Tzafrir Cohen wrote:
Interesting. This part was originally ifdef-ed out in chan_zap:
http://bugs.digium.com/13786
I get a 404 NOT FOUND on that link.
I'll dig up an older version of that code and compare.
Thanks for the tip.
Jim
___
--
When it fails, I get this message:
[Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of
type 'DAHDI' (cause 0 - Unknown)
Can you enable debug logging? Do you see any message about the casue for
that?
Yes, I enabled logging, however, no additional logging was
I've been having trouble with making outbound calls to my
TELCO from a TDM400 card (FXS KS signalling) after upgrading
from 1.6-beta9 to 1.6.0. The problem is completely intermittent.
When it fails, I get this message:
[Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of
, Jim Duda wrote:
Tzafrir,
Thanks for the tip. I'm researching answeronpoliaryswitch. I suspect
this will solve my issue. I never would have know to look for this.
Thanks much! You made my day :-)
Hmm... I might have misled you. By default Asterisk ignores all polarity
events. Using
Inbound calls on DAHDI work fine.
At some point, outbound starts working. I just cannot figure out what
the trigger is. At first, I thought the trigger was receiving at least
one inbound call. But that isn't always true. Once it starts working,
it seems to continue until a restart.
Does anyone know what this error message means?
Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
I've upgraded to 1.6.0 with dahdi 2.0.
For some reason my outbound dahdi calls are not going through.
At some point, it starts to work, but I don't know what the
trigger is. Out of the
Can anyone tell me what this message means?
Got event 17 (Polarity Reversal)...
I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0.
It appears that I get this Polarity Reversal each time an inbound call
hangs up. This results in another ring, but no one is there. It appears
as
Tzafrir,
Thanks for the tip. I'm researching answeronpoliaryswitch. I suspect
this will solve my issue. I never would have know to look for this.
Thanks much! You made my day :-)
Jim
Tzafrir Cohen wrote:
On Fri, Oct 10, 2008 at 12:57:27PM -0400, Jim Duda wrote:
Can anyone tell me what
,
CALLERID(number)=781-736-1994) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/107-b4e703f0,
CALLERID(name)=Jim Duda) in new stack
-- Executing [EMAIL PROTECTED]:3] Set(SIP/107-b4e703f0,
DYNAMIC_FEATURES=outflash) in new stack
-- Executing [EMAIL PROTECTED]:4] Dial(SIP/107
-- Executing [EMAIL PROTECTED]:2] Set(SIP/107-b4e703f0,
CALLERID(name)=Jim Duda) in new stack
-- Executing [EMAIL PROTECTED]:3] Set(SIP/107-b4e703f0,
DYNAMIC_FEATURES=outflash) in new stack
-- Executing [EMAIL PROTECTED]:4] Dial(SIP/107-b4e703f0,
Dahdi/4/18005551212,40,tr) in new stack
[Oct 5
Even that isn't always true.
Sometimes dial out on DAHDI works, sometimes it doesn't.
I'm not sure what makes it start working, but once it does,
it appears to stay working.
Jim
Jim Duda wrote:
I don't know how to explain this.
After receiving 1 inbound call on the DAHDI channel attached
I have a standard analog POTS service attached to a TDM401 card.
My zapata.conf for this line has:
; PSTN connected here
;immediate=no
;busydetect=yes
;busycount=8
;musiconhold=default
mailbox=100
mwimonitor=fsk
mwilevel=512
mwimonitornotify=/usr/local/sbin/zapnotify.sh
faxdetect=incoming
= s,n,Set(DYNAMIC_FEATURES=inflash)
on an outgoing call:
exten = _1XX,1,Set(DYNAMIC_FEATURES=outflash)
in incoming calls the user has to press *4
on outgoing calls the user has to press *3
On Sat, Aug 2, 2008 at 4:59 PM, Jim Duda [EMAIL PROTECTED] wrote:
I've seen a few posts
I've seen a few posts on this issue, however, no definitive answer.
My PSTN is connected to Zap/4. I have simple Call Waiting service on
the PSTN line.
All the other phones are SIP clients.
When I'm on an Zap/SIP connection and another call comes in, I can hear
the Call Waiting Tone on the
I've been trying to get Telco MWI to work with asterisk 1.6. I'm
currently using the beta9 version. I haven't been able to get this new
feature working on any 1.6 beta release version.
I've turned on the full asterisk debug:
logger.conf:
full = notice,warning,error,debug,verbose
The only
I'm trying to get the Telco MWI recognition working in asterisk-1.6-beta7. I'm
told that it's supposed to work provided
my telco support FSK MWI signalling.
I have Verzon FIOS. I believe I have FSK MWI signaling as I can hear the
standard stutter tone when I pick up a live
handset in front
PROTECTED] wrote:
Two ways, use n priority or add 'g' iption in dial command.
