[asterisk-users] Questions about fax detection
Hello, I want to offer SIP phone user a custom fax-to-email feature. Here is how I would describe this feature: - for every SIP phone,a custom email address is defined - when a SIP phone answers an incoming call (from a trunk or another SIP endpoint), it detects the call is coming from a fax machine and then : + it plays a pre-recorded audio file to the receiving user (You are now receiving a fax call, please check you email box) + while at the same time, the incoming channel is forwarded to an appropriate statement within Asterisk dialplan. - when an unanswered call is forwarded to a voicemail, the fax call is also detected and teated appropriately. 1. It is possible to play a pre-recorded audio file to the receiving user ? If positive, how can it be done ? 2. What is the exact purpose of sip.conf faxdetect setting in this case given the assumption faxdetect is set to yes in general section of sip.conf. I would say the following applies: faxdetect for the incoming channel has no influence at all. If faxdetect is set to yes or unset in the outgoing channel, then Asterisk will jump to fax extension. If faxdetect is set to no in the outgoing channel, then Asterisk will jump to fax extension. Do you agree ? 3. Using CLI, is there a way to read the faxdetect parameter value of a given SIP peer ? To me sip show peer foo doesn't (seem to) display this. 4. When I type fax show settings in, I've got (on an Asterisk 10 box): CLI fax show settings FAX For Asterisk Settings: ECM: Enabled Status Events: Off Minimum Bit Rate: 2400 Maximum Bit Rate: 14400 Modem Modulations Allowed: V17,V27,V29 FAX Technology Modules: Spandsp (Spandsp FAX Driver) Settings: CLI This last line troubles me a little (did I forget to configure spandsp ?). Should I care ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on VM with NO DAHDI hardware
2012/9/13 Shaun Ruffell sruff...@digium.com On Wed, Sep 12, 2012 at 11:52:40PM -0400, Mark Robinson wrote: I know that asterisk on virtual machine require a timing source. What would you suggest to use for timing? We will plan to use only SIP and IAX2. If you're on a newish kernel (something later than v2.6.22), can use app_confbridge instead of app_meetme, and do not need app_page (unless you can can join the beta / wait for Asterisk 11) you can simply use res_timining_timerfd. How do you specify this ? Passing options when compiling asterisk from source ? This will free you from any kernel module dependencies. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax and sending to mail
2012/9/13 bilal ghayyad bilmar...@yahoo.com Hi All; Is there a module (addon or already built in) that enable us to receive the fax on the telephony card and save it as image (or any other format) and sent it to email? have a look at receivefax (core show application ReceiveFAX) Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX detection in chan_dahdi 1.8.15
2012/8/31 Jeff LaCoursiere j...@sunfone.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Tuesday, August 28, 2012 3:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] FAX detection in chan_dahdi 1.8.15
Hi,
I recently replaced a site that was using 1.4.[mumble] with
hylafax/iaxmodem. They have an RBS T1 and were using about half of
their 50
DID numbers for fax to email. This all broke with the new system :(
The original chan_dahdi.conf had no mention of faxdetect, so I assume
it
was operating with whatever is the default. Off?
The new box originally had faxdetect=no, and I found that all my test
faxes failed with negotiation errors. When I finally tried
faxdetect=incoming test faxes from another machine running hylafax went
through fine, and I thought I was done.
The following week the customer reported that inbound faxes weren't
working.
When I looked at the log, I saw lots of these:
chan_dahdi.c: -- Redirecting DAHDI/24-1 to fax extension
Which in my FreePBX setup eventually goes to a no service message and
hangs up. I've never defined a fax extension and don't really know
what
that is about. Turns out that any fax machine that calls ends up
following
this path. If my other hylafax server calls, it follows the normal path
and
gets answered by my pool of iaxmodems. I don't really understand the
difference between the two types of calls, first of all.
So it seems from this experience and a recent thread on -users that
enabling
faxdetection in chan_dahdi sets up some additional buffering that at
least
in my case, in 1.8, seems to be required (without it all inbound faxes
fail
from my hylafax server with negotiation problems).
Unfortunately for me, this also seems to bypass normal DID handling and
sends calls to an undefined fax extension.
Can anyone shed some light?
Thanks,
j
On Tue, 2012-08-28 at 15:28 -0500, Danny Nicholas wrote:
IIRC correctly this is sort of like the s extension; you set up your
fax
handler in [default,fax,1]. Not sure how that is done in FreePBX.
I've managed to hack a fix for this... in chan_dahdi.c I found two
places where an async goto happens right after the message
Redirecting to fax extension. I simply commented it out in both
places.
While looking a the source I noticed that an attempt is made to create a
new channel variable FAXEXTEN with a comment save the DID number
before sending to the fax extension. I created a fax context in
extensions.conf and tried to use ${FAXEXTEN} to properly route my
inbound fax to email calls, but it turns out that it is just set to s,
which isn't useful at all. I spent some time trying to figure out where
in the channel structure the actual DID information exists, as properly
setting the FAXEXTEN variable is arguably the right fix for my
problem. But I'm just not familiar enough with the internals :( I'm
surprised others haven't had this issue...
Anyway commenting out the redirection did the trick for me.
Cheers,
j
Which Freepbx version are you using ?
Have you installed Freepbx's Fax Configuration module ?
Are you trying to attach fax-to-email service to incoming DIDs or to SIP
endpoints ?
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Re: [asterisk-users] Asterisk 11 - BLF on Custom devices
2012/8/21 isr...@gmail.com She's talking about asterisk 11 not asterisk 1.8.11 -Original Message- From: Phil Frost p...@macprofessionals.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 21 Aug 2012 15:19:31 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 11 - BLF on Custom devices -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users In the first place, are we certain Custom Device States includes RINGING, for instance ? To me INUSE or NOT_INUSE are acceptable values but RINGING, I'm not sure about that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trouble phoning via HUAWEI E169
Hi there,
I'm setting up a Asterisk network and I ran into some problems ... as you
might have guessed :)
The set up is like this:
Internal Communication in the company should be handled through softphones
over an asterisk server (works).
