On 9 Sep 2006, at 00:42, Kevin Smith wrote:
Hi everyone,
I am looking to log CDR records to our MSSQL database for further
examination on the records. From what I gathered from the wiki I
have to choose between FreeTDS and unixODBC. Is there a better
choice? Which option would be better
Hello,
my name is dominik, and i'm using asterisk with
voip without isdn, only sip.
I'm using Asterisk Version 1.0.7 on Debian
3.0.
I've configured the fax receive in the
/etc/asterisk/extensions.conf:
exten =>
99,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => 9
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I can confirm the same problem, it looks like the oct612x directory tree
is missing from the tarball
Stu
Bill Maidment wrote:
> Hi
> I've just tried to compile the zaptel-1.2.9 release and I get the
> following error:
>
> HOSTCC /usr/local/src/zap
Hi
I've just tried to compile the zaptel-1.2.9 release and I get the
following error:
HOSTCC /usr/local/src/zaptel-1.2.9/wct4xxp/fw2h
/usr/local/src/zaptel-1.2.9/wct4xxp/fw2h
/usr/local/src/zaptel-1.2.9/wct4xxp/OCT6114-128D.ima
/usr/local/src/zaptel-1.2.9/wct4xxp/vpm450m_fw.h
make[3]: *** No rul
Try this
exten => s,1,Disa(no-password)
It's a dirty hack, but it might work. It will dump the phone straight into
the disa application, which will play dialtone and allow you to dial into the
current context.
-Tim
On September 8, 2006 16:12, Henrik Woffinden wrote:
> That's exactly what happ
You need to make a cron job that runs a script that creates a spool file.
If you dump a file into /var/spool/asterisk/outgoing/ it will create a call
based on the information within the file. Do a search for asterisk spool
outgoing for all the formats and examples. I have used it and it works
Hello,
While trying to compile the new zaptel 1.2.9 on a Fedora Core 4 system, i get
the following errors.
it is looking for octasic-helper? which it can't find... am I missing
something that is needed for this new version?
thanks,
earl
rm -f version.h.tmp
/lib/modules/2.6.17-1.2142_FC4/build
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Hall, Eric M. wrote:
>
> Hello group
> I have a customer that has asked me to build an auto dialer that will
> call customer a few day before an appt and remind them of the time and
> date of the appt.
>
> Does anyone have any good links for apps t
Bruce,
How do you go about accomplishing configuring the phone,
zipping it up and sending it over to your family?
Thanks
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
ReevesSent: Thursday, September 07, 2006 8:37 AMTo:
Asterisk Users Mailing List - Non-Commercial
himy asterisk -r shows me Most of the times Outgoing Spool Failed. Can some one tell me why is it happening and how to solve this issue. Is it a problem ? thanks in advance.arun
___
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asterisk-users ma
Mike wrote:
Thanks Tony. Its possible that the phone stops sending RTP stream (but it
certainly is receiving some!). How do I get Asterisk to stop caring whether
it receives RTP or not?
Yes there is a NAT between the phone the the Internet. The Asterisk server
doesn't have NAT though.
My Son
They recommended changing the default value of 1000 up or down
incrementally until it works better. We’re currently at 2000, and we’re
still not completely free of events.
What were the proposed changes to VPM_DEFAULT_DTMFTHRESHOLD ? -- -- Steven http://www.glimasoutheast.org
That would be problematic. I am using a cheap Linksys router where my
Polycom 501 is located and I see no such setting. It probably is hardcoded.
Can I force the Polycom 501 to send empty RTP packet?
(actually, I tried using comfort noise but I got an asterisk error message
rtp.c:330 process_r
Several Linksys models have had a problem in the past allowing multiple
devices on the inside lan to nat properly with something on the outside wan.