2008/3/9, Jim Duda [EMAIL PROTECTED]:
How do I get a context to continue to execute commands after the caller
hangs up after a Dial command? I'm using the e option to the Dial
application. I though the e option would
I take back that statement about the h extension.
This works fine.
[sphinx]
exten = s,1,AGI(MisterHouse.agi,Sphinx Connect)
exten = s,2,Dial(CONSOLE/1)
exten = h,1,AGI(MisterHouse.agi,Sphinx Disconnect)
exten = h,2,Hangup
Thanks for the help!
Jim
Jim Duda wrote:
Thanks for the responses
How do I get a context to continue to execute commands after the caller
hangs up after a Dial command? I'm using the e option to the Dial
application. I though the e option would allow the context to
continue. This doesn't want to work for me.
I'm using asterisk-1.6.beta5
I never get to 3
Edwin,
I feel your pain. I struggled getting fax to work reliably with both
1.2 and 1.4 versions. Any combination I tried, usually caused a crash.
I recently upgraded to 1.6.beta4 and installed the app_fax from the
addons installation and it worked first time out of the box :-)
I've
I'm struggling to get my dialplan to work with a simple analog fax
machine.
I have TDM400B zaptel card with an FXO and FXS port. I have the FXO
port connected to the POTS machine and the FAX machine connected to the
FXS port.
The FAX machine itself works fine, I can FAX outgoing messages
I've upgraded to asterisk-1.6.0-beta2.
I'm trying to get the new Telco MWI detection function working. It
doesn't appear to be working.
I have this in zapata.conf
; PSTN connected here
;immediate=no
;busydetect=yes
;busycount=8
;musiconhold=default
mwimonitor=yes
;mwilevel=512
I use a TDM400 card to interface with my telco. I used asterisk voice
mail. However, if I'm on the telco line while another call comes in,
obviously it cannot go to asterisk voice mail but instead bounces to the
Telco voice mail.
Is there any means by which I can get asterisk detect the
Thanks Russell, that's what I'm looking for.
Any idea when this will become part an official asterisk release?
Jim
Russell Bryant wrote:
Brian J. Murrell wrote:
On Thu, 2007-12-20 at 16:29 -0500, Jim Duda wrote:
Is there any means by which I can get asterisk detect the Message
Waiting
I use Asterisk in my house. Each phone is a different extension. I
really like the ability to have multiple simultaneous calls in the
house. However, I do miss being able to be able to pick up a phone in a
different room. Currently, I have to either transfer the call or
transfer the call
Ever since upgrading from 1.2.X to 1.4.2, I'm having trouble with
voicemail. When played back, the messages start out okay, but after 10
seconds or so, the playback speed starts to increase and the voice
becomes illegible. It seems like some kind of audio timing problem.
Phone calls seem
The zttest program results in 99%.
Jim
Tzafrir Cohen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
On Tue, Mar 27, 2007 at 09:30:36PM -0400, Jim Duda wrote:
Lacy,
I don't have any zaptel cards installed. I do however have ztdummy
installed.
Is there some tweaks to ztdummy
if anything else is required, for
example,
udev configuration.
I use Fedora Core 5.
Jim
Travis Schafer [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Looks like output from the 'lsmod' command.
Lacy Moore - Aspendora [EMAIL PROTECTED] 3/27/2007 11:34 PM
On 3/27/07, Jim Duda
provider?
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
Many times the speed of an inbound voice call changes. It's similiar
to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible.
A speed change is the best way to describe it, seems like the voice
Many times the speed of an inbound voice call changes. It's similiar
to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible.
A speed change is the best way to describe it, seems like the voice
packets are being played out too fast.
Can anyone explain what might cause this?
Lacy,
I don't have any zaptel cards installed. I do however have ztdummy
installed.
Is there some tweaks to ztdummy which I might need?
Is there a special kernel setting which ztdummy requires?
Jim
Lacy Moore - Aspendora wrote:
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
Many times
!!
Jim
Lacy Moore - Aspendora wrote:
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
I don't have any zaptel cards installed. I do however have ztdummy
installed.
Hmm... Not sure. But this really sounds like ztdummy is not working
correctly. Hopefully someone else can jump in here. The only
this look normal?
Jim
Jim Duda wrote:
Lacy,
I'm using asterisk 1.4.2 and zaptel 1.4.1.
I read the READMEs again.
I believe I need to change my kernel RTC to 1000HZ.
Also, I didn't have enhanced_real_time clock enabled, as such,
ztdummy wasn't loading properly.
I have rebuilt and started
in message news:[EMAIL PROTECTED]
On 7 Feb 2007, at 03:59, Jim Duda wrote:
Thanks for the reply Lacy.
Yes, I know that I am using IAX2 and not SIP for my connection to teliax.
