On Mon, 9 Apr 2007, Peer Oliver Schmidt wrote:
Hello Armin (and happy easter),
thanks for you continuing support.
Can you please try HEAD version of SVN trunk (443)?
Did checkout the 443.
It works without any verbosity.
THANK YOU! I'll buy you a beer, if you ever happen to come to
Hi all,
is there anyway i can set SIP_HEADER(To) to the value i like?
--
Regards
Rizwan Hisham
Software Engineer
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Something like
exten = s,1,SetVar(ALERT_INFO=something)
--
Chris Mason
(264) 497-5670 http://www.snapanumber.com/ Fax: (264) 497-8463
http://www.snapanumber.com/
Int: (305) 704-7249 http://www.snapanumber.com/ Fax: (815)301-9759
http://www.snapanumber.com/ UK 44.207.183.0271
No, The problem is, When i dial like this:
Dial(SIP/[EMAIL PROTECTED])
The To header field received on the peer asterisk contains the extension
which i dialed. and whenevr i dial like this:
Dial(SIP/user)
The To Header field received on peer asterisk contains the s extension
instead of the
Steve Prior [EMAIL PROTECTED] Wrote: 4/6/2007 8:30 PM:
Steve Prior wrote:
I've seen in the wiki that it is possible to use a celldock device to
use a cell phone as a PSTN line to Asterisk, but I haven't seen any
comments as to how well this actually works. I was thinking about
hooking a
Rizwan Hisham wrote:
is there anyway i can set SIP_HEADER(To) to the value i like?
If voip-info is correct, you can read, but you can't change.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
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Hi All,
I would appreciate a lot if you could help me. I have installed Asterisk
1.4.1 and zaptel 1.4.2 on my redhat enterprise linux 4. I have also
installed 1 FXO port card: Digium TDM400P.
After loading zaptel driver I could see my digium card's led glow green.
Tested with zttool that its in
Joe Acquisto wrote:
Steve Prior [EMAIL PROTECTED] Wrote: 4/6/2007 8:30 PM:
Steve Prior wrote:
I've seen in the wiki that it is possible to use a celldock device to
use a cell phone as a PSTN line to Asterisk, but I haven't seen any
comments as to how well this actually works. I was thinking
Hello
I have an office with a T1 that provides 4 (out of 8) analog PSTN
lines thru an adtran board. I want to add 4 more analog lines.
Currently I have a Digium TDM40B. I'm wondering what the best
upgrade path is, where I define 'best' as the solution that
is most likely to work without problems
Christopher Chan wrote:
Welcome to China. Most Hong Kong'ers loathe mainland chinese and if they
could, they would never get a job in China. I get plenty of hack
attempts too from China however I doubt that is due to the same
sentiment in China.
If you want to find someone to blame, please
Look at a channel bank with MGCP/TDMoE/USB interface (there are plenty
around). We too prefer to keep fxs/fxo hardware outside of the * box.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit us at
www.generationd.com
I think it's a small, feather covered appendage.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906
I have tried it, it doesnt work
On 4/9/07, Hermann Wecke [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
is there anyway i can set SIP_HEADER(To) to the value i like?
If voip-info is correct, you can read, but you can't change.
Or a new Digium TDM880B replacing the old TDM40B for only one IRQ...
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Jim Freeze
Envoyé : lundi 9 avril 2007 15:15
À : Asterisk Users Mailing List - Non-Commercial
I never get this far, apparently. While the connection seems to be made, and
calls can be completed (rings, answers) there is no audio. On CLI, I can
see what appears to be call being made and connected. These are x-lite phones
(for testing, one hopes) there appears to be no codec
. . .
I have a Dock-n-Talk at home I use to connect my motorola V60i via a
cable so I can't comment on bluetooth. I needed it because for some
reason I can only get good cell reception in my bedroom. It works well
enough. You can certainly tell you are talking over a cell connection
Using the latest SVN of zaptel and asterisk, I can no longer receive
incoming analog calls. The caller just hears it ringing but Asterisk
doesn't pick up.
I am seeing these error messages:
[Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID
returned with error on
Vijay Gaur wrote:
Hi All,
I would appreciate a lot if you could help me. I have installed
Asterisk 1.4.1 and zaptel 1.4.2 on my redhat enterprise linux 4. I have
also installed 1 FXO port card: Digium TDM400P.
After loading zaptel driver I could see my digium card's led glow green.
I struggled with this one too, try this:
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
I use the above for intercom w/ Sipura SPA-941 and it works.