Outbound Communication should be handled through a HUAWEI E169 stick,
accessed by the chan_dongle project.
http://code.google.com/p/asterisk-chan-dongle/
When I call internal numbers, everything works fine, but when I try to
access outside, I get the following error:
== Using SIP RTP CoS mark 5
-- Executing [06766770031@internal:1] Answer(SIP/1001-0023, )
in new stack
-- Executing [06766770031@internal:2] Dial(SIP/1001-0023,
dongle0/r1/06766770031,20,r) in new stack
[Sep 13 11:33:31] WARNING[9835]: channel.c:5603 ast_request: No channel
type registered for 'dongle0'
[Sep 13 11:33:31] WARNING[9835]: app_dial.c:2218 dial_exec_full: Unable to
create channel of type 'dongle0' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [06766770031@internal:3] Hangup(SIP/1001-0023, )
in new stack
== Spawn extension (internal, 06766770031, 3) exited non-zero on
'SIP/1001-0023'
From googling my way around, I know this type of error normally relates to
a module not being loaded, but chan_dongle.so shows up when I type a
module show. I've been fiddling around with this for days and frankly I
don't really know where the problem could lie.
Below are excerpts from sip.conf and extensions.conf
SIP.conf
code
[general]
bindport = 5060
bindaddr = 192.168.61.25
tcpbindaddr = 192.168.61.25
tcpenable = yes
context = internal
transport = udp
disallow = all
allow = gsm
allow = ulaw
allow = alaw
[dongle0]
type=friend
context=internal
audio=/dev/ttyUSB1
data=/dev/ttyUSB2
imei=359638011610601
imsi=232018830482446
transport=udp
disallow = all
allow = gsm
allow = ulaw
allow = alaw
[1000]
type=friend
callerid = Benny 1000
secret=1000
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=1000
disallow=all
allow=gsm
allow=ulaw
allow=alaw
transport=udp
context=internal
[1001]
type=friend
callerid = Timme 1001
secret=1001
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=1001
disallow=all
allow=gsm
allow=ulaw
allow=alaw
/code
Extensions.conf
code
[internal]
; for 4-digit numbers, assume it's a SIP number in our own context
; call it
exten = _,1,Answer()
exten = _,n,Dial(SIP/${EXTEN},20,r)
exten = _,n,Hangup
; else
; for a number starting with zero try to call via Dongle
exten = _0X.,1,Answer()
exten = _0X.,n,Dial(dongle0/r1/${EXTEN},20,r)
exten = _0x.,n,Hangup
/code
Please shed some light on this .
Kind regards,
Benedikt
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Re: [asterisk-users] Trouble phoning via HUAWEI E169
2012/9/13 Benedikt Schöffmann benedikt.schoeffm...@gmail.com
Hi there,
I'm setting up a Asterisk network and I ran into some problems ... as you
might have guessed :)
The set up is like this:
Internal Communication in the company should be handled through softphones
over an asterisk server (works).
Outbound Communication should be handled through a HUAWEI E169 stick,
accessed by the chan_dongle project.
http://code.google.com/p/asterisk-chan-dongle/
When I call internal numbers, everything works fine, but when I try to
access outside, I get the following error:
== Using SIP RTP CoS mark 5
-- Executing [06766770031@internal:1] Answer(SIP/1001-0023, )
in new stack
-- Executing [06766770031@internal:2] Dial(SIP/1001-0023,
dongle0/r1/06766770031,20,r) in new stack
[Sep 13 11:33:31] WARNING[9835]: channel.c:5603 ast_request: No channel
type registered for 'dongle0'
[Sep 13 11:33:31] WARNING[9835]: app_dial.c:2218 dial_exec_full: Unable to
create channel of type 'dongle0' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [06766770031@internal:3] Hangup(SIP/1001-0023, )
in new stack
== Spawn extension (internal, 06766770031, 3) exited non-zero on
'SIP/1001-0023'
From googling my way around, I know this type of error normally relates to
a module not being loaded, but chan_dongle.so shows up when I type a
module show. I've been fiddling around with this for days and frankly I
don't really know where the problem could lie.
Below are excerpts from sip.conf and extensions.conf
SIP.conf
code
[general]
bindport = 5060
bindaddr = 192.168.61.25
tcpbindaddr = 192.168.61.25
tcpenable = yes
context = internal
transport = udp
disallow = all
allow = gsm
allow = ulaw
allow = alaw
[dongle0]
type=friend
context=internal
audio=/dev/ttyUSB1
data=/dev/ttyUSB2
imei=359638011610601
imsi=232018830482446
transport=udp
disallow = all
allow = gsm
allow = ulaw
allow = alaw
[1000]
type=friend
callerid = Benny 1000
secret=1000
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=1000
disallow=all
allow=gsm
allow=ulaw
allow=alaw
transport=udp
context=internal
[1001]
type=friend
callerid = Timme 1001
secret=1001
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=1001
disallow=all
allow=gsm
allow=ulaw
allow=alaw
/code
Extensions.conf
code
[internal]
; for 4-digit numbers, assume it's a SIP number in our own context
; call it
exten = _,1,Answer()
exten = _,n,Dial(SIP/${EXTEN},20,r)
exten = _,n,Hangup
; else
; for a number starting with zero try to call via Dongle
exten = _0X.,1,Answer()
exten = _0X.,n,Dial(dongle0/r1/${EXTEN},20,r)
exten = _0x.,n,Hangup
/code
Please shed some light on this .
Kind regards,
Benedikt
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I've never tried chan_dongle, but to me, the Dial statement is incorrect.