Ordinarily a sip phone on the inside of the lan attempts to register
with an external asterisk box, and the Linksys keeps track of source IP,
sou
I have to say, I had quite a chuckle imagining "zip /dev/phone_sitting_on_desk" being run on someone's command prompt =)-brandonOn 9/7/06,
Ferguson, Michael <[EMAIL PROTECTED]> wrote:
Bruce,
How do you go about accomplishing configuring the phone,
zipping it up and sending it over to your
Hello
group
I have a customer that has asked me to build an auto dialer that
will call customer a few day before an appt and remind them of the time and date
of the appt.
Does
anyone have any good links for apps that could do this type of auto calling?
They also request that information
The Asterisk Development Team is pleased to announce new releases of Asterisk
and Zaptel!
Asterisk 1.2.12 includes a number of bug fixes, including fixes for two
regressions that occurred in the 1.2.11 release. Specifically, the AGI 'GET
VARIABLE' command has now gone back to its previous beha
You also have to make sure that on the web config for Grandstream that you allow it to receive auto-answer (or something to that effect).
Ok, actually it's under the settings for the Lines and is called: Allow Auto Answer by Call-Info:
Make sure Yes is selected here.
You can use what Barry h
Oh my god, it works! This is a nice friday! Thanks for the tip. In the
end, what I did was I retraced my steps, and I realized I hadn't
connected the card to a power cable, and I had the signalling on
zapata.conf backwards... Also I disabled most of the devices on the
motherboard, including USB (g
I don't know if this helps, but this is how my 80+ Polycom phones are
set up.
dialplan.digitmap="9,1[2-9]xx[2-9]xx|9,[2-9]xx|[2-8]xxx|9,[2-9]11|911|1x|9,011x.T|*xx"
dialplan.digitmap.timeOut="5"/>
Mike wrote:
Here it is:
dialplan.digitmap.timeOut="3"/>
When I dial 845
Here it is:
When I dial 845, I get fast busy. When I dial 9-555-555-, it dials
without the need to press send. All good result.
When I dial 9-555-5 and wait, nothing happens
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of E
I need to set up some SIP extens in such a way that a beep
plays back once per minute on an active call to remind the calling party that
their call is being recorded. Can anyone provide pointers? Thanks.
Dallas Carter
Mailmessenger Networks, Inc.
dallas (at)
mailmessenger (dot) com
wcfxo is used only for the X100P (and some clones). It is not used for
any other card. wctdm supports both FXO and FXS. Maybe you are just
confused about which module is associated with which channel.
Iván Vega R. wrote:
Upon further investigation, I tried the following:
lsmod | grep 'wc*'
Upon further investigation, I tried the following:
lsmod | grep 'wc*'
I can see wctdm (I believe this is the fxs module, no?), wcfxo,
zaptel... so I think so far so good. Then:
cat /proc/interrupts
There I only see the wctdm:
50:9556184 0 IO-APIC-level uhci_hcd:usb3, wctdm
I
Hi everyone,
I am looking to log CDR records to our MSSQL database for further
examination on the records. From what I gathered from the wiki I have to
choose between FreeTDS and unixODBC. Is there a better choice? Which
option would be better in the log run?
Also configuration asterisk to u
What IS your Polycom dialplan, and do you have the digit.impossiblematch
set?
Eric "ManxPower" Wieling wrote:
Then you are doing something else wrong. If the call gets to Asterisk
then the exten => lines I gave should match if they are in that context.
I use this all the time.
Mike wrote:
Then you are doing something else wrong. If the call gets to Asterisk
then the exten => lines I gave should match if they are in that context.
I use this all the time.
Mike wrote:
But that's the whole freaking problem!!!
If I could do that, I would. But Asterisk keeps on sending the "484 Ad
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Mike wrote:
> But that's the whole freaking problem!!!
>
> If I could do that, I would. But Asterisk keeps on sending the "484 Address
> incomplete" message, and the Polycom keeps on waiting silently and patiently
> for me to put in the needed extra d
Thanks Dave. Unfortunately I've been through this already. I understand
that digitmap are used to automatically press "send" when a certain pattern
is reached. Nowhere can I say "if the pattern isn't fully match within x
seconds then consider it a bad extension".
