in the same way I understand them to be, the 800 numbers are just forwards to a did (or your main number in the instance). You’ll need to get a specific did setup just for your 800 number to use then you can just recognize the specific DID. Best Regards, William J McCloskey Information Technology Manager 503-827-8141 www.timbercon.com http://www.timbercon.com/ *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Jeremy Mann *Sent:* Tuesday, August 14, 2007 9:19 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Recognize 800 number Is there a way to recognize if someone called our PRI using an 800 number? The DID is showing my 4 digit primary line, not anything obvious signifying that an 800 number is called? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by *MailScanner* http://www.mailscanner.info/, and is believed to be clean. ___ --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] Recognize 800 number
Jeremy Mann [EMAIL PROTECTED] writes: Is there a way to recognize if someone called our PRI using an 800 number? The DID is showing my 4 digit primary line, not anything obvious signifying that an 800 number is called? Can you just point the 800 number to an unused DID and track the calls by anything coming to that DID? I don't think 800 numbers actually pass that they are 800 #s. -- Kyle Sexton ___ --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] Recognize 800 number
Kyle Sexton wrote: Jeremy Mann [EMAIL PROTECTED] writes: Is there a way to recognize if someone called our PRI using an 800 number? The DID is showing my 4 digit primary line, not anything obvious signifying that an 800 number is called? Can you just point the 800 number to an unused DID and track the calls by anything coming to that DID? I don't think 800 numbers actually pass that they are 800 #s. On dedicated LD circuits, the DNIS is the toll free number. On switched, the toll free is pointed to a DID. Thanks, 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
[asterisk-users] Dial plan suggestions
Hello all, I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). Here's the gotcha. She doesnt like having to transfer calls to another phone; she'd rather just pick up the phone and have the call be active there as well (like good 'ol land lines). What I was thinking on how to do this is using some sort of call parking for the hunt group of all the phones in the house. Once the call is picked up, it then places both the SIP phone and caller into a meetme conference room. To simply join that static assigned room, one of the other phones picks up and joins that room. What I have a concern about is if we hang up, the caller can still sit there and listen in. When no phones are active, it should disconnect the caller. Does this implementation make sense? Has anyone else done something like this? Thanks. ___ --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] Patent issues, what features we can't use?
This legal question pops up every now and then, and depending on how paranoid you are you can eventually start thinking that the US patent office is under your bed.(I'm just checking now) First thing to note is that you aren't worth suing. This is a game that only applies to very big companies where you want to screw your competiton and have more lawyers than they do. Second thing to note is that this applies to US based companies only, (the country that still doesn't have universal health care). World wide, anything else falls into the UN or the EU. None of these organizations are dumb enough to get involved in anything as petty as this. Simply put, no-one outside corporate (telecom) America gives a monkey's balls who first coined the term Call Forwarding or ADSL There are more important things in life -- Henry L. Coleman. Kyle Sexton John Novack [EMAIL PROTECTED] writes: There is no guarantee of no risk. Your mother lied to you when she said everything would be alright Maybe we can convince Digium to have an indemnification program for people who purchase the business edition! :) -- Kyle Sexton ___ --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] Dial plan suggestions
Here's some details for you all. Asterisk 1.2 Polycom 301/601 phones As for my existing dial plan, I'm considering starting from scratch. Thanks again. Gerald A wrote: Hiya, On 8/14/07, *Russell Handorf* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). Wives are the most difficult customers. They are demanding, and you can't ever get away with not making them happy. :) What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). Here's the gotcha. She doesnt like having to transfer calls to another phone; she'd rather just pick up the phone and have the call be active there as well (like good 'ol land lines). You leave out what kind of phones you are using. It might be as easy as using a line appearance for a parking lot - or not - depending on what kind of phone you are using. SIP, ZAP and IAX phones, and even some within that may or may not support it. I have some GXP2000's, and I think I would do it that way. HTH ( a bit), Gerald. ___ --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] Recognize 800 number