Asterisk 1.2.17 / extensions.ael
Rizwan Hisham wrote:
I have tried it, it doesnt work
On 4/9/07, *Hermann Wecke* [EMAIL PROTECTED]
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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Neglected to mention the host operating system:
Linux myhost 2.6.20-1.2307.fc5 #1 Sun Mar 18 20:44:48 EDT 2007 i686
i686 i386 GNU/Linux
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Hi,
it seems that there is a bug in asterisk's dtmf mode autodetection.
Assume following sip.conf:
[sipprovider]
disallow=all
allow=g726
dtmfmode=auto
DTMF does not work. It seems rfc2833 mode is chosen despite it being
obvious that this cannot work!
The following configuration is
Only when the chicken is provided with sufficient stimulation.
Salvatore Giudice wrote:
I think it's a small, feather covered appendage.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins
Sent: Monday, April 09, 2007 9:21 AM
To: Asterisk
On Apr 9, 2007, at 8:32 AM, [EMAIL PROTECTED] f6hqz-
[EMAIL PROTECTED] wrote:
Or a new Digium TDM880B replacing the old TDM40B for only one IRQ...
Do you know if this board will fit in a 2U machine?
Thanks
Jim
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De :
On 4/9/07, Jim Freeze [EMAIL PROTECTED] wrote:
Or a new Digium TDM880B replacing the old TDM40B for only one IRQ...
Do you know if this board will fit in a 2U machine?
The TDM800P is about the same height as the TDM400P and is about an
inch longer, so you should have no problem putting it in
Arik Raffael Funke wrote:
Hi,
it seems that there is a bug in asterisk's dtmf mode autodetection.
Assume following sip.conf:
[sipprovider]
disallow=all
allow=g726
dtmfmode=auto
DTMF does not work. It seems rfc2833 mode is chosen despite it being
obvious that this cannot work!
Why could
Is there a way to use privacy manager without requiring the user to
enter their name? Essentially I am just looking for a way to force the
called user to enter 1 to accept the call. I don't need a name
recording. I want a call to come in, a message to be played, music on
hold, call out to
On Apr 9, 2007, at 9:29 AM, William Moore wrote:
On 4/9/07, Jim Freeze [EMAIL PROTECTED] wrote:
Or a new Digium TDM880B replacing the old TDM40B for only one
IRQ...
Do you know if this board will fit in a 2U machine?
The TDM800P is about the same height as the TDM400P and is about an
inch
I dont understand it
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
whats it doing here?
On 4/9/07, Karl J. Vesterling [EMAIL PROTECTED] wrote:
I struggled with this one too, try this:
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
I use the above for intercom w/ Sipura SPA-941 and it
Hi Stephen,
I made the call from outside phone(cellphone) to my vonage number. Call
went to the voicemail. My extension.conf has
[incoming]
exten=s,1,Answer() and more lines.
I assume core show channels should show 1 active channel. Its showing 0.
Thanks a lot again Stephen.
Regards,
Vijay
The higher price on the Sangoma is for hardware echo cancellation. There
should be a model (A20400) that doesn't have the echo cancellation and it
probably is less expensive than the Digium card.
Bobby
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim
Vijay Gaur wrote:
Hi Stephen,
I made the call from outside phone(cellphone) to my vonage number.
Call went to the voicemail. My extension.conf has
[incoming]
exten=s,1,Answer() and more lines.
I assume core show channels should show 1 active channel. Its showing 0.
Your calls are not
Guys, i have solved my problem thru different means, i only need to pass the
dnid when the user is using asterisk to regiter as a peer. so...here is
my solution
exten= 123,1,Gotoif($[${SIPPEER(abc:useragent)} = Asterisk PBX]?20:30)
exten= 123,20,Dial(SIP/[EMAIL PROTECTED],,Tt)
exten=
I hear ring few times and then it goes to voice mail. Looks like call is not
going to asterisk. My regular phone attached to that line works fine.
Regards,
Vijay
On 4/9/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Vijay Gaur wrote:
Hi Stephen,
I made the call from outside phone(cellphone)
Joe Acquisto wrote:
. . .
I have a Dock-n-Talk at home I use to connect my motorola V60i via
a cable so I can't comment on bluetooth. I needed it because for
some reason I can only get good cell reception in my bedroom. It
works well enough. You can certainly tell you are talking over a
cell
Vijay Gaur wrote:
I hear ring few times and then it goes to voice mail. Looks like call is
not going to asterisk. My regular phone attached to that line works fine.
When you plug the phone into the port on the Vonage ATA that you're
using to connect to Asterisk, the phone rings when you call
Just curious,
Christopher, what is a chicken boner?