That`s the only relevant thing
But that's the whole freaking problem!!!
If I could do that, I would. But Asterisk keeps on sending the "484 Address
incomplete" message, and the Polycom keeps on waiting silently and patiently
for me to put in the needed extra digit(s).
When I pick up my home phone, and I forget a number, the
Hi yet again,
Is a TDM13B card the correct one if I have three phone lines and I
want to use the extra port to connect a "normal" phone?
Thanks!
___
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asterisk-users mailing list
To UNSUBSCRIBE or u
Not much you can do about that other than:
exten => _X.,1,Playback(dial-real-number-you-moron)
exten => _X.,2,Hangup
Mike wrote:
That's a good idea, and I tried, but as far as I know the digitmap setting
of the Polycom allows me to enable the phone to dial automatically after a
pattern is used
That's a good idea, and I tried, but as far as I know the digitmap setting
of the Polycom allows me to enable the phone to dial automatically after a
pattern is used (ex : [9]xx), but it doesnt allow me to consider a
too short string as being invalid (ex if I miss a digit and just dial
9-5
Hi again!
So while ztcfg ran without errors, now asterisk won't run. Here's the
relevant part of the error:
WARNING[13055]: chan_zap.c:921 zt_open: Unable to specify channel 1:
No such device
ERROR[13055]: chan_zap.c:6879 mkintf: Unable to open channel 1: No
such device here = 0, tmp->channel =
Amazing how one can find an answer seconds before asking for help... I
was confused with zaptel.conf... I changed the order of the channels
there and it works now, hehe.
On 9/8/06, Iván Vega R. <[EMAIL PROTECTED]> wrote:
> Hi everyone,
>
> I'm new on Asterisk. I'm trying to follow a few tutorial
Or better yet, set dialplan.impossibleMatchHandling to 2. This should
disable earlydial altogether.
CP
On Sep 8, 2006, at 2:49 PM, Eric "ManxPower" Wieling wrote:
Mike wrote:
> It's not a silly idea, I've been doing some sniffing and debugging
with my
> limited knowledge of the whole process
Hi everyone,
I'm new on Asterisk. I'm trying to follow a few tutorials on the net,
and fortunately this has been the only stumbling block so far.
I do:
modprobe zaptel
modprobe wcfxs
modprobe wcfxs
modprobe wcfxs
modprobe wcfxo
I have this on zapata.conf:
[channels]
busydetect=1
busycount=7
re
You do not mention the device you are using.
I'll assume Zap.
Enable three way calling and conference in zapata.conf then use FLASH.
Bart Fisher wrote:
It appears the only way to cause a 3-way call (or a screened transfer)
is by using conference - nasty
This mean SLT would need to transfer to
Mike wrote:
It's not a silly idea, I've been doing some sniffing and debugging with my
limited knowledge of the whole process. I found this in the debug stream
after having dialed "965").
Notice this line: SIP/2.0 484 Address Incomplete.
Is this what I was suspecting, that it knows it could ma
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Mike wrote:
> It's not a silly idea, I've been doing some sniffing and debugging with my
> limited knowledge of the whole process. I found this in the debug stream
> after having dialed "965").
>
> Notice this line: SIP/2.0 484 Address Incomplete.
>
If the extension is 5156598, then you need to have member=>Agent/5156598
in queues.conf (and the equivalent entry in agents.conf) or add
agent/5156598 as a dynamic member, not agent1 or agent2 etc
Julian.
gc wrote:
The calling extension is 5156598.
After I dial into 881112 from this phone. It
Do you go any further than a casual glance and a lot of opinion in your
spoutings?