On Aug 14, 2007, at 10:19 AM, Jeremy Mann wrote: Is there a way to recognize if someone called our PRI using an 800 number? The DID is showing my 4 digit primary line, not anything obvious signifying that an 800 number is called? some carriers wont' forward 10 digits DID by default, some do 3, some do 4, some do 7, so you want to call your PRI provider and ask them to pass 10 digit DID This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 Andres Paglayan --Harmony is more important than being right Bapak ___ --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] Recognize 800 number
On Tue, Aug 14, 2007 at 12:00:25PM -0500, Kyle Sexton wrote: Can you just point the 800 number to an unused DID and track the calls by anything coming to that DID? I don't think 800 numbers actually pass that they are 800 #s. It has traditionally been the case that non-trunk INWATS rang down on standard phone lines, and there was iindeed no way to tell the calls were dialled to the INWATS number. On trunked service (usually from an IXC, but sometimes from a LEC or CLEC), you could tell either by reserving timeslots/trunks for the INWATS traffic, or -- if you had realtime ANI or DNIS (usually on a PRI, these days) -- you could find out which specific number was being called. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --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] Dial plan suggestions
On Tue, 14 Aug 2007, Russell Handorf wrote: Hello all, I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). Here's the gotcha. She doesnt like having to transfer calls to another phone; she'd rather just pick up the phone and have the call be active there as well (like good 'ol land lines). Give her a DECT phone so she can carry it about with her. Worked for my wife! Gordon ___ --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] Asterisk 1.2 TDM24xx and B410P
Florent Barbier wrote: Hi here, Did you get any solution ? I've quiet the same pb : http://forums.digium.com/viewtopic.php?t=17394 Thank you for your answer. flo_turc Sorry for the late reply :-( We are aware of that particular issue, and working on tracking it down. Very big sorry for the inconvenience in the mean time :-( -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --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] Macro Overlap
yeah, 'enough' adds back the gray area that the black-and-white 'atomic' obscures... :P Moj Philipp Kempgen wrote: Mojo with Horan Company, LLC wrote: set your own mutex using astdb? It may just be atomic enough for you to get by. atomic enough - that's a nice term :-) Regards, Philipp Kempgen ___ --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] Dial plan suggestions
Gordon Henderson wrote: On Tue, 14 Aug 2007, Russell Handorf wrote: Hello all, I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). Here's the gotcha. She doesnt like having to transfer calls to another phone; she'd rather just pick up the phone and have the call be active there as well (like good 'ol land lines). Give her a DECT phone so she can carry it about with her. Worked for my wife! Gordon Parking is pretty easy once you try it, then no running to the other phone. Thanks, 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
Re: [asterisk-users] BLF with Aastra
Well that was it... it is no longer timing out. On 8/14/07, James FitzGibbon [EMAIL PROTECTED] wrote: On 8/14/07, Matt [EMAIL PROTECTED] wrote: I have a 536i expansion module attached to a 57i-CT. The BLF lights on the 536i will light up and work fine for a while... however after a bit they seem to loose their ability to see if someone is on a phone. They still work to dial, if I try to dial, however, they don't light up when someone makes a call, or if their phone rings. If I reboot the phone, the lights start working again (for a while). Does 'sip show subscriptions' indicate that the 57i is still subscribed to the extension for updates? If not, you might have to do a test with 'sip debug peer aastraname' to confirm that the subscription is being made properly on phone startup and not being removed by the phone in response to some state change. A quick glance at chan_sip.c indicates that if a user agent tries to subscribe with an expiry time greater than 'maxexpiry' from sip.conf (default 3600 seconds), the subscription expiry in Asterisk will be silently changed to whatever the allowed maximum is. So if the Aastra is trying to subscribe for say 3 hours and Asterisk doesn't allow subscriptions greater than one hour, then notify messages will stop being sent after one hour until the Aatra re-subscribes. I haven't delved in very deep, so I can't tell if the response to the UA indicates the actual expiry Asterisk used, but even so you'd have to be certain that the Aastra respects an expiry in the response that differs from what it asked for. When you're doing the debug (hopefully on a quiet system), watch the phone boot, then use 'sip show subscriptions' to get the call-id of the subscription. Then watch for console messages indicating that the call has been destroyed (which should come at the 1 hour mark or whatever time the Aastra used for it's subscription length. -- 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] Patent issues, what features we can't use?