Maybe the following would be better:
exten = _0X.,n,Dial(dongle/dongle0/r1/${EXTEN},20,r)
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Re: [asterisk-users] Fax Detect on Demand
2012/8/16 Eric Wieling ewiel...@nyigc.com Using Asterisk 1.8.mumble. We would like to use fax detect on demand. Both chan_dahdi and chan_sip support setting fax detetect on a static basis, For curiosity's sake, could you make it work first using static settings ? but no way I've been able to find to enable/disable it on demand in the dialplan. In 1.4 we used the NVFaxDetect 3rd party app, but that no longer appears to be maintained. Does anyone have any suggestions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Streaming MeetMe Conference
Hi, I was wondering if anyone has any experience in streaming a MeetMe conference so that others might listen in to it? It would be nice if the audio format could be AAC, but at first any format will do. I did come across this: http://www.voip-info.org/wiki/index.php?page_id=991 Which looks interesting, but if anyone knows of a better way I would be interested! Thanks in advanced, Adam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi Answer a Call On ringing State.
Hi, I am also facing same issue. Is this resolved? Please reply. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Volume issue.
Hi experts. Recently I've insalled a PCI Khomp Pane on my server and inserted 4 chips to make call with it. The calls are good and no issue was noticed but I got reports that when someone call the chips the call volume is uncommonly low for both sides and they deploy some failures on the audio, only when the call comes from outside. When an extension at the same network makes a call that goes through the pane the calls are really good. The Khomp pane is an KGSM 40spx with 4 modules. All the 4 modules have chip installed and working. I got outbound routes to make cellphone calls goes out through the chips. If someone calls the chips, the call are redirected to our IVR. I'm using Asterisk 1.6.2.13, dahdi driver version 2.3.0.1. If any information make itself needed just tell me and I post. And sorry for the text, english is not my native language. -- Att.* *** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi Answer a Call On ringing State.
More info??? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Parveen Lamba Sent: 13 September 2012 01:16 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dahdi Answer a Call On ringing State. Hi, I am also facing same issue. Is this resolved? Please reply. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unplanned community service outage Sept 13th 2012
We are experiencing an outage with at least issues.asterisk.org and potentially other services. We don't have an expectation of how long these services will be down and are currently in the process of troubleshooting. Digium's Asterisk Development Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on VM with NO DAHDI hardware
On Thu, Sep 13, 2012 at 09:49:37AM +0200, Olivier wrote: 2012/9/13 Shaun Ruffell sruff...@digium.com On Wed, Sep 12, 2012 at 11:52:40PM -0400, Mark Robinson wrote: I know that asterisk on virtual machine require a timing source. What would you suggest to use for timing? We will plan to use only SIP and IAX2. If you're on a newish kernel (something later than v2.6.22), can use app_confbridge instead of app_meetme, and do not need app_page (unless you can can join the beta / wait for Asterisk 11) you can simply use res_timining_timerfd. How do you specify this ? Passing options when compiling asterisk from source ? It should build by default if your platform supports it. In menuconfig, check that the module is enabled: Resource Modules : [*] res_timing_timerfd Ensure that res_timing_timerfd.so is loaded in /etc/asterisk/modules.conf: [modules] noload = res_timing_dahdi.so noload = res_timing_pthread.so load = res_timing_timerfd.so And you can ensure you have a timing source in asterisk with timing test, which will also show which timing source you're using: *CLI timing test Attempting to test a timer with 50 ticks per second. Using the 'timerfd' timing module for this test. It has been 1000 milliseconds, and we got 50 timer ticks -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on VM with NO DAHDI hardware
On Thu, Sep 13, 2012 at 09:35:55AM -0500, Shaun Ruffell wrote: On Thu, Sep 13, 2012 at 09:49:37AM +0200, Olivier wrote: 2012/9/13 Shaun Ruffell sruff...@digium.com On Wed, Sep 12, 2012 at 11:52:40PM -0400, Mark Robinson wrote: I know that asterisk on virtual machine require a timing source. What would you suggest to use for timing? We will plan to use only SIP and IAX2. If you're on a newish kernel (something later than v2.6.22), can use app_confbridge instead of app_meetme, and do not need app_page (unless you can can join the beta / wait for Asterisk 11) you can simply use res_timining_timerfd. How do you specify this ? Passing options when compiling asterisk from source ? It should build by default if your platform supports it. In menuconfig, check that the module is enabled: Resource Modules : [*] res_timing_timerfd Ensure that res_timing_timerfd.so is loaded in /etc/asterisk/modules.conf: [modules] noload = res_timing_dahdi.so noload = res_timing_pthread.so load = res_timing_timerfd.so And you can ensure you have a timing source in asterisk with timing test, which will also show which timing source you're using: *CLI timing test Attempting to test a timer with 50 ticks per second. Using the 'timerfd' timing module for this test. It has been 1000 milliseconds, and we got 50 timer ticks You may also want to read over this wiki page: https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unplanned community service outage Sept 13th 2012
On 9/13/2012 9:20 AM, Asterisk Development Team wrote: We are experiencing an outage with at least issues.asterisk.org and potentially other services. We don't have an expectation of how long these services will be down and are currently in the process of troubleshooting. Digium's Asterisk Development Team The problems have been resolved, and all the affected services appear to have been restored. Please let us know if you see any further issues. Digium's Asterisk Development Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