Me -- my site currently points to an affiliate program (and a lousy one
at that). I receive a small percentage from each customer who signs up,
so as such, I wouldn't be responsible anyway. That said, it's not
It appears the only way to cause a 3-way call (or a screened transfer)
is by using conference - nasty
This mean SLT would need to transfer to conference than add second
party, then add themselves.
I've searched and I can't find anything that works in asterisk like the
Telco method or am I blin
Steven wrote:
Because the Telco is government owned.
They are the PSTN, so only they can route and charge for PSTN calls.
Making a call from an Indian office to a US office over VOIP is legal.
Forwarding a PSTN call over that same VOIP trunk is illegal.
In other countries where the Telco is n
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Arun Kumar wrote:
> hi
>
> thanks for reply.
>
> I'm using vicidial to make calls at 2.0 dial level it is able to make calls
> but when I see the asterisk -r most of the time it shows Outgoing Spool
> Failed. Which Spool File ?
Er, probably the best
It's not a silly idea, I've been doing some sniffing and debugging with my
limited knowledge of the whole process. I found this in the debug stream
after having dialed "965").
Notice this line: SIP/2.0 484 Address Incomplete.
Is this what I was suspecting, that it knows it could match a pattern
For some reason your phone is dialing an empty extension as soon as you
go off hook.
"exten => s" would be the same as "exten => ''"
Henrik Woffinden wrote:
immediate is already set to "immediate=no", so that's not it.
Best regards,
Henrik Woffinden
Eric "ManxPower" Wieling wrote:
Remove
Because the Telco is government owned.
They are the PSTN, so only they can route and
charge for PSTN calls.
Making a call from an Indian office to a US office
over VOIP is legal.
Forwarding a PSTN call over that same VOIP trunk is
illegal.
-- -- Steven
http://www.glimasoutheast.org
immediate is already set to "immediate=no", so that's not it.
Best regards,
Henrik Woffinden
Eric "ManxPower" Wieling wrote:
> Remove immediate=yes from /etc/asterisk/zapata.conf
>
> Henrik Woffinden wrote:
>> That's exactly what happens:
>>
>> When I pick up the handle, this is what I get:
>>
Remove immediate=yes from /etc/asterisk/zapata.conf
Henrik Woffinden wrote:
That's exactly what happens:
When I pick up the handle, this is what I get:
-- Extension 's' in context 'from-inside' from '11' does not
exist. Rejecting call on channel 0/2, span 2
Do you know what to do in the
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Mike wrote:
> Thanks Tim.
>
> I've been trying to find out what's happening. Basically, somehow, it seems
> that my Polycom 501 knows what extensions are valid and which aren't in my
> dialplan. Obviously, the 501 doesn't really know that, but Aster
Hi,I'm trying to connect my cell phone (motorola V3) to asterisk, using this guide: http://www.thetechguide.com/howto/asterisk/chanbluetooth.html
Everything has worked ok, but when I actually want to start asterisk, my phone doesn't connect all the way. All I'm getting in the asterisk CLI is this:[
Mike wrote:
Let's just take 1) and 2). Why is Asterisk not going into the i extension
like it should?
Because the "i" extension is for IVRs and things like that.
___
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asterisk-users mailing lis
Protectionism; it’s not that
uncommon. Any number of countries around the world still have similar laws.
(even Australia until about 1998 I think).
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brandon Galbraith
Sent: Friday, 8 Septem
Brandon Galbraith wrote:
Steve,
Forgive my ignorance, but why does India institute that policy?
Why does France blow up bombs in the south pacific? Each country can do
as it pleases - unfortunately - but that is also good for us VoIP
carriers because it creates and protects high retail pr
That's exactly what happens:
When I pick up the handle, this is what I get:
-- Extension 's' in context 'from-inside' from '11' does not
exist. Rejecting call on channel 0/2, span 2
Do you know what to do in the dialplan?