After consulting with more experienced folk in the industry, some of which are running telecom companies for years, I came to the same conclusion what Henry has said. Now I feel much better and relaxed. On 8/14/07, Henry L.Coleman [EMAIL PROTECTED] wrote: This legal question pops up every now and then, and depending on how paranoid you are you can eventually start thinking that the US patent office is under your bed.(I'm just checking now) First thing to note is that you aren't worth suing. This is a game that only applies to very big companies where you want to screw your competiton and have more lawyers than they do. Second thing to note is that this applies to US based companies only, (the country that still doesn't have universal health care). World wide, anything else falls into the UN or the EU. None of these organizations are dumb enough to get involved in anything as petty as this. Simply put, no-one outside corporate (telecom) America gives a monkey's balls who first coined the term Call Forwarding or ADSL There are more important things in life -- Henry L. Coleman. ___ --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] Faulty voicemail
I would suggest writing these events into a db in realtime then you can
search through them by the caller id number and piece back together the
call using the unique id. Then you can know exactly what is happening.
Anthony
Adrian Marsh wrote:
Hmm... He swears he heard a voice saying he'd dialed the number
incorrectly.. But that's no-where in the dialplan, and I do see the
incoming calls correctly for the times he's saying..
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: 14 August 2007 15:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Faulty voicemail
Adrian Marsh wrote:
Hi All,
I was made aware today that some of my calls coming in are not going
to
voicemail... Below are some logs, and the macro that should run on
the
incoming_pstn context for that extension. I can see that theres a
non-zero exit before it gets to voicemail, but I've no idea why. In
this case theres 2 SIP clients to sim-call. On other occasions it
works
fine. In the CDR logs, I can see NO ANSWER and ANSWERED - what
would be there if voicemail answers?
Asterisk: 1.2.23
[macro-ext-group-home]
; ${ARG1} - Virtual Extension (e.g. 2005)
exten =
s,1,ExecIF($[${RECORDSIP}=TRUE],Monitor,wav|${TIMESTAMP}-${CALLERID(
num)}-${MACRO_EXTEN}-${UNIQUEID}.WAV)
exten =
s,2,Dial(SIP/2${ARG1:-2}SIP/4${ARG1:-2}SIP/6${ARG1:-2},${OFFICE_TIMEOU
T},rw)
exten = s,3,Voicemail(u${ARG1})
exten = s,103,Voicemail(u${ARG1})
The call logs show:
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-1
,SIP/600-08e0b990,Dial,SIP/200SIP/400SIP/600|15|rw,2007-08-14
08:49:16,,2007-08-14 08:49:18,2,0,NO ANSWER,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-2
,SIP/600-08e19d58,Dial,SIP/200SIP/400SIP/600|15|rw,2007-08-14
08:49:46,,2007-08-14 08:49:56,10,0,NO ANSWER,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-1
,SIP/600-08e0b990,VoiceMail,u2000,2007-08-14
08:50:37,2007-08-14 08:50:52,2007-08-14
08:51:00,23,8,ANSWERED,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-2
,SIP/600-08e19d58,Dial,SIP/200SIP/400SIP/600|15|rw,2007-08-14
08:51:35,,2007-08-14 08:51:45,10,0,NO ANSWER,DOCUMENTATION
,07x,2000,incomming_pstn,07x,IAX2/ubigradin-1
,SIP/600-08e0b990,VoiceMail,u2000,2007-08-14
08:52:19,2007-08-14 08:52:34,2007-08-14
08:52:38,19,4,ANSWERED,DOCUMENTATION
And my messages log for that time (for one failed call) shows:
ubiphone*CLI
-- Accepting AUTHENTICATED call from 193.111.200.135:
requested format = alaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw|alaw),