Sorry, that's anti-spammer jargon for spammer. I used to be a mail admin
for an outfit that handles over 40 million mailboxes and over 200
million email transactions daily. Guess what composed the majority of
the daily 200 million
Can someone give me a little detail as to what this error message
means and why it may be occuring?
I keep seeing tons of these roll by on the CLI on one of our systems.
Thanks!
Apr 9 11:05:40 WARNING[19263]: chan_iax2.c:7538 socket_read: Received mini fra
me before first full voice frame
How do you guys like the 330 and 320? I've been looking at this as my
standard phone, since it's relatively cheaper than the 501 which is the
phone I currently push with my PBX systems. Most of my customers do not use
more than one line per phone, so having 3 lines on the 501 is not
necessarily
Yes when I plug my phone to vonage adapter it rings fine.
I will run and send you the output soon.
Thanks
Vijay
On 4/9/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Vijay Gaur wrote:
I hear ring few times and then it goes to voice mail. Looks like call is
not going to asterisk. My regular phone
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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Hi all,
I am having the following trouble with recording calls:
When calls come into the support line did number, the call starts to
record on the first queue, but appears to hang up when the call actually
connects to the engineer (ie I see got hangup request on the cli and
then mixmonitor ends.)
Apparently it sets a SIP_HEADER variable named Call-Info to a value of
answer-after=0 effectively telling the Sipura to answer the call and
put it through to speakerphone.
I will say that extensions.ael is a bit different from regular line
based extensions.conf in that I seem to have to escape
. . .
Can't be worse than my POTS lines. The cable runs here are about 30
years old, and run underground, supposedly, where crossing a
government right of way. This run is ancient, as well.
Supposedly, during wet weather, this becomes a grounding problem.
Certainly the audio quality
Hi Joe,
The debug trace you've enclosed is a NOTIFY message sent from * for the
message waiting feature - and is not related to the call.
You can however tell that something is wrong since the message is being
retransmitted since the server didn't receive 200 OK in reply - while it
could be due
On 9 Apr 2007, at 17:10, voiplist wrote:
Can someone give me a little detail as to what this error message
means and why it may be occuring?
I keep seeing tons of these roll by on the CLI on one of our systems.
Thanks!
Apr 9 11:05:40 WARNING[19263]: chan_iax2.c:7538 socket_read:
Received
Joshua Colp wrote:
Arik Raffael Funke wrote:
The auto setting also does not encompass the info
DTMF option for sending.
Thanks. I was not aware of this.
Ragards,
- Arik
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Hi.
Is there a way to isolate what shows on CLI to just the conversation with that
extension? There appears to be a lot of stuff unrelated to this extension.
Packet traces are not out of the question, but cannot be done today.
joe a.
Yossi Ben Hagai [EMAIL PROTECTED] Wrote: 4/9/2007 12:56
Could someone please remove this person from the list. It seems that the
person is saying they will be away for (9) nine months, with their
auto-reply set.
dave
On Mon, 2007-04-09 at 18:26 +0200, [EMAIL PROTECTED] wrote:
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre
David,
It's not US format. He's away April 4th through April 11th. There was a big
discussion about FB and his absence on this list a few days ago.
Alex
On 4/9/07, David Boyd [EMAIL PROTECTED] wrote:
Could someone please remove this person from the list. It seems that the
person is saying
Joe Acquisto wrote:
Sometimes it's just a matter of finding a clean pair in the cable. Have
you tried asking Verizon to fix the problem?
Don't get me started. That's how I know so much about the situation.
They seem disinclined to address the matter, except with happy talk about
FIOS in my
On Mon, 2007-04-09 at 09:47 -0500, Peder @ NetworkOblivion wrote:
Is there a way to use privacy manager without requiring the user to
enter their name? Essentially I am just looking for a way to force the
called user to enter 1 to accept the call. I don't need a name
recording. I want a
Want to speak French? Just pick up the phone
http://r.smartbrief.com/resp/gChsiHrfsXpnbACibucXDiJM
A new mobile-phone service provides users with pictures of fictitious
lovers, and even fake conversations that let you pretend you're speaking
French. So many people make fake phone calls to impress
Robert La Ferla wrote:
Neglected to mention the host operating system:
Linux myhost 2.6.20-1.2307.fc5 #1 Sun Mar 18 20:44:48 EDT 2007 i686 i686
i386 GNU/Linux
You also neglected to mention the version of Asterisk you are running;
'latest SVN' means nothing when there are 20+ branches of
On Monday 09 April 2007 02:48:32 pm Steve Murphy wrote:
On Mon, 2007-04-09 at 09:47 -0500, Peder @ NetworkOblivion wrote:
Is there a way to use privacy manager without requiring the user to
enter their name? Essentially I am just looking for a way to force the
called user to enter 1 to
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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To
Stephen Bosch [EMAIL PROTECTED] Wrote on: 4/9/2007 2:16 PM:
Joe Acquisto wrote:
Sometimes it's just a matter of finding a clean pair in the cable. Have
you tried asking Verizon to fix the problem?