After exchanging the cable with other equipment was running smoothly in my computer the problem persisted while the other team the cable that could be bad worked. With this test done, now suspect the problem I have it on the Digium card. I perform the test specified in: http://kb.digium.com/articles/Troubleshooting/How-do-I-run-a-pattern-loopback-test-on-a-Digium-E1-T1-card?retURL=%2Fpopup=false With this results call3 tools # ./patlooptest /dev/dahdi/1 -t 300 -v Using Timeout of 300 Seconds Going for it... Timeout achieved Ending Program Test ran 36499 loops of 2039 bytes/loop with 0 errors call3 tools # ./patlooptest /dev/dahdi/32 -t 300 -v Using Timeout of 300 Seconds Going for it... Timeout achieved Ending Program Test ran 36550 loops of 2039 bytes/loop with 0 errors This test would indicate that the board has no problem ... but where is the fault? On Wed, Sep 12, 2012 at 12:56 PM, equis software equissoftw...@gmail.comwrote: Thanks, I'll try changing cables. On Wed, Sep 12, 2012 at 12:21 PM, Shaun Ruffell sruff...@digium.comwrote: On Wed, Sep 12, 2012 at 12:19:41PM -0300, equis software wrote: I have a server with an asterisk ss7 link connected to a Siemens working well for over a year. A few days ago I started having problems with signaling. I found the following logs in / var / log / messages [1018427.030959] dahdi: Master changed to TE2/0/2 [1018427.120740] dahdi: Master changed to TE2/0/1 [1018427.789173] dahdi: Master changed to TE2/0/2 [1018427.884828] dahdi: Master changed to TE2/0/1 [1018431.209621] dahdi: Master changed to TE2/0/2 [1018431.300289] dahdi: Master changed to TE2/0/1 [1018434.763742] dahdi: Master changed to TE2/0/2 If I stop the asterisk and dahdi driver just let the messages continue to appear. Any ideas?? My guess is you have a cabling problem and that span TE2/0/1 is going into and out of alarm. Each time it goes into alarm the core of DAHDI will look for a new span to use as a timing source for asterisk. If TE2/0/1 is the preferred timing source, when it comes out of alarm, DAHDI will switch the timing back to it. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
On 9/13/2012 2:31 PM, equis software wrote: After exchanging the cable with other equipment was running smoothly in my computer the problem persisted while the other team the cable that could be bad worked. With this test done, now suspect the problem I have it on the Digium card. I perform the test specified in: http://kb.digium.com/articles/Troubleshooting/How-do-I-run-a-pattern-loopback-test-on-a-Digium-E1-T1-card?retURL=%2Fpopup=false With this results Have you done loopback testing with your telco to make sure your line is clean? I'd point fingers there before blaming the Digium card or a cable. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
I would still consider the cable. They are funny things and they make nice meters to check them without putting your communications at risk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Sharp Sent: Thursday, September 13, 2012 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message On 9/13/2012 2:31 PM, equis software wrote: After exchanging the cable with other equipment was running smoothly in my computer the problem persisted while the other team the cable that could be bad worked. With this test done, now suspect the problem I have it on the Digium card. I perform the test specified in: http://kb.digium.com/articles/Troubleshooting/How-do-I-run-a-pattern-l oopback-test-on-a-Digium-E1-T1-card?retURL=%2Fpopup=false With this results Have you done loopback testing with your telco to make sure your line is clean? I'd point fingers there before blaming the Digium card or a cable. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
My ultimate intention is to blame the board, but we have changed teh cable and the LTG in the central and the problem continues, while the same LTG with another server works fine. Is too strange!! On Thu, Sep 13, 2012 at 4:14 PM, Danny Nicholas da...@debsinc.com wrote: I would still consider the cable. They are funny things and they make nice meters to check them without putting your communications at risk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Sharp Sent: Thursday, September 13, 2012 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message On 9/13/2012 2:31 PM, equis software wrote: After exchanging the cable with other equipment was running smoothly in my computer the problem persisted while the other team the cable that could be bad worked. With this test done, now suspect the problem I have it on the Digium card. I perform the test specified in: http://kb.digium.com/articles/Troubleshooting/How-do-I-run-a-pattern-l oopback-test-on-a-Digium-E1-T1-card?retURL=%2Fpopup=false With this results Have you done loopback testing with your telco to make sure your line is clean? I'd point fingers there before blaming the Digium card or a cable. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX detection in chan_dahdi 1.8.15
On 09/13/2012 03:20 AM, Olivier wrote:
2012/8/31 Jeff LaCoursiere j...@sunfone.com mailto:j...@sunfone.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Tuesday, August 28, 2012 3:24 PM
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
Subject: [asterisk-users] FAX detection in chan_dahdi 1.8.15
Hi,
I recently replaced a site that was using 1.4.[mumble] with
hylafax/iaxmodem. They have an RBS T1 and were using about half
of their 50
DID numbers for fax to email. This all broke with the new
system :(
The original chan_dahdi.conf had no mention of faxdetect, so I
assume it
was operating with whatever is the default. Off?
The new box originally had faxdetect=no, and I found that all
my test
faxes failed with negotiation errors. When I finally tried
faxdetect=incoming test faxes from another machine running
hylafax went
through fine, and I thought I was done.
The following week the customer reported that inbound faxes
weren't working.
When I looked at the log, I saw lots of these:
chan_dahdi.c: -- Redirecting DAHDI/24-1 to fax extension
Which in my FreePBX setup eventually goes to a no service
message and
hangs up. I've never defined a fax extension and don't really
know what
that is about. Turns out that any fax machine that calls ends
up following
this path. If my other hylafax server calls, it follows the
normal path and
gets answered by my pool of iaxmodems. I don't really understand the
difference between the two types of calls, first of all.
So it seems from this experience and a recent thread on -users
that enabling
faxdetection in chan_dahdi sets up some additional buffering
that at least
in my case, in 1.8, seems to be required (without it all inbound
faxes fail
from my hylafax server with negotiation problems).
Unfortunately for me, this also seems to bypass normal DID
handling and
sends calls to an undefined fax extension.
Can anyone shed some light?
Thanks,
j
On Tue, 2012-08-28 at 15:28 -0500, Danny Nicholas wrote:
IIRC correctly this is sort of like the s extension; you set
up your
fax
handler in [default,fax,1]. Not sure how that is done in FreePBX.