priority = mine
ubiphone*CLI
-- Executing Macro(IAX2/ubigradin-2, ext-group-home|2000) in
new
stack
-- Executing ExecIf(IAX2/ubigradin-2,
0|Monitor|wav|20070814-085135-07xx-2000-1187077895.3392.WAV)
in new stack
-- Executing Dial(IAX2/ubigradin-2,
SIP/200SIP/400SIP/600|15|rw) in new stack
ubiphone*CLI
-- Called 200
Aug 14 08:51:35 NOTICE[30952]: app_dial.c:1076 dial_exec_full: Unable
to
create channel of type 'SIP' (cause 3 - No route to destination)
-- Called 600
ubiphone*CLI
-- SIP/600-08e19d58 is ringing
ubiphone*CLI
-- SIP/200-08e0b990 is ringing
ubiphone*CLI
== Spawn extension (macro-ext-group-home, s, 2) exited non-zero on
'IAX2/ubigradin-2' in macro 'ext-group-home'
== Spawn extension (macro-ext-group-home, s, 2) exited non-zero on
'IAX2/ubigradin-2'
-- Hungup 'IAX2/ubigradin-2'
Adrian Marsh
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Exited non-zero here looks like the caller hungup before going to
voicemail, the caller is the one thing you can't control.
Anthony
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Re: [asterisk-users] asterisk 1.2.24 installation
looks broken, is there an apps dir in the source directory? Mark Quitoriano wrote: is there a new way to install asterisk? im using centos 4.5 and trying to install asterisk. when i do make clean and make install i get this error. # make clean --snip-- make[1]: Leaving directory `/usr/src/asterisk- 1.2.24/apps' make: *** codecs: No such file or directory. Stop. make: *** [clean] Error 1 --snip-- # make --snip-- make[1]: Leaving directory `/usr/src/asterisk-1.2.24/apps' make: *** codecs: No such file or directory. Stop. make: *** [depend] Error 1 --snip-- ___ --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] Dial plan suggestions
You want a key system, the fianl frontier of an asterisk implementation, and currently my holy grail. The best way to do it in an ugly way is to park the call and have a speed dial for pickup. Some phones like Aastra 55i and 57i can even have their hold button reprogrammed to blind transfer to the call parking. Russell Handorf wrote: Hello all, I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). Here's the gotcha. She doesnt like having to transfer calls to another phone; she'd rather just pick up the phone and have the call be active there as well (like good 'ol land lines). What I was thinking on how to do this is using some sort of call parking for the hunt group of all the phones in the house. Once the call is picked up, it then places both the SIP phone and caller into a meetme conference room. To simply join that static assigned room, one of the other phones picks up and joins that room. What I have a concern about is if we hang up, the caller can still sit there and listen in. When no phones are active, it should disconnect the caller. Does this implementation make sense? Has anyone else done something like this? Thanks. ___ --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] FXO Modules and Sip Outbound
Erik,
In the sip.conf file, would I put my Asterisk Box's ip address in the
host field? What would I do with the registration field? Leave it alone?
Thanks in advance.
Best Regards,
John
From: Erik Anderson [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] FXO Modules and Sip Outbound
Date: Mon, 13 Aug 2007 16:36:08 -0500
On 8/13/07, John Meksavan [EMAIL PROTECTED] wrote:
Asterisk Users,
I have never done a dial plan for this scenario before. Is it
possible to
have Sip Phones make outbound calls through the PSTN? What would the
call
routing/dial plan would look like?