Best regards,
Henrik Woffinden
Tim St. Pierre wrote:
> Could you
Steve,Forgive my ignorance, but why does India institute that policy?-brandonOn 9/8/06, Steven <
[EMAIL PROTECTED]> wrote:
Even in India, you can use VOIP for overseas calls
coming from your own company.
You just can't sell services that allow people to
call a PSTN number and then have thei
Even in India, you can use VOIP for overseas calls
coming from your own company.
You just can't sell services that allow people to
call a PSTN number and then have their call sent over VOIP to another
location.
-- -- Steven
http://www.glimasoutheast.org
"Alex Robar" <[EMAIL PROTE
Lol, ok - last post on the topic...I promise.
I'm going to let Ze Frank have the last word about the state of
government policy under the current administration and the role it plays
in the international stewardship here in the USA
http://www.zefrank.com/theshow/archives/2006/09/share_090806.html
Now that is really odd.
Try sip debug peer (peername of the polycom)
This will let you see the sip packets go by when you do this, so you can see
the responses it is, or isn't getting.
I'll have to look up the SIP response codes, but I do know that there is one
for "not found" which should cor
This isn't working for me either. I was about to ask this same question, but discovered this recent thread.
I have the following set up in my extensions.conf file, as per Granstream instructions:
[macro-page-grandstream]
exten => s,1,ChanIsAvail(${ARG1}|js); j is for jump, s is for ANY call
Mike wrote:
I've been running into an issue with my Polycom 501 and Asterisk.
I realized, after much mucking around, that when I dial a number (and press
the send key) that is invalid , but could still match an Asterisk pattern
(example: I dial 567, which is not a valid extension, but my diapl
Thanks Tim.
I've been trying to find out what's happening. Basically, somehow, it seems
that my Polycom 501 knows what extensions are valid and which aren't in my
dialplan. Obviously, the 501 doesn't really know that, but Asterisk seems
to return it this info (sort of :"valid", "invalid" or "cou
Could you send us some CLI output?
Look for something like this
Invalid extension "s" in context
It could be that lifting the handset without dialing is opening a channel to
the "s" extension, since there are no digits being dialed. There is a
workaround for this, but it means creating a dia
Tony Mountifield wrote:
Try enabling intense PRI debugging "pri intense debug span N". You may want
to direct the PRI debugging to a file with "pri set debug file filename".
It's not clear from the log you posted whether q931_hangup() was called
because of a Q.931 message Asterisk received, or
Isn't there a way to specify a context based on the incoming domain in
sip.conf?
On September 8, 2006 14:03, Ricardo Carvalho wrote:
> So... does anybody know how can I do this?
> Maybe using a way to distinguish users not by their username, but by
> other fields of SIP INVITE messages?
>
> Re
With SIP, asterisk processes the digits it receives in the invite from the
Polycom.
There is no communication of dialplan information in SIP. The polycom should
send the digits as soon as you press dial. You can program the polycom with
a dialplan that will tell it when to send the digits, bu
Check your Dial() string to make sure that you haven't mistyped and put
gafachi-o instead of gafachi-out. Specifiying the full host name will also
work.
As a hint, you can refresh these changes with out restarting your server (and
therefore without disrupting any calls in progress)
extensions
Hi,
I'm using Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s.
I've got 3 ISDN phones attached.
When I want to dial out I can do it in 2 ways..
1) Type in number with handle still on.. Lift handle and we dial the
number
2) Lift handle and then press the number
Both methods should work, but only
This looks like a networking issue - asterisk isn't receiving any
replies to signaling packets and assumes that the UA is no longer
reachable.
CP
On 8-Sep-06, at 10:33 AM, Noc Phibee wrote:
anyone know this error ??
Noc Phibee a écrit :
> Hi
>
> today, i have a big problems with my ast
hithanks for reply.I'm using vicidial to make calls at 2.0 dial level it is able to make calls but when I see the asterisk -r most of the time it shows Outgoing Spool Failed. Which Spool File ?thanks
arunOn 9/8/06, Matt Riddell (IT) <[EMAIL PROTECTED]> wrote:
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I've been running
into an issue with my Polycom 501 and Asterisk.