Don't get me started. That's how I know so much about the situation.
They seem disinclined to
I just opened 0009509 and used Explicit Call Acceptance as the name.
Ben Klang wrote:
On Monday 09 April 2007 02:48:32 pm Steve Murphy wrote:
On Mon, 2007-04-09 at 09:47 -0500, Peder @ NetworkOblivion wrote:
Is there a way to use privacy manager without requiring the user to
enter their name?
Hi Folks,
I have a T100P ordered from Digium in the past. Its working perfectly fine
but I now need to more the server to India (out-sourcing) and we will be
getting an E1 line there.
Does anyone know if T100P was capable (upgradeable) or swappable so that I
could use it with the E1 lines
Hey all,
The Toronto AUG has been working with Clue.ca and IT360
(LinuxWorld/NetworkWorld), and has put together a mini-asterisk
conference within their larger conference:
http://www.it360.ca/asterisk.cfm
If you're interested, as an 'association' we get 25% off the listed
prices. Our dicount
There are 6 different ${QUEUESTATUS} variable values defined in asterisk
1.2, I am attempting to make sure I have a full understanting of when
they would be set;
If someone could correct errors with these definitions ot would be
appreciated;
TIMEOUT - the max time specified in the queue
On Mon, 2007-04-09 at 12:50 -0700, Ritesh Agrawal wrote:
Hi Folks,
I have a T100P ordered from Digium in the past. Its working perfectly
fine but I now need to more the server to India (out-sourcing) and we
will be getting an E1 line there.
Does anyone know if T100P was capable
Mike [EMAIL PROTECTED] wrote:
How do you guys like the 330 and 320?
Mike,
As far as I am aware, neither of these handsets are presently shipping from
Polycom, so most people's experience will be limited to PDF brochures and
pretty pictures. On the face of it, this looks like a good
A couple of weekends ago I decided to see if I could get Asterisk to
play nice with TellMe's VoiceXML studio. They provide the VoiceXML
studio for free, and you can access it through SIP, so I thought this
would be a fun and cheap way to integrate voice recognition into my
IVR. I have posted a
Ah, thanks. I didn't realize this.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Nickerson
Sent: Monday, April 09, 2007 16:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 330/320
Hi Josh, fantastic implementation.
It's a real shame tellme doesn't think that the 30,000+ Asterisk
installations don't warrant an ASP prepaid solution.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).
Mike,
I don't have much information, except they are due for shipment soon
(mid to end of April to distribution from Polycom). We've demoed a
couple and I personally believe they'll be a tough phone to find in
stock for the first few months their released. Demand on these from
what I'm
Hello list members.
I would like to know how to playback an audio file to the caller, and while
it's played asterisk to continue executing the next priorities on
extensions.conf
That's not the case when using playback command, because the next priority
is executed until the audio file ends
On Mon, 9 Apr 2007, Alejandro Mej?a wrote:
I would like to know how to playback an audio file to the caller, and while
it's played asterisk to continue executing the next priorities on
extensions.conf
That's not the case when using playback command, because the next priority
is executed until
Hi,
I like my recordings to go to date wise folder i mean to say that for
example today is 20070409 so all recordings should go directly to that
folder instead of one folder for whole month. and then next day's recordings
should go to next date folder.
so how can i do that my current monitor
In a system connected to a verizon T1, Digium TE411P (quad T1 echo
cancellation), client is complaining it is too quiet.
The complaint regards calls over the T1, not in house SIP only calls.
Their description indicates they want some earpiece feedback of themselves
speaking. Also, they
Thanks Steve.
I'll try what you suggest.
Cheers!
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Steve Edwards
Enviado el: Lunes, 09 de Abril de 2007 04:08 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users]
On Tue, 10 Apr 2007, Faisal Ashraf wrote:
I like my recordings to go to date wise folder i mean to say that for
example today is 20070409 so all recordings should go directly to that
folder instead of one folder for whole month. and then next day's recordings
should go to next date folder
Joe Acquisto wrote:
In a system connected to a verizon T1, Digium TE411P (quad T1 echo
cancellation), client is complaining it is too quiet.