Yes - certainly possible. There's nothing different about the call
routing going from SIP-Zap as from SIP-SIP really. Assuming that
you already have your zaptel device(s) configured correctly, something
like this in your dialplan is all you'll need. This also assumes you
want to dial 9 to get an outside line.
[globals]
OUTBOUND-TRUNK=Zap/g0
[outbound]
exten = _9NXXNXX,1,Dial(${OUTBOUND-TRUNK}/${EXTEN:1})
-Erik
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_
See what youre getting into
before you go there
http://newlivehotmail.com/?ocid=TXT_TAGHM_migration_HM_viral_preview_0507
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Re: [asterisk-users] Converting an audio file to a .gsm format
Digium has a handy tool online! http://www.digium.com/en/products/voice/audioconverter.php :-) -- Chris Alex Balashov [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On Sun, 12 Aug 2007, MOSBAH ABDELKADER wrote: Hello all, have anyone an idea about converting an audio file (.wav, .mp3, .au,...) to a .gsm audio file to use it as a voicemail file with Asterisk. 'sox' should be able to do this, AFAIK. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Pickup command
Not really sure about SIP exactly, but for Asterisk 1.0 versions, I know that the Pickup only works with Zaptel channels -- so to use it for any sort of IP channel, IAX for example, you have to use an addon/patch google it, 'pickup2' I think it's called works well, allows the Pickup command to grab any ZAP or IAX channel -- Chris Earle Carlos Chavez [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] ___ --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] asterisk 1.2.24 installation
On Tue, 14 Aug 2007, Anthony Francis wrote: looks broken, is there an apps dir in the source directory? Built OK for me: unicorn*CLI show version Asterisk 1.2.24 built by root @ unicorn on a i686 running Linux on 2007-08-11 08:22:22 UTC I didn't do anything special... Gordon ___ --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] BLF with Aastra
What did you change? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: 14 August 2007 20:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] BLF with Aastra Well that was it... it is no longer timing out. On 8/14/07, James FitzGibbon [EMAIL PROTECTED] wrote: On 8/14/07, Matt [EMAIL PROTECTED] wrote: I have a 536i expansion module attached to a 57i-CT. The BLF lights on the 536i will light up and work fine for a while... however after a bit they seem to loose their ability to see if someone is on a phone. They still work to dial, if I try to dial, however, they don't light up when someone makes a call, or if their phone rings. If I reboot the phone, the lights start working again (for a while). Does 'sip show subscriptions' indicate that the 57i is still subscribed to the extension for updates? If not, you might have to do a test with 'sip debug peer aastraname' to confirm that the subscription is being made properly on phone startup and not being removed by the phone in response to some state change. A quick glance at chan_sip.c indicates that if a user agent tries to subscribe with an expiry time greater than 'maxexpiry' from sip.conf (default 3600 seconds), the subscription expiry in Asterisk will be silently changed to whatever the allowed maximum is. So if the Aastra is trying to subscribe for say 3 hours and Asterisk doesn't allow subscriptions greater than one hour, then notify messages will stop being sent after one hour until the Aatra re-subscribes. I haven't delved in very deep, so I can't tell if the response to the UA indicates the actual expiry Asterisk used, but even so you'd have to be certain that the Aastra respects an expiry in the response that differs from what it asked for. When you're doing the debug (hopefully on a quiet system), watch the phone boot, then use 'sip show subscriptions' to get the call-id of the subscription. Then watch for console messages indicating that the call has been destroyed (which should come at the 1 hour mark or whatever time the Aastra used for it's subscription length. -- 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 ___ --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] Dial plan suggestions
Anthony Francis wrote: You want a key system, the fianl frontier of an asterisk implementation, and currently my holy grail. The best way to do it in an ugly way is to park the call and have a speed dial for pickup. Some phones like Aastra 55i and 57i can even have their hold button reprogrammed to blind transfer to the call parking. Isn't this what Shared Line Appearance is supposed to do? (Supported in 1.4...) -Stephen- ___ --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] misdn and incoming fax detection
