Sometimes I get these in my kernel log:
[3763662.549761] __sdla_bus_read_4:888: wanpipe PCI Error: Illegal Register
read: 0x0040 = 0x
[4014422.079673] __sdla_bus_read_4:888: wanpipe PCI Error: Illegal Register
read: 0x0040 = 0x
Anyone who knows what this is about?
2010/2/25
hello I would like to implement anonymous calls. is someone who can help
me with an idea in extensions.conf
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Hello !
My macro to avoid voicemail of a cellphone is not really working. Can
you take a look at it :
This is the macro :
[macro-testgsm]
exten = s,1,NoOp(inside macro testgsm)
exten = s,n,Wait(2)
exten = s,n,Read(INPUT,,1,1,1)
exten = s,n,GoToIf($[${INPUT}==1]?exit:hangup)
exten =
Hello
I have successfully compiled OSLEC for echo cancellation for DAHDI channel.
Is there any way to do echo cancellation for SIP Channel.
Is any, please suggest me.??
Thanks in advance..
--
Regards,
Chandrakant Solanki
--
On Wed, Mar 03, 2010 at 04:41:49PM -0600, Jason Marble wrote:
Is there any easy way to identify which script or service is
connecting to the Asterisk manager? Somewhere on my system a script or
service is trying to connect with a bad user name or password. I get
the following error: connect
Hi All,
I have an asterik machine which is connected via a PRI to the SIP server.When
i call from the Asterisk machine to the SIP server i am not getting the caller
id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf file is
attached.
Hi, This comes up with a load of results however I can't see anything
that relates directly to what I'm talking about. Is anyone doing this in
their setup?
Cheers,
Mark
On Thu, Mar 04, 2010 at 02:34:17PM +1300, Alec Davis wrote:
Search bugs.asterisk.org and enter 'digital' in the search field.
HI All,
Below is the ones i tried
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234003,1,Set(CALLERID(num)=${8001234003})
exten = 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
Hi,
You need to set the callerid before making the call, not after. Also, I guess it's a typo that the priority in this dialplan is all 1; it should be
exten = 8001234003,1,Set(CALLERID(num)=8001234003)exten = 8001234003,n,Set(CALLERID(name)="Line 5")
exten = 8001234003,n,Dial(DAHDI/34,,rt)
- Mark Adams m...@campbell-lange.net escreveu:
Hi, thanks for your response.
I'm not sure if I explained correctly. I need asterisk to provide an
ISDN data function, whilst also routing voice calls over the same PRI.
Is this possible?
Regards,
Mark
On 3 Mar 2010, at 17:58,
Hi All,
Please note that this is a lab setup and we are not connected to any external
telcos
Rgds
Venu
From: asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnaiyer
Venugopal-Q16770
Sent: Thu 3/4/2010 5:20 PM
To: Asterisk Users Mailing
Hi Jimmy,
Appreciate your help.
I tried the one below and cudnt get the caller ID.I am getting Private Call
and Out of Area in the sip phone display when i call from asterisk.
My current extensions.conf looks like below
[general]
static=yes
writeprotect=no
autofallthrough=no
Hello list.
ChanIsAvail returns 20 for ${AVAILSTATUS}. What does this '20' mean ??
...
exten = 1,n,ChanIsAvail(SIP/sin10)
exten = 1,n,NoOp(chanisavail == ${AVAILSTATUS})
...
[Mar 4 15:10:16] -- Executing [...@sin:7]
ChanIsAvail(IAX2/testlocal-14088, SIP/sin10) in new stack
[Mar 4
At the risk of being flamed
Has anyone had any success get the 'El cheapo' Wildcard W100P clone's
(£20 flavour) to work with UK Caller ID?
I'm not sure what the status of Asterisk 1.6 is with respect to UK
caller ID, being that we have an odd method of sending the FSK ahead of
the ring, but
*Hello,*
*
*
*Just thought to post our experiences trying to get a Polycom Soundpoint 450
working through a Sofaware to an endpoint doing SIP natting.*
*
*
*As mentioned above our situation was such. We use Asterisk as our PBX and
have SIP natted through the corporate firewalls. A remote user has
On Thu, 4 Mar 2010, Brian wrote:
At the risk of being flamed
Has anyone had any success get the 'El cheapo' Wildcard W100P clone's
(£20 flavour) to work with UK Caller ID?
Looks like there is a patch to Zaptel to make it work:
I had a customer ask me about time/date information being sent to his
analog (attached to a Linksys SPA2102) answering machine. I didn't know
that POTS could carry this information. Is this something Asterisk could
send over SIP?