I realized, after
much mucking around, that when I dial a number (and press the send key) that is
invalid , but could still match an Asterisk pattern (example: I dial 567, which
is not a valid extension, but my diaplan accep
Tim, this is the way I have Gafachi set up in sip.conf and works well with channels that have an ATA attached to it but not the virtual one. I have changed the host in extensions.conf to the .sip.gafachi.com. But I have calls on the server and cannot restart it yet. I'll keep you poste
Hi all,
I have a box with an ISDN HFC card (1 BRI) connected to an Italian
ISDN, the card is using zaphfc driver and it's receiving and
originating calls quite regularly.
There are some numbers (mostly toll free numbers) that I cannot
connect to, here is what I get from the CLI:
-- Cal
The calling extension is 5156598.
After I dial into 881112 from this phone. It no longer accept call from
queue but the 'show agents' still show it is available.
- Original Message -
From: "Julian Lyndon-Smith" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discus
On 9/8/06, gc <[EMAIL PROTECTED]> wrote:
After using PauseQueueMember in my dialplan. I used 'show agents' cli to
show the agent status. It is still show that agent available. Here is the
output from asterisk console:
5156597 (Agent1 ) available at '[EMAIL PROTECTED]' (musiconhold is
'defau
Friends,
Two times I've taken Asterisk to the SIPit interoperability tests,
both times has led to
much improved functionality and a lot of new ideas. It is important
to meet other
software developers struggling with the SIP standards, wanting to
make sure that
equipment and servers work to
So... does anybody know how can I do this?
Maybe using a way to distinguish users not by their username, but by
other fields of SIP INVITE messages?
Regards,
Ricardo.
Ricardo Carvalho wrote:
In extensions.conf I want to implement a dial plan that distinguishes
the users that wish to dial
The biggest problem with your argument is that VoIP is not illegal anywhere. Voice over _internet_ is, but voice over the Internet protocol is not. Anyone in China is free to setup Asterisk to use within their offices. There's nothing illegal about it. It's using VoIP, but it's not transmitting it
gc wrote:
After using PauseQueueMember in my dialplan. I used 'show agents' cli
to show the agent status. It is still show that agent available. Here is
the output from asterisk console:
5156597 (Agent1 ) available at '[EMAIL PROTECTED]'
(musiconhold is 'default')
5156598 (Agent2 ) no
Do you have gafachi-o in your sip.conf?
Since it's not a valid host name, you need to have an entry in sip.conf to
tell asterisk how to make a call to gafachi-o.
That's why it is telling you "No such host".
On September 8, 2006 12:57, [EMAIL PROTECTED] wrote:
> It sounds like a good idea, I t
After using PauseQueueMember in my dialplan.
I used 'show agents' cli to show the agent status. It is still show that agent
available. Here is the output from asterisk console:
5156597 (Agent1 ) available at '[EMAIL PROTECTED]'
(musiconhold is 'default')5156598 (Agent2 ) not logged in
Yes I am aware he is the second executive to be arrested..the first
is still yet to be charged and is still awaiting trial and has fallen
off the face of the general media which is why I'm 'motivated' to draw
attention and outrage to this second case.
Yes you are right it does belong off this
anyone know this error ??
Noc Phibee a écrit :
Hi
today, i have a big problems with my asterisk ...
when i want call i have this error :
Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1226 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 102 (Critical
Reques
No it’s totally legal. Sportingbet
PLC is a legal and registered company in the USA.
He was arrested in NY for a charge that is
going to be extradited and charged in Louisiana.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Why don't you keep political diatribe to your blog? This is OT, and
quite frankly it displays that you have less than perfect grasp on reality.