hmmm.. no ideas?! :-| tom Thomas Artner wrote: Hi! At the moment i am using a digium tdm400 card for my analog phone lines. The zaptel driver supports fax detection, so incoming faxes are redirected to the fax extension automatically. This works without problems with asterisk 1.2. But now I would like to switch to ISDN (mISDN) and asterisk 1.4. I have a ISDN Card which supports the mISDN channel. Everything is working fine, but I dont know how I can do such incoming fax detection as I did it with the tdm400 card. :-( Is the only way using nvFaxDetect? Its terrible to get it run with asterisk 1.4. As well http://www.newmantelecom.com/asterisk/faxdetect/ seems not to be available anymore. Has anyone some ideas how I can do such incoming fax detection with mISDN and asterisk 1.4 ? thx, Tom ___ --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] Dial plan suggestions
Since I dont use 1.4 then you tell me. :) Stephen Bosch wrote: Anthony Francis wrote: You want a key system, the fianl frontier of an asterisk implementation, and currently my holy grail. The best way to do it in an ugly way is to park the call and have a speed dial for pickup. Some phones like Aastra 55i and 57i can even have their hold button reprogrammed to blind transfer to the call parking. Isn't this what Shared Line Appearance is supposed to do? (Supported in 1.4...) -Stephen- ___ --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] Dial plan suggestions
Anthony Francis wrote: Since I dont use 1.4 then you tell me. :) This functionality is supposed to be supported in 1.4, though I've never personally tested it. When it's configured it gives the key system behaviour you describe. -Stephen- ___ --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] Pickup command
Chris Earle wrote: Not really sure about SIP exactly, but for Asterisk 1.0 versions, I know that the Pickup only works with Zaptel channels -- so to use it for any sort of IP channel, IAX for example, you have to use an addon/patch google it, 'pickup2' I think it's called works well, allows the Pickup command to grab any ZAP or IAX channel Have you used this yourself? I need something like this. (This limitation is frustrating.) -Stephen- ___ --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] Some advice
I need a quick bit of advice from the list. We purchased an asterisk based phone system back about 6 months ago and we are using Cisco 7940G phones (I know, not everyone's favorites). We are using the second line on the phones for paging with a auto-answer, now my question is having the system call 20 of these paging extensions, should that be enough load to cause instability in the system? Our vendor is claiming it is causing the problems we are having, and I really find that hard to believe. Thoughts? Should that be enough to cause major stability problems? It's an Athlon 3800+ with 512mb ram and a Sangoma card with one PRI. Total of about 60 extensions (40 phones) on the system but only about 2-4 active calls at any given time with very little transcoding or other such intensive processes going on. Thanks, William ___ --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] Some advice
William McCloskey wrote: I need a quick bit of advice from the list. We purchased an asterisk based phone system back about 6 months ago and we are using Cisco 7940G phones (I know, not everyone's favorites). We are using the second line on the phones for paging with a auto-answer, now my question is having the system call 20 of these paging extensions, should that be enough load to cause instability in the system? Our vendor is claiming it is causing the problems we are having, and I really find that hard to believe. Can you be more specific about the stability problems? That's a bit vague -- it makes it hard to understand what's really happening. -Stephen- ___ --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] Dial plan suggestions
Stephen Bosch wrote: Anthony Francis wrote: Since I dont use 1.4 then you tell me. :) This functionality is supposed to be supported in 1.4, though I've never personally tested it. When it's configured it gives the key system behaviour you describe. -Stephen- ___ --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 Looks like you are right, man I cannot wait for them to fix the CDR problems in 1.4 so that I can move to it. http://www.voip-info.org/wiki/view/Asterisk+SLA Anthony ___ --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] Patent issues, what features we can't use?