Cheers,
j
--
when there;s incoming call from
pstn. result as shwon here
-- Executing [...@from-pstn:1] Set(DAHDI/1-1, CallTime=20100304
13:45:30) in new stack
-- Executing [...@from-pstn:2] Set(DAHDI/1-1, CallerIDString=
01935x) in new stack
-- Executing [...@from-pstn:3] System(DAHDI/1-1, /bin/echo
On Thu, 2010-03-04 at 15:35 +, Gordon Henderson wrote:
On Thu, 4 Mar 2010, Brian wrote:
At the risk of being flamed
Has anyone had any success get the 'El cheapo' Wildcard W100P clone's
(£20 flavour) to work with UK Caller ID?
Looks like there is a patch to Zaptel to make it
On Thu, 4 Mar 2010, Brian wrote:
(your email is a bit weird - my clients not quoting it right )-:
On Thu, 2010-03-04 at 15:35 +, Gordon Henderson wrote:
On Thu, 4 Mar 2010, Brian wrote:
At the risk of being flamed
Has anyone had any success get the 'El cheapo' Wildcard W100P
Jeff LaCoursiere wrote:
I had a customer ask me about time/date information being sent to his
analog (attached to a Linksys SPA2102) answering machine. I didn't know
that POTS could carry this information. Is this something Asterisk could
send over SIP?
Cheers,
j
Time and date
Brian wrote:
At the risk of being flamed
Has anyone had any success get the 'El cheapo' Wildcard W100P
clone's (£20 flavour) to work with UK Caller ID?
I'm not sure what the status of Asterisk 1.6 is with respect
to UK caller ID, being that we have an odd method of sending
the
Given the PSTN rings once a month if that - all the rest is SIP via
ITSP, I can't justify the cost of 5 times a telephone for a card - but
thanks for the heads up GK old chap.
It's not a big issue - but an irritation if anything. I suspect a £2.50
callerID unit, a serial IC and a
Thanks Again steve .
Actually I feel that is expensive for my initial requirement of maling Asterisk
and Zaptel work on my Linux box.
I saw this in ebay.
only 1 FXO.
Asterisk X100P(B2) FXO PCI For IP-PBX From U.S
On 4 Mar 2010, at 17:22, Aditya Kumar wrote:
I saw this in ebay.
only 1 FXO.
Asterisk X100P(B2) FXO PCI For IP-PBX From U.S
link is :
Hi IRFAN,
Thanks for that, actually, I think my FXO card already struck by
lightning. I;ve changed to another card, and now work like charm.
-
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New to
Hi,
Im one of the user for this card. It works like charm.
in my country i have to set the signalling to fxs_ls and it works.
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On Thu, 4 Mar 2010, Dave Fullerton wrote:
Jeff LaCoursiere wrote:
I had a customer ask me about time/date information being sent to his
analog (attached to a Linksys SPA2102) answering machine. I didn't know
that POTS could carry this information. Is this something Asterisk could
send
Jeff LaCoursiere wrote:
On Thu, 4 Mar 2010, Dave Fullerton wrote:
Jeff LaCoursiere wrote:
I had a customer ask me about time/date information being sent to his
analog (attached to a Linksys SPA2102) answering machine. I didn't know
that POTS could carry this information. Is this something
Hi Guys,
i am using the following config in pbx1:
register = pbx1:endop...@172.16.200.175 pbx1%3aendop...@172.16.200.175
[pbx2]
type=friend
host=dynamic
trunk=yes
sercret=password
context=[default]
deny=0.0.0.0/0.0.0.0
permit=172.16.200.175/255.255.255.128
in pbx2:
register =
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
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New to Asterisk? Join us for a live
Record a muted channel for 30 minutes like this:
exten = s,1,Answer(1)
exten = s,n,Progress()
exten = s,n,record(silence_long.gsm|1800|s)
exten = s,n,hangup
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC
Sent:
- David @ULC ucoms2...@gmail.com wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
Enable recording using monitor() or mixmonitor() in GSM format, call, then put
your handset on mute for 30 minutes. :-)
--Tim
--
On Fri, 5 Mar 2010, David @ULC wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
Record yourself thinking of the solution for 1/2 of an hour.
--
Thanks in advance,
Record a muted channel for 30 minutes like this:
exten = s,1,Answer(1)
exten = s,n,Progress()
exten = s,n,record(silence_long.gsm|1800|s)
exten = s,n,hangup
Above option looks easy.
What I have to dial from soft phone to get this ?
On Fri, Mar 5,
On Thu, Mar 4, 2010 at 1:00 PM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
My Linksys PAP2T-NA at home has it's own clock / NTP settings that sends the
timestamp out to my analog phones. Check through the settings tab on your
Linksys for a time setting.
--
Thanks,
--Warren
On Thu, 4 Mar 2010, Steve Edwards wrote:
On Fri, 5 Mar 2010, David @ULC wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
Record yourself thinking of the solution for 1/2 of an hour.
Use sox to concatenate 6.9 copies of
I believe we GSM of 8 bit for Asterisk ?