I've managed to hack a fix for this... in chan_dahdi.c I found two
places where an async goto happens right after the message
Redirecting to fax extension. I simply commented it out in both
places.
While looking a the source I noticed that an attempt is made to
create a
new channel variable FAXEXTEN with a comment save the DID number
before sending to the fax extension. I created a fax context in
extensions.conf and tried to use ${FAXEXTEN} to properly route my
inbound fax to email calls, but it turns out that it is just set
to s,
which isn't useful at all. I spent some time trying to figure out
where
in the channel structure the actual DID information exists, as
properly
setting the FAXEXTEN variable is arguably the right fix for my
problem. But I'm just not familiar enough with the internals :( I'm
surprised others haven't had this issue...
Anyway commenting out the redirection did the trick for me.
Cheers,
j
Which Freepbx version are you using ?
Have you installed Freepbx's Fax Configuration module ?
Are you trying to attach fax-to-email service to incoming DIDs or to
SIP endpoints ?
FreePBX 2.10
Originally, just because I usually do an install and let FreePBX load
all available modules, it did load that module. Original advice was to
remove that module, which had no effect on my problem. It is removed
now anyway.
The intention is individual DIDs go to specific fax email addresses. I
saw your explanation for what you are trying to accomplish, and it
sounds interesting. Curious to see if you get it working.
Cheers,
j
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Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
On Thu, Sep 13, 2012 at 05:13:04PM -0300, equis software wrote: My ultimate intention is to blame the board, but we have changed teh cable and the LTG in the central and the problem continues, while the same LTG with another server works fine. Is too strange!! I don't quite understand this comment. Is it possible to reword it? Is the LTG the line interface card? Are both these test servers using the same provider drop? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trunk Config
Hello Everyone I have configure two sip turnk line on my trixbox now my client want If you dial turnk 1 DID and if that number is busy the call should automatically transfer to second line. Is there any way to do this. Please let me know -- Thanks Farooq Hussain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trunk Config
That's a telco feature, if I understand how you have worded this. You are saying that when an outside caller calls DID1 and it is busy, the call should roll to DID2? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Farooq Hussain Sent: Thursday, September 13, 2012 3:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Trunk Config Hello Everyone I have configure two sip turnk line on my trixbox now my client want If you dial turnk 1 DID and if that number is busy the call should automatically transfer to second line. Is there any way to do this. Please let me know -- Thanks Farooq Hussain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Detect on Demand
Yes. Trivial
Off the top of my head:
exten = 12125551212,1,Set(EMAIL=john@example.com)
exten = 12125551212,n,Answer
exten = 12125551212,n,Wait(4)
exten = 12125551212,n,Dial(SIP/1212)
exten = fax,1,ReceiveFax(/tmp/myfax.tiff)
exten = fax,n,AGI(yourfax2emailagi.php,${FAXEXTEN},${EMAIL})
The Answer and Wait are required for faxdetection to work.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, September 13, 2012 6:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Detect on Demand
2012/8/16 Eric Wieling ewiel...@nyigc.com
Using Asterisk 1.8.mumble. We would like to use fax detect on demand.
Both chan_dahdi and chan_sip support setting fax detetect on a static
basis,
For curiosity's sake, could you make it work first using static settings ?
but no way I've been able to find to enable/disable it on demand in the
dialplan.
In 1.4 we used the NVFaxDetect 3rd party app, but that no longer
appears to be maintained.
Does anyone have any suggestions?
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[asterisk-users] alsa channel
I have had a case where after a hangup on the Alsa channel asterisk still thinks the line or call is active. I have: rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 in my sip.conf file to help with this but it had no effect. How can I ensure a session HANGS up and is not stale Is there a way for the next incoming call to VERIFY that console/ALSA channel is still valid. I dont want to hangup a real connection - I want to give a busy tone for sure. But if the session is not valid I need it gone. How can I do that. I am using 1.4.43 Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trunk Config
Danny Nicholas, Thanks for you quick response... yeah exactly what i want. So, I need to ask my mnf to do this. Thanks Farooq Husain On Fri, Sep 14, 2012 at 1:56 AM, Danny Nicholas da...@debsinc.com wrote: That’s a “telco feature”, if I understand how you have worded this. You are saying that when an outside caller calls DID1 and it is busy, the call should roll to DID2? ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Farooq Hussain *Sent:* Thursday, September 13, 2012 3:52 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Trunk Config ** ** Hello Everyone I have configure two sip turnk line on my trixbox now my client want If you dial turnk 1 DID and if that number is busy the call should automatically transfer to second line. Is there any way to do this. Please let me know -- Thanks Farooq Hussain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Farooq Hussain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.16.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.16.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.16.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR (Closes issue ASTERISK-20132. Reported by Zubair Ashraf of IBM X-Force Research) * --- AST-2012-013: Resolve ACL rules being ignored during calls by some IAX2 peers (Closes issue ASTERISK-20186. Reported by Alan Frisch) * --- Handle extremely out of order RFC 2833 DTMF (Closes issue ASTERISK-18404. Reported by Stephane Chazelas) * --- Resolve severe memory leak in CEL logging modules. (Closes issue AST-916. Reported by Thomas Arimont) * --- Only re-create an SRTP session when needed; respond with correct crypto policy (Issue ASTERISK-20194. Reported by Nicolo Mazzon) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.16.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10.8.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 10.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR (Closes issue ASTERISK-20132. Reported by Zubair Ashraf of IBM X-Force Research) * --- AST-2012-013: Resolve ACL rules being ignored during calls by some IAX2 peers (Closes issue ASTERISK-20186. Reported by Alan Frisch) * --- Handle extremely out of order RFC 2833 DTMF (Closes issue ASTERISK-18404. Reported by Stephane Chazelas) * --- Resolve severe memory leak in CEL logging modules. (Closes issue AST-916. Reported by Thomas Arimont) * --- Only re-create an SRTP session when needed (Issue ASTERISK-20194. Reported by Nicolo Mazzon) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.8.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trunk Config