The complaint regards calls over the T1, not in house SIP only calls.
Their description indicates they want some earpiece feedback of
themselves
Stephen Bosch [EMAIL PROTECTED] Wrote: 4/9/2007 7:12 PM:
Joe Acquisto wrote:
In a system connected to a verizon T1, Digium TE411P (quad T1 echo
cancellation), client is complaining it is too quiet.
The complaint regards calls over the T1, not in house SIP only calls.
Their description
Joe Acquisto wrote:
Stephen Bosch [EMAIL PROTECTED] Wrote: 4/9/2007 7:12 PM:
Have you been able to test this yourself? (Three to four seconds seems
inordinately long. That's as bad as a satellite link.)
No, not tested by me, I only heard about it today, via email.
I don't doubt that they
On Apr 9, 2007, at 1:51 PM, Kevin P. Fleming [EMAIL PROTECTED]
wrote:
You also neglected to mention the version of Asterisk you are running;
'latest SVN' means nothing when there are 20+ branches of Asterisk on
our SVN server.
Sorry about that. It is the 1.4 trunk:
Asterisk
Is there a way to use privacy manager without requiring the user to enter
their name? Essentially I am just looking for a way to force the called
user to enter 1 to accept the call. I don't need a name recording. I
want a call to come in, a message to be played, music on hold, call out
Hello
My asterisk is receiving calls from OpenSER but all calls hangup in 20
seconds.
This only happens because Im using Asterisk2Billing's AGI (without
A2Billing it doesnt hang up).
does someone knows whats the problem??
Here is my Asterisk debug:
(xxx.xxx.xxx.xxx - the phone's IP)
Apr
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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asterisk-users mailing list
To
In the case of our Sangoma card, the echo cancellation module
constituted approximately half the price of the card, so yes you should
find it considerably cheaper than the Digium card.
Just be aware of the extra fiddling around having to install the Sangoma
drivers in addition to the Zaptel
; charset=iso-8859-1
Hi,
I like my recordings to go to date wise folder i mean to say that for
example today is 20070409 so all recordings should go directly to that
folder instead of one folder for whole month. and then next day's
recordings
should go to next date folder.
so how can i do that my
On 4/9/07, Damon Estep [EMAIL PROTECTED] wrote:
There are 6 different ${QUEUESTATUS} variable values defined in asterisk
1.2, I am attempting to make sure I have a full understanting of when they
would be set;
If someone could correct errors with these definitions ot would be
appreciated;
On Mon, 9 Apr 2007, Steve Edwards wrote:
On Tue, 10 Apr 2007, Faisal Ashraf wrote:
I like my recordings to go to date wise folder i mean to say that for
example today is 20070409 so all recordings should go directly to that
folder instead of one folder for whole month. and then next day's
The jury found that three of five disputed patents were infringed and
all are valid, while rejecting Verizon's claim that the infringement was
willful. The patents cover a method of translating calls between the
Internet and standard phones, call-waiting features and wireless handsets.
Is it
Has anyone tried pushing calls to a Vonage ATA attached to an FXO card in
Asterisk and had your account terminated by Vonage?
I'm curious as to whether they will stop your service if you push too many
calls through their ATA in a specific period of time.
Thanks in advance for the info, SG
I
Does anybody have callerid name coming in on a Cisco PRI via a Cisco
gateway via SIP to *? I've seen a few people ask and a few people that
say it should work, but I've never seen an actual working config.
I do a debug on our Cisco gateway and I can see the callerid name,
however none of the
I have a PRI fed into a Cisco AS5300 media gateway and sent to Asterisk via
SIP, and caller ID works fine. I can probably help you figure it out, even
though I don't have any immediate insights. Feel free to e-mail me off
list if you like.
In general, however, I don't have isdn
Does anybody have callerid name coming in on a Cisco PRI via a Cisco
gateway via SIP to *? I've seen a few people ask and a few people that
say it should work, but I've never seen an actual working config.
I have it working, but it depends on the specific configuration. I have it
working via
Hi folks.
My client is wanting to use call forwarding configured on their phones
(Linksys SPA942), with a PRI from their provider. When we configure call
forwarding, we invariably get a The number you have dialed is not in
service message from the providers.
Examining the detailed dial plan
I have a Digium TDM400b11, 1FXO [port2] 1FXS [port 1]
When I reload the chan_zap I get:
[chan_zap.so] = (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Apr 9 22:39:36 ERROR[3541]: chan_zap.c:10388 setup_zap: Signalling must be
specified before any channels are.
Apr
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