On Fri, Mar 5, 2010 at 4:35 AM, David @ULC ucoms2...@gmail.com wrote:
Record a muted channel for 30 minutes like this:
exten = s,1,Answer(1)
exten = s,n,Progress()
exten = s,n,record(silence_long.gsm|1800|s)
exten = s,n,hangup
On 4 Mar 2010, at 23:11, Steve Edwards wrote:
On Thu, 4 Mar 2010, Steve Edwards wrote:
On Fri, 5 Mar 2010, David @ULC wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
Record yourself thinking of the solution for 1/2 of an
I'm trying to setup a situation where I have agents on POTS lines at remote
locations. I want to allow them to call a DID, log into the Asterisk
system, and be an agent. Ultimately I'd like Asterisk to call them at the
number they were at when they logged in.
Does this functionality exist in
On Thu, 2010-03-04 at 23:27 +, Steve Howes wrote:
On 4 Mar 2010, at 23:11, Steve Edwards wrote:
On Thu, 4 Mar 2010, Steve Edwards wrote:
On Fri, 5 Mar 2010, David @ULC wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create
On Thu, 4 Mar 2010, Matt wrote:
I'm trying to setup a situation where I have agents on POTS lines at remote
locations. I want to allow them to call a DID, log into the Asterisk
system, and be an agent. Ultimately I'd like Asterisk to call them at the
number they were at when they logged
Steve,
Already found it -- but I was under the impression this was deprecated and
removed in 1.6?
On Thu, Mar 4, 2010 at 6:44 PM, Steve Edwards asterisk@sedwards.comwrote:
On Thu, 4 Mar 2010, Matt wrote:
I'm trying to setup a situation where I have agents on POTS lines at
remote
On Thu, 2010-03-04 at 23:27 +, Steve Howes wrote:
On 4 Mar 2010, at 23:11, Steve Edwards wrote:
On Thu, 4 Mar 2010, Steve Edwards wrote:
On Fri, 5 Mar 2010, David @ULC wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create
I am developing a few AGI programs using PHPAGI. This is the first
time developing for Asterisk 1.6 and I am having a lot of problems
reading variables with the $agi-get_variable construct. While an AGI
debug shows me that I am asking for the variable and Asterisk is giving
me the
Andreas Brodmann wrote:
the dialplan currently holds 1792 lines. It's a plain old .conf file.
That's interesting, because I have a dialplan over 2400 lines and it seems to
load fine...
However, I'm using this on an ABE machine which is based on 1.4. Perhaps I'll
try loading this on a 1.6.2
Matt wrote:
Already found it -- but I was under the impression this was deprecated
and removed in 1.6?
Try looking in the doc/ subdirectory of your Asterisk 1.6.2 source. You're
looking for the building_queues.txt file.
Leif.
--
Tzafrir Cohen wrote:
IIRC this issue is fixed in latest SVN, and also in 1.2.6.3-rc2 (1.2.6.5
is based on 1.2.6.2).
Also I just finished releasing several new release candidates which should have
the fix as well if it is indeed resolved.
See the release announcement for the next set of
I'm having this EXACT same problem, I haven;t been able to narrow down the
cause of it yet, but it seems to me that users are receiving notifications
for voicemails in mailboxes that belong to other people, as sometimes their
mail count magically disappears, which I have been suspecting is when
- Chandrakant Solanki solanki.chandrak...@gmail.com escreveu:
Hello
I have successfully compiled OSLEC for echo cancellation for DAHDI
channel.
Is there any way to do echo cancellation for SIP Channel.
Is any, please suggest me.??
Thanks in advance..
--
Regards,
I've got a ton of files in doc but not that file.
On Thu, Mar 4, 2010 at 7:32 PM, Leif Madsen leif.mad...@asteriskdocs.orgwrote:
Matt wrote:
Already found it -- but I was under the impression this was deprecated
and removed in 1.6?
Try looking in the doc/ subdirectory of your Asterisk
On Thursday 04 March 2010 16:51:54 David @ULC wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
One of the nicest things about gsm files is that having no file header,
you can concatenate multiple files and get the same effect
Nobody to take this one!
Am I missing anything in knowing following issue?
--Hi Group,
--Can anybody explain me in detail how the codec translation happens on
--asterisk side when 2 endpoints have different codecs?
--Thanking you in advance.
SM
--
Very informative post Vinícius !
2010/3/5 Vinícius Fontes vinic...@canall.com.br
- Chandrakant Solanki solanki.chandrak...@gmail.com escreveu:
Hello
I have successfully compiled OSLEC for echo cancellation for DAHDI
channel.
Is there any way to do echo cancellation for SIP
Hi All,
Finally I am able to get the number displayed at the SIP side using
exten = _988.,1,Set(CALLERID(num)=8001234000)
exten = _988.,n,Dial(DAHDI/g1/${EXTEN},20)
However this number is fixed and I want to display the number of the
individual lines whoever is calling. I tried with
exten
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