I can't check the voicemail for the switchboard. Asterisk hangs up for some
unknown reason...
- s n i p -
-- Executing [EMAIL PROTECTED]:1] Wait(SIP/597-00f0c410, 1) in new stack
-- Executing [EMAIL PROTECTED]:2] VMAuthenticate(SIP/597-00f0c410,
[EMAIL PROTECTED]|s) in new stack
Quoting Vincent [EMAIL PROTECTED]:
exten = h,n,Set(WAV_FILE=${IF($[ ${STAT(e,/tmp/${CALLTIME}.wav)}
]?${CALLTIME}.wav)})
^ ^
^
To start, to many spaces... And a missing end parenthesis... And a end
parenthesis
Quoting Doug Lytle [EMAIL PROTECTED]:
Anybody else encountered this?
I have. But i did it manually, not from a cron job...
It didn't restart for me either... I had to resort
to a full restart with the init script...
--
PLO toluene SEAL Team 6 supercomputer president DES Waco, Texas
Cocaine NSA
We have now got our new PRI line (10 channels, 100 numbers) connected
and everything is working except the outgoing caller ID. Whatever
SIP phone I'm using, the CID that's shown is the very first number...
- s n i p -
[default]
include = outgoing
include = priin
[outgoing]
exten =
mail-lists == mail-lists [EMAIL PROTECTED] writes:
mail-lists I don't know if the same is true for you but we had to
mail-lists call our telco and have them set our callerid settings
mail-lists to 'station level'. Not sure if your telco offers this
mail-lists but they should.
Quoting Anciso, Roy [EMAIL PROTECTED]:
I do this to tie extensions to a particular number:
exten = _9X./_2XXX,1,SET(CALLERID(all)=Manistee ISD2317231516)
exten = _9X./_1XXX,1,SET(CALLERID(all)=MISD Tecnology2317234264)
Tried that but couldn't get it to work. I've tried all the CALLERID()
Quoting Joe Acquisto [EMAIL PROTECTED]:
My thanks to all. Problem resolved with the assistance.
Would be nice if you posted HOW it was fixed to... I have this exact
same problem at home, but the work phones displays time correctly...
joe a.
On 11/1/2007 at 1:43 PM, Joe Acquisto [EMAIL
Quoting mail-lists [EMAIL PROTECTED]:
Turbo Fredriksson wrote:
We have now got our new PRI line (10 channels, 100 numbers) connected
and everything is working except the outgoing caller ID. Whatever
SIP phone I'm using, the CID that's shown is the very first number...
Enabling PRI debugging
p -
This results in (doctored):
- s n i p -
-- Executing [MY_CELL_NO@default:1] Macro(IAX2/graham-1,
dial|MY_CELL_NO|30|r) in new stack
-- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/graham-1, Trying
extension/number: MY_CELL_NO from Turbo Fredriksson 528) in new stack
Quoting Eric \ManxPower\ Wieling [EMAIL PROTECTED]:
Remember Caller*ID Number is either country code + area code + number or
area code + number. You never put a 1 or 0 at the beginning of the
number. CallerID Number also can not have spaces, dashes, or other crud.
Darn! This last part
I'm trying to load ztdummy on my Asterisk, running in a XEN
domain.
I've modified the code to disable the use of an RTC.
I can load the zaptel module just fine, the ztdummy also
loads without problem. But when running ztcfg I get this
error.
- s n i p -
graham:~# ztcfg -
Zaptel
Tzafrir == Tzafrir Cohen [EMAIL PROTECTED] writes:
Tzafrir Take a look at /proc/zaptel/1
It's empty.
Tzafrir Any chance that this is ztdummy ?
It is.
Tzafrir You don't need a span line (or running ztcfg at all) for
Tzafrir ztdummy.
Ah! Doh. That isn't in any documentation
I've been trying to setup AddQueueMember() as a replacement
for the deprecated AgentCallbackLogin(), but I get _tree_
Queue()'s.