I believe the telco term is hunt group. You tell your mnf to put both numbers into the hunt group and it rolls to two when one is busy. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Farooq Hussain Sent: Thursday, September 13, 2012 4:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk Config Danny Nicholas, Thanks for you quick response... yeah exactly what i want. So, I need to ask my mnf to do this. Thanks Farooq Husain On Fri, Sep 14, 2012 at 1:56 AM, Danny Nicholas da...@debsinc.com wrote: That's a telco feature, if I understand how you have worded this. You are saying that when an outside caller calls DID1 and it is busy, the call should roll to DID2? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Farooq Hussain Sent: Thursday, September 13, 2012 3:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Trunk Config Hello Everyone I have configure two sip turnk line on my trixbox now my client want If you dial turnk 1 DID and if that number is busy the call should automatically transfer to second line. Is there any way to do this. Please let me know -- Thanks Farooq Hussain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Farooq Hussain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own
On 13/09/12 00:47, Vladimir Mikhelson wrote: On 9/12/2012 5:33 PM, Sebastian Arcus wrote: On 10/08/12 18:38, Chad Wallace wrote: On Tue, 31 Jul 2012 09:44:26 +0100 Sebastian Arcuss...@open-t.co.uk wrote: I have two setups with SIP hardware phones as extensions and POTS lines as trunks. Internal SIP to SIP calls are crystal clear, but all calls bridged to POTS have a significant amount of static noise. The problem is that if I plug a POTS phone directly into the line, there is almost no static noise - the line is clean. It's like Asterisk (or the hardware) amplifies the static noise. What I've tried so far: 1. Connect Asterisk with a short cable directly into the master phone socket, where it enters the building. 2. One of the lines carries ADSL - so I double filtered it. 3. Tried three different phone sets (one Grandstream, two Cisco models). 4. Tried an OpenVox A400P PCI card and a Sangoma U100 USB adapter as analogue-to-digital interfaces. Have you run fxotune? I remember doing that when we had analog lines. You'd have to look up how--maybe just in the fxotune man page. Thanks for replying Chad - and sorry for the delay in my reply. I should have mentioned that I ran fxotune and made no difference. I also checked the interrupts, and even changed motherboard, and tried a USB analog adapter (Sangoma U100) instead of the current OpenVox PCI adapter. None solved the problem. I have given in and asked the client to order ISDN lines I'm afraid. Sebastian Sebastian, I understand it is too late for your client, but for the sake of consistency. I am experiencing a similar issue with Digium TDM410 on FXS lines. In my case the static noise is always present on analog extensions provided by TDM410. I had a ticket opened with Digium, and they admitted the following, but refused to make the findings public: 1. The static noise is produced by Digium analog equipment on certain motherboards. I specifically tried various Dell Dimension Pentium III machines with several power supplies. They all consistently started producing the noise the moment DAHDI drivers loaded, even before Asterisk was loaded. When I tried Dell Dimension Pentium IV machine the noise was not there. 2. On a machine which produced noise as described in #1 switching to a non-Digium TDM card fixed the noise problem. FXS daughter card stayed the same, just the base card was swapped. I do understand your situation involved FXO, and switching to OpenVox or Sangoma did not help. But I feel the root cause well may be the same. Regards, Vladimir Thanks for sharing Vladimir. It is interesting that somebody else is experiencing at least similar symptoms as I am. The hardware is generating the static noise (or at least it amplifies it greatly) - but I just couldn't find any way to reduce it sufficiently. After a chat with somebody working for a telecom company which installs proprietary PBX's - I reach the conclusion that it is not worth the effort. He said that whenever they encounter noise on analog lines, they don't tend to waste time tuning and adjusting things. They just recommend switching to ISDN - so I figured it's the only reasonable thing left to do. Of course, this won't make up of the many, many unchargeable hours spent troubleshooting this :-) Also, I'm afraid it isn't much of a solution for your case, if you are using FXS. Good luck, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble phoning via HUAWEI E169
On 13/09/12 11:16, Olivier wrote:
2012/9/13 Benedikt Schöffmann benedikt.schoeffm...@gmail.com
mailto:benedikt.schoeffm...@gmail.com
Hi there,
I'm setting up a Asterisk network and I ran into some problems ...
as you might have guessed :)
The set up is like this:
Internal Communication in the company should be handled through
softphones over an asterisk server (works).
Outbound Communication should be handled through a HUAWEI E169
stick, accessed by the chan_dongle project.
http://code.google.com/p/asterisk-chan-dongle/
When I call internal numbers, everything works fine, but when I try
to access outside, I get the following error:
== Using SIP RTP CoS mark 5
-- Executing [06766770031@internal:1]
Answer(SIP/1001-0023, ) in new stack
-- Executing [06766770031@internal:2] Dial(SIP/1001-0023,
dongle0/r1/06766770031,20,r) in new stack
[Sep 13 11:33:31] WARNING[9835]: channel.c:5603 ast_request: No
channel type registered for 'dongle0'
[Sep 13 11:33:31] WARNING[9835]: app_dial.c:2218 dial_exec_full:
Unable to create channel of type 'dongle0' (cause 66 - Channel not
implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [06766770031@internal:3]
Hangup(SIP/1001-0023, ) in new stack
== Spawn extension (internal, 06766770031, 3) exited non-zero on
'SIP/1001-0023'
From googling my way around, I know this type of error normally
relates to a module not being loaded, but chan_dongle.so shows up
when I type a module show. I've been fiddling around with this for
days and frankly I don't really know where the problem could lie.