Mark Spencer makes a software product that is perfectly legal to use
anywhere in the world, even in India (as long as it stays within a
building and use
IS running a gambling site illegal in the
UK?
If so, perhaps they had a warrant out for him, and
we (US) are just going to extradite him.
-- -- Steven
http://www.glimasoutheast.org
"Dean Collins" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]...
Exactly so why a
Is autoframing set to yes in the [general] section? A current
limitation in the code is that a global autoframing will
override a user/peer setting.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of yusuf
Sent: Friday, September 08, 2006 2:26 AM
To:
It sounds like a good idea, I tried it and get this error
Sep 8 09:52:17 WARNING[27193]: chan_sip.c:1968 create_addr: No such host: gafachi-o
Sep 8 09:52:17 NOTICE[27193]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
In Extension
Exactly so why aren’t they trying to
arrest the 50 million people in the USA who have gambled online?
Mark (as far as I know) isn’t actively checking
with asterisk users for what country they are in so therefore in the reciprocal
eyes of the indian government he is similarly breaking th
The guy was arrested because he was conciously allowing US citizens to gamble online, something illegal in the US at this point. He ran a business who's sole profit source came from this illegal activity. Spencer is an entirely different situation. He's not running a business in which he provides V
8 sep 2006 kl. 17.18 skrev Michel Zenone:
Hi!
I try to make my Asterisk contact a SIP user thanks to a redirect
server. In fact Asterisk try to reach a SIP address that is redirected
to the good one.
The error response is:
*CLI> -- Executing Dial("OSS/dsp", "sip/[EMAIL PROTECTED]|
30|
Did this email go through the first time
or is the email down?
I’ve been reading about this online
everywhere this morning and I haven’t seen it come back on the asterisk server.
Cheers,
Dean
From:
Dean Collins
Sent: Friday, 8 September 2006
10:22 AM
To: 'Asterisk
Tony Mountifield wrote:
It looks like the PRI connection is going down first, and when that channel
exits, it causes the SIP channel to be hung up. So concentrate on the PRI.
Yep, that's what I've seen so far. Been trying to concentrate on the
PRI, but not seeing any indication of what is tr
It's all in the Asterisk database, which is a Berkeley DB format as far as I
know. If it's just for migration, you could probably just move the database.
If you want to do this while the system is up, maybe using an external
database for this information would work better.
-Tim
On September
I highly recommend the 3ware line of SATA RAID cards for doing SATA RAID. I've installed them in upwards of 400-500 servers, and they're rock solid cards and affordable.Disclaimer: I do not work for 3ware or AMCC, but am a very satisfied customer.
-brandonOn 9/8/06, shadowym <[EMAIL PROTECTED]> wro
Hi!
I try to make my Asterisk contact a SIP user thanks to a redirect
server. In fact Asterisk try to reach a SIP address that is redirected
to the good one.
The error response is:
*CLI> -- Executing Dial("OSS/dsp", "sip/[EMAIL PROTECTED]|30|
H|g") in new stack
-- Called [EMAIL PROTECTED
It has nothing to do with Asterisk as far as I know. The kernel needs to
support it I believe. DO NOT even attempt SATA RAID without a hardware RAID
card that is supported in the kernel or your asking for headaches. For
non-RAID, many BIOS's emulate IDE so if that is the case you should have no
Hi All, I try to transcode Speex to G711 through Asterisk, however, the voice quality is crappy. Does anyone has a clue to fix this? The current version i running now is 1.2.4, and the Speex codec was sent from eyeBeam.The following warning below:Sep 7 13:48:55 WARNING[5297]: codec_speex.c:278 sp
In article <[EMAIL PROTECTED]>,
Jamin W. Collins <[EMAIL PROTECTED]> wrote:
> I have an asterisk box configured to perform media translation (TDM <->
> SIP). With this configuration, calls are essentially only passing
> through the asterisk box. Thus, I would think that a disconnect request
>
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