Massaged extensions.conf (can provide the original if need be):
- s n i p -
[default]
include = agent-loginout
include = local
; --
[agent-loginout]
Quoting WipeOut [EMAIL PROTECTED]:
Anyone had any experience with an Asterisk server as a VMWare virtual
machine?
I was trying to run it under XEN and got into trouble so in all my
searches, the conclusion was that running it under VMWare didn't
work because of the faulty timer in VMWare...
Quoting C F [EMAIL PROTECTED]:
Version? What is the CLI output? What phone are you using?
It appears that it is hanging up cleanly, and the reorder tone is from
the phone.
This is asterisk v1.4.13. The CLI say:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/2401-081e4440, Unconditional
Anything to do about portscans? Is there any way (should I) to see
if the connection is a legit (only SIP currently) connection BEFORE
my * answers?
[2007-10-17 19:23:46] WARNING[4191]: chan_sip.c:6624 determine_firstline_parts:
Bad request protocol 01@ASTERISK_IP SIP/2.0
-- Executing
Quoting Mojo with Horan Company, LLC [EMAIL PROTECTED]:
He's worried that the Hangup application returns non-zero.
And that the phone (Polycom SoundPoint IP430 SIP) 'indicates' an
error even though there wasn't.
It does this because of the forced Hangup() as I see/understand it.
Took some examples from voip-info.org to deal with
call forwarding etc:
exten = _*21*X.,1,NoOp(Unconditional Call Forward on extension
${CALLERID(num)} to ${EXTEN:4})
exten = _*21*X.,n,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
exten = _*21*X.,n,Hangup()
Problem is that * don't hangup cleanly:
I'm using Swedish on version 1.4.13. The full part of the
log is:
[Oct 13 12:51:16] WARNING[7810] file.c: File digits/ett does not exist in any
format
[Oct 13 12:51:16] WARNING[7810] file.c: Unable to open digits/ett (format 0x8
(alaw)): No such file or directory
The word 'ett' means 'one'.
Anselm == Anselm Martin Hoffmeister [EMAIL PROTECTED] writes:
Anselm You could also copy the file en.gsm which should exist
Anselm there over to ett.gsm - wrong reading will result, but I
Anselm guess people understand what is meant, like they would
Anselm understand you have
I can't seem to get the [s]tart to work in my extensions...
- s n i p -
[default]
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Voicemail(${EXTEN}, b)
exten = 2403,1,Dial(sip/${EXTEN},20,t)
exten = _X.,2,Playback(pbx-invalid)
- s n i p -
If I dial '2403' with is
Philipp == Philipp Kempgen [EMAIL PROTECTED] writes:
files come from? I.e. who recorded them/whos voice it is?
Philipp Only you can tell where you got the sound files you use.
I thought they came with Asterisk (v1.4.13).. Sorry, that was a separate
package Got the supplier, thanx.
Quoting Philipp Kempgen [EMAIL PROTECTED]:
Turbo Fredriksson wrote:
I can't seem to get the [s]tart to work in my extensions...
- s n i p -
[default]
exten = s,n,Goto(s-${DIALSTATUS},1)
The first priority in an extension must be 1 not n.
Actually, I did. I just had it commented
Quoting Philipp Kempgen [EMAIL PROTECTED]:
exten = s,1,Answer()
exten = s,n,Goto(s-${DIALSTATUS},1)
This still doesn't make sense because you did not Dial()
before jumping based on ${DIALSTATUS}.
Ok, make sense. But still no go:
- s n i p -
[default]
exten = s,1,Answer()
exten =
Sorry for this. This is most likely a HOWTO or FAQ question, but
it's so much information and documentation to wade through so
I hope someone could take a minute to answer anyway.
If not, no worries. I'll get to it sooner or later :)
I'm trying to understand what Asterisk actually is and the
zoachien == zoachien [EMAIL PROTECTED] writes:
zoachien Turbo Fredriksson wrote:
How do I connect to a 'normal' (i.e. analog) telephone?
zoachien - you can take a voip provider and not buy any hardware.
I was thinking in this way to, but I was unsure if I can still use
Asterisk
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