Below are excerpts from sip.conf and extensions.conf
SIP.conf
code
[general]
bindport = 5060
bindaddr = 192.168.61.25
tcpbindaddr = 192.168.61.25
tcpenable = yes
context = internal
transport = udp
disallow = all
allow = gsm
allow = ulaw
allow = alaw
[dongle0]
type=friend
context=internal
audio=/dev/ttyUSB1
data=/dev/ttyUSB2
imei=359638011610601
imsi=232018830482446
transport=udp
disallow = all
allow = gsm
allow = ulaw
allow = alaw
[1000]
type=friend
callerid = Benny 1000
secret=1000
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=1000
disallow=all
allow=gsm
allow=ulaw
allow=alaw
transport=udp
context=internal
[1001]
type=friend
callerid = Timme 1001
secret=1001
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=1001
disallow=all
allow=gsm
allow=ulaw
allow=alaw
/code
Extensions.conf
code
[internal]
; for 4-digit numbers, assume it's a SIP number in our own context
; call it
exten = _,1,Answer()
exten = _,n,Dial(SIP/${EXTEN},20,r)
exten = _,n,Hangup
; else
; for a number starting with zero try to call via Dongle
exten = _0X.,1,Answer()
exten = _0X.,n,Dial(dongle0/r1/${EXTEN},20,r)
exten = _0x.,n,Hangup
/code
Please shed some light on this .
Kind regards,
Benedikt
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I've never tried chan_dongle, but to me, the Dial statement is incorrect.
Maybe the following would be better:
exten = _0X.,n,Dial(dongle/dongle0/r1/${EXTEN},20,r)
Looking at the chan_dongle documentation, it looks like you need to have
a dongle.conf in /etc/asterisk. Do you have it and does it contain the
right stuff? It looks like some of the stuff you've added to the
sip.conf should really go in dongle.conf, at least according to this page:
http://wiki.e1550.mobi/doku.php?id=configuration
Actually, I'm not sure you should have any settings connected with the
dongle in sip.conf - as SIP and dongle are different channel types and
use different configuration files.
According to the examples on the same page, your Dial string should not
include the name of the device, but the channel type, more like:
exten = _0X.,n,Dial(Dongle/r1/${EXTEN},20,r)
Also, what do you get when you run in Asterisk CLI:
dongle show devices
That should give you idea if the dongle is setup correctly in dongle.conf.
Hope the above helps,
Sebastian
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[asterisk-users] Voice Mail message should transfer to email address
Hello Everyone, Just caught with another problem... When i got a voice mail in one of my account how would i email that voice message to email address. Any one have any idea... -- Thanks Farooq Hussain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice Mail message should transfer to email address
On Fri, 14 Sep 2012, Farooq Hussain wrote: When i got a voice mail in one of my account how would i email that voice message to email address. Any one have any idea... I have 3 ideas. 1) Read the 'manual' -- google 'atfot.pdf' 2) Google 'asterisk voicemail email' 3) Read voicemail.conf -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own
On 9/13/2012 5:24 PM, Sebastian Arcus wrote: On 13/09/12 00:47, Vladimir Mikhelson wrote: On 9/12/2012 5:33 PM, Sebastian Arcus wrote: On 10/08/12 18:38, Chad Wallace wrote: On Tue, 31 Jul 2012 09:44:26 +0100 Sebastian Arcuss...@open-t.co.uk wrote: I have two setups with SIP hardware phones as extensions and POTS lines as trunks. Internal SIP to SIP calls are crystal clear, but all calls bridged to POTS have a significant amount of static noise. The problem is that if I plug a POTS phone directly into the line, there is almost no static noise - the line is clean. It's like Asterisk (or the hardware) amplifies the static noise. What I've tried so far: 1. Connect Asterisk with a short cable directly into the master phone socket, where it enters the building. 2. One of the lines carries ADSL - so I double filtered it. 3. Tried three different phone sets (one Grandstream, two Cisco models). 4. Tried an OpenVox A400P PCI card and a Sangoma U100 USB adapter as analogue-to-digital interfaces. Have you run fxotune? I remember doing that when we had analog lines. You'd have to look up how--maybe just in the fxotune man page. Thanks for replying Chad - and sorry for the delay in my reply. I should have mentioned that I ran fxotune and made no difference. I also checked the interrupts, and even changed motherboard, and tried a USB analog adapter (Sangoma U100) instead of the current OpenVox PCI adapter. None solved the problem. I have given in and asked the client to order ISDN lines I'm afraid. Sebastian Sebastian, I understand it is too late for your client, but for the sake of consistency. I am experiencing a similar issue with Digium TDM410 on FXS lines. In my case the static noise is always present on analog extensions provided by TDM410. I had a ticket opened with Digium, and they admitted the following, but refused to make the findings public: 1. The static noise is produced by Digium analog equipment on certain motherboards. I specifically tried various Dell Dimension Pentium III machines with several power supplies. They all consistently started producing the noise the moment DAHDI drivers loaded, even before Asterisk was loaded. When I tried Dell Dimension Pentium IV machine the noise was not there. 