On 8/14/07, Henry L.Coleman [EMAIL PROTECTED] wrote: (the country that still doesn't have universal health care). Sorry for hijacking this thread but I just couldn't resist. This is about the only thing in your email I have to disagree with. I am thankful that we (meaning citizens of the USA) don't have a universal health care like in Canada or UK. My mothers family coming and still living in Canada, and my in laws in the UK, I have seen the universal health care systems what they are, and I hope the US never adopts one. ___ --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] 20min waiting time
On 15:02, Sun 12 Aug 07, OCOSA ListAcct wrote: exten=5,2,Dial(SIP/supportSIP/support2,2,tr) Make this line read: exten=5,2,Dial(SIP/supportSIP/support2,,tr) That should do the trick -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] Patent issues, what features we can't use?
On Tue, Aug 14, 2007 at 06:53:26PM -0400, C F wrote: Sorry for hijacking this thread but I just couldn't resist. This is about the only thing in your email I have to disagree with. Sorry to hijack your thread, but I'll just note that yoou have just stepped out of this list's topic. So let's agree to disagree. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Some advice
The stability problems we have seem to be related to asterisk crashing the apache install on the box when the PHP scripts are performing functions via asterisk. Don't know exactly how they work it all, but that's the gist of it. Best Regards, William J McCloskey Information Technology Manager 503-827-8141 www.timbercon.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Tuesday, August 14, 2007 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Some advice William McCloskey wrote: I need a quick bit of advice from the list. We purchased an asterisk based phone system back about 6 months ago and we are using Cisco 7940G phones (I know, not everyone's favorites). We are using the second line on the phones for paging with a auto-answer, now my question is having the system call 20 of these paging extensions, should that be enough load to cause instability in the system? Our vendor is claiming it is causing the problems we are having, and I really find that hard to believe. Can you be more specific about the stability problems? That's a bit vague -- it makes it hard to understand what's really happening. -Stephen- ___ --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] Patent issues, what features we can't use?
We have differing views. I am a Canadian and was born in the UK I would not be alive today had it not been for the National Health Service. I don't see any merit in a system that has over 35 million people that have no health care and a government that could afford health care for every man, women and child for less than the cost of the war in Iraq. I am sorry if I offend you, but I'm just calling it like I see it. You have the right to say mind your own business. I just hope you are not one of the 35 million. -- Henry L. Coleman. C F On 8/14/07, Henry L.Coleman [EMAIL PROTECTED] wrote: (the country that still doesn't have universal health care). Sorry for hijacking this thread but I just couldn't resist. This is about the only thing in your email I have to disagree with. I am thankful that we (meaning citizens of the USA) don't have a universal health care like in Canada or UK. My mothers family coming and still living in Canada, and my in laws in the UK, I have seen the universal health care systems what they are, and I hope the US never adopts one. ___ --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] Patent issues, what features we can't use?
No problem, lets move on.. -- Henry L. Coleman. Tzafrir Cohen On Tue, Aug 14, 2007 at 06:53:26PM -0400, C F wrote: Sorry for hijacking this thread but I just couldn't resist. This is about the only thing in your email I have to disagree with. Sorry to hijack your thread, but I'll just note that yoou have just stepped out of this list's topic. So let's agree to disagree. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Patent issues, what features we can't use?
I must agree and I apologize, I agree with Tzafrir. On 8/14/07, Henry L.Coleman [EMAIL PROTECTED] wrote: No problem, lets move on.. -- Henry L. Coleman. Tzafrir Cohen On Tue, Aug 14, 2007 at 06:53:26PM -0400, C F wrote: Sorry for hijacking this thread but I just couldn't resist. This is about the only thing in your email I have to disagree with. Sorry to hijack your thread, but I'll just note that yoou have just stepped out of this list's topic. So let's agree to disagree. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 ___ --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] Pickup command
On 8/10/07, Carlos Chavez [EMAIL PROTECTED] wrote:
I am having a bit of a problem implementing the pickup command in
my
dial plan. I have setup this rule:
exten = _*8XXX,1,Pickup(${EXTEN:2})
This works as expected when someone dials an extensions number and
I
can get the call. The problem I have is that when a call enters my
welcome menu and does not press anything there is a timeout that sends
them to the recepcionist. The rule is:
exten = t,1,Macro(stdexten,${OPERVM},${OPER})
If I understand the command correctly, the pickup you would need to initiate
for this would be picking up the 't' extension. Instead of writing 't' like
that, I would use a go to:
exten = t,1,Goto([some context],[some extension],1)
Then, your pickup would be picking up [some extension].