IAX2 is the preferred protocol for
connection to teliax. I have the firewall configured to prioritorize port
in message news:[EMAIL PROTECTED]
Jim,
I too am a Teliax user. Talk to their technical support. IAX2 is NOT
preferred. They'll tell you to use SIP.
Jim Duda wrote:
Thanks for the reply Lacy.
Yes, I know that I am using IAX2 and not SIP for my connection to teliax.
IAX2 is the preferred
I'm struggling to get my VOIP installation to be acceptable. I'm
looking for advice on what else I can look for.
My system:
o Teliax VOIP service, voip-ny1 proxy
o RCN Cable Internet Service (3Mbps download, 500kbps upload, 6ms
average jitter)
o 3.2 GHZ P4 Server (runs asterisk, firewall,
and 500
Kbit upload speeds.
Jim
Lacy Moore wrote:
Jim Duda wrote:
I've been on the shorewall firewall and confirmed that I have the
firewall configured properly for VOIP QOS.
What exactly have you done here? You do know that you are apparently
using IAX2 and not SIP. Those are not the same
-of-date), however, USE_RTC is NOT to be found in any source file. I
could
add it, however, doesn't appear to be used anywhere.
Jim
Tony Mountifield [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED]
wrote:
I'm running asterisk
PROTECTED], Jim Duda [EMAIL PROTECTED]
wrote:
Check ztdummy.h and ztdummy.c and make doubly sure USE_RTC is defined.
It sounds like you are using a version compiled without USE_RTC on one
of the newer kernels that has fewer than 1000 jiffies per second.
I'm using zaptel from CVS
: (none)
Sticky Date: (none)
Sticky Options: (none)
Tony Mountifield [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED]
wrote:
Check ztdummy.h and ztdummy.c and make doubly sure USE_RTC is defined.
It sounds like
that ztdummy would have a non zero value if * was using it.
Jim
Tony Mountifield [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED]
wrote:
I did cvs update -A, which brought in new files.
make clean
make
make install
make config
/etc
I'm running asterisk on FC4. All works fine, including musiconhold.
I tried installing ztdummy as directed, since the documentation
indicates that ztdummy is required for good music quality.
However, installing ztdummy on FC4 causes moh to play very slow.
If I remove ztdummy, all works okay
I saw the posting concerning goiax offering free DIDs. I went ahead,
created an account, and got myself a DID.
Who is goiax, and how can they be doing this for free? It's nice, but
how can they offer that?
I have outbound calling working from asterisk, to 800 numbers.
I cannot seem to get
I knew about that one. I have Silence Suppression set to NO.
Jim
Tony Mountifield [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
In article [EMAIL PROTECTED],
Jim Duda [EMAIL PROTECTED] wrote:
-=-=-=-=-=-
-=-=-=-=-=-
I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000
=rfc2833
mailbox=100
disallow=all
allow=ulaw
allow=gsm
Can you recommend a method to which I can post the configuration from the
grandstream bt100 device?
Jim
Tony Mountifield [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED
I've attached my zip file. Thanks for the help.
Jim
Tony Mountifield [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED]
wrote:
Thanks for the assistance.
I'm running version 1.0.6.7 of the software, tftp updated a few weeks
Can anyone tell me if the CallerID information is automatically displayed on
the LCD screen of the 301?
Can asterisk manipulate the LCD screen for the purposes of displaying
callerid?
Is this a good quality phone? Or, is the 501 worth the added expense?
I believe the only real differences
I'm using 1
BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The
analog phones with the Sipura seem to work great. Voice quality is fine on
both ends on the Sipura. I'm using the Teliax service and I use the Ulaw
codec for all phones.
However, I'm
struggling with the
Can anyone tell me if the CallerID information is automatically displayed
on the LCD screen of the 301?
Can asterisk manipulate the LCD screen for the purposes of displaying
callerid?
Is this a good quality phone? Or, is the 501 worth the added expense?
I believe the only real differences
I use the Teliax service with the IAX2 protocol. I noticed 2 days ago
that I was not registered with the Teliax server. I used the iax2 show
registry command and found I was not registered with Teliax. I issued
a reload command in asterisk in order to connect again.
I went to the Teliax
I have these messages in my log file
Jun 12 23:33:57 NOTICE[1707]: Registration of 'jduda' rejected:
Registration Refused
Jun 12 23:34:47 NOTICE[1707]: Registration of 'jduda' rejected:
Registration Refused
Jun 13 00:19:00 NOTICE[1707]: Registration of 'jduda' rejected:
Registration Refused
Jun
I've been using
Asterisk for a few weeks now. I have a (1) BT100 phone and a Sipura-2000
for all my analog phones. All has worked rather flawlessly, until
today.
I was on the BT100
phone today. During my phone conversation, the BT100 disconnected and went
into a "click" mode. 2 "clicks"
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