Ex.:
exten = t,1,Goto(internal,100,1)
your pickup command would be:
exten = _*8XXX,1,Pickup(${EXTEN:2})
and then you would enter *8100 to pickup the ringing caller.
See http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup, and more
specifically the REMARK heading.
${OPERVM} and ${OPER} are set to the mailbox and sip device of the
receptionist. How can I directly pick up that call? I stopped using
the default *8 because the client kept picking up calls that they did
not intend to so I need to make it specific.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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--
Lacy Moore
Somewhere I wish I wasn't
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Re: [asterisk-users] Playback a video file?
As far as I know, yes. Someone even published a how-to on making up video IVR's. PaulH On Sun, 2007-08-12 at 08:54 -0400, SIP wrote: Is it possible to record or playback a video file in Asterisk? N. ___ --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] Need Help in changing Voice message
You could look at processing the dialstatus (with a goto(s-dialstatus)) as used in macro-voicemail..We did that for a client that got different beeps, not messages. PaulH On Fri, 2007-08-10 at 14:56 +1000, Farooq Ahmed wrote: Thank you very much who answered to the questions. You have realy saved in wondering around the darkness. Yes it was related the phone not with the asterisk and i was looking in the asterisk yesterday. Because when i used xlite softphone i got the message which have stated in my mail and when i used other softphone i got differnt reply only warning beep . Regards Farooq ___ --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] Some advice
so you are not talking about vanilla asterisk, there are some other applications involved. Paging by nature is resource intensive, but still not sure what else is going on in your system. On 8/14/07, William McCloskey [EMAIL PROTECTED] wrote: The stability problems we have seem to be related to asterisk crashing the apache install on the box when the PHP scripts are performing functions via asterisk. Don't know exactly how they work it all, but that's the gist of it. Best Regards, William J McCloskey Information Technology Manager 503-827-8141 www.timbercon.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Tuesday, August 14, 2007 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Some advice William McCloskey wrote: I need a quick bit of advice from the list. We purchased an asterisk based phone system back about 6 months ago and we are using Cisco 7940G phones (I know, not everyone's favorites). We are using the second line on the phones for paging with a auto-answer, now my question is having the system call 20 of these paging extensions, should that be enough load to cause instability in the system? Our vendor is claiming it is causing the problems we are having, and I really find that hard to believe. Can you be more specific about the stability problems? That's a bit vague -- it makes it hard to understand what's really happening. -Stephen- ___ --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 ___ --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] LumenVox Speech Recognition
randulo wrote: Nitesh, I've messed with the Lumenvox starter kit. If you are serious about this field, I think it's a must see. It was easy to set up and there are demos available. Their support is excellent. There is a quiet mailing list where questions are never ignored and most problems are solved AFAIK. I'll agree that the Lumevox starter kit is a great starting point. I started out with the dialplan version of their pizza demo and quickly realized that using a dialplan would degenerate into spaghetti code very quickly. So using the Asterisk-java library I ported their pizza demo to Java (my code is now one of the demos on that page). I found this combined with Cepstral to be a great platform that could be used to write data driven voice recognition applications. If you are doing something interesting, the Lumenvox folks are very helpful in getting you going. Steve Take a look here for demos, etc. http://lumenvox.com/partners/integrator/digium/applicationzone/i /r ___ --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