2. On a machine which produced noise as described in #1 switching to a non-Digium TDM card fixed the noise problem. FXS daughter card stayed the same, just the base card was swapped. I do understand your situation involved FXO, and switching to OpenVox or Sangoma did not help. But I feel the root cause well may be the same. Regards, Vladimir Thanks for sharing Vladimir. It is interesting that somebody else is experiencing at least similar symptoms as I am. The hardware is generating the static noise (or at least it amplifies it greatly) - but I just couldn't find any way to reduce it sufficiently. After a chat with somebody working for a telecom company which installs proprietary PBX's - I reach the conclusion that it is not worth the effort. He said that whenever they encounter noise on analog lines, they don't tend to waste time tuning and adjusting things. They just recommend switching to ISDN - so I figured it's the only reasonable thing left to do. Of course, this won't make up of the many, many unchargeable hours spent troubleshooting this :-) Also, I'm afraid it isn't much of a solution for your case, if you are using FXS. Good luck, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sebastian, There is a major difference between a noisy analog line your TELCO friend was talking about and a computer component generating the noise on it its own under certain circumstances. Switching to ISDN will help with one leg. If another leg stays analog it will be noisy in my scenario. I think your problem may have been of the same nature since you mentioned connecting an analog phone to that line eliminated the noise. As far as wasted hours go, I am with you - 100%. Regards, Vladimir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on VM with NO DAHDI hardware
Thanks Shaun. Very usefully head-up. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems
Hi, Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) chan_dahdi.conf contains: usecallerid = yes cidsignalling=dtmf cidstart=polarity_in Signalling is fxsks. Log (calling number is 9811066XXX): [Sep 14 08:21:11] DEBUG[9337]: chan_dahdi.c:11895 do_monitor: Monitor doohicky got event Ring Begin on channel 1 [Sep 14 08:21:11] DEBUG[9337]: sig_analog.c:3621 analog_handle_init_event: channel (1) - signaling (5) - event (ANALOG_EVENT_RINGBEGIN) [Sep 14 08:21:11] DEBUG[9337]: chan_dahdi.c:11895 do_monitor: Monitor doohicky got event Ring/Answered on channel 1 [Sep 14 08:21:11] DEBUG[9337]: sig_analog.c:3621 analog_handle_init_event: channel (1) - signaling (5) - event (ANALOG_EVENT_RINGOFFHOOK) [Sep 14 08:21:11] DEBUG[9337]: dsp.c:471 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Sep 14 08:21:11] DEBUG[9337]: dsp.c:471 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Sep 14 08:21:11] DEBUG[9337]: dsp.c:1576 ast_dsp_set_busy_pattern: dsp busy pattern set to 0,0 [Sep 14 08:21:11] DEBUG[9315]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for DAHDI - 1 [Sep 14 08:21:11] DEBUG[9315]: devicestate.c:458 do_state_change: Changing state for DAHDI/1 - state 2 (In use) [Sep 14 08:21:11] DEBUG[9315]: devicestate.c:438 devstate_event: device 'DAHDI/1' state '2' [Sep 14 08:21:11] DEBUG[11186]: sig_analog.c:1769 __analog_ss_thread: __analog_ss_thread 1 -- Starting simple switch on 'DAHDI/1-1' [Sep 14 08:21:11] DEBUG[11186]: sig_analog.c:2392 __analog_ss_thread: Receiving DTMF cid on channel DAHDI/1-1 [Sep 14 08:21:11] DEBUG[9350]: app_queue.c:1487 handle_statechange: Device 'DAHDI/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Sep 14 08:21:12] DEBUG[11186]: sig_analog.c:1602 analog_handle_dtmf: Begin DTMF digit: 0x31 '1' on DAHDI/1-1 [Sep 14 08:21:12] DEBUG[11186]: chan_dahdi.c:2026 my_handle_dtmf: Begin DTMF digit: 0x31 '1' on DAHDI/1-1 [Sep 14 08:21:12] DEBUG[11186]: dsp.c:1424 ast_dsp_process: DTMF Detected - Reset busydetector [Sep 14 08:21:12] DEBUG[11186]: sig_analog.c:1602 analog_handle_dtmf: End DTMF digit: 0x31 '1' on DAHDI/1-1 [Sep 14 08:21:12] DEBUG[11186]: chan_dahdi.c:2026 my_handle_dtmf: End DTMF digit: 0x31 '1' on DAHDI/1-1 [Sep 14 08:21:12] DEBUG[11186]: sig_analog.c:2426 __analog_ss_thread: CID got digit '1' [Sep 14 08:21:12] DEBUG[11186]: sig_analog.c:1602 analog_handle_dtmf: Begin DTMF digit: 0x36 '6' on DAHDI/1-1 [Sep 14 08:21:12] DEBUG[11186]: chan_dahdi.c:2026 my_handle_dtmf: Begin DTMF digit: 0x36 '6' on DAHDI/1-1 [Sep 14 08:21:12] DEBUG[11186]: dsp.c:1424 ast_dsp_process: DTMF Detected - Reset busydetector [Sep 14 08:21:12] DEBUG[11186]: sig_analog.c:1602 analog_handle_dtmf: End DTMF digit: 0x36 '6' on DAHDI/1-1 [Sep 14 08:21:12] DEBUG[11186]: chan_dahdi.c:2026 my_handle_dtmf: End DTMF digit: 0x36 '6' on DAHDI/1-1 [Sep 14 08:21:12] DEBUG[11186]: sig_analog.c:2426 __analog_ss_thread: CID got digit '6' [Sep 14 08:21:13] DEBUG[11186]: sig_analog.c:3509 analog_exception: analog_exception 1 [Sep 14 08:21:13] DEBUG[11186]: sig_analog.c:3603 analog_exception: Exception on 16, channel 1 [Sep 14 08:21:13] DEBUG[11186]: sig_analog.c:2660 __analog_handle_event: __analog_handle_event 1 [Sep 14 08:21:13] DEBUG[11186]: sig_analog.c:2687 __analog_handle_event: Got event ANALOG_EVENT_RINGBEGIN(12) on channel 1 (index 0) [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:3509 analog_exception: analog_exception 1 [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:3603 analog_exception: Exception on 16, channel 1 [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:2660 __analog_handle_event: __analog_handle_event 1 [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:2687 __analog_handle_event: Got event ANALOG_EVENT_RINGOFFHOOK(2) on channel 1 (index 0) [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:3043 __analog_handle_event: Ring detected [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:2441 __analog_ss_thread: CID got string '16' [Sep 14 08:21:14] WARNING[11186]: callerid.c:243 callerid_get_dtmf: Couldn't detect start-character. CID parsing might be unreliable [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:2443 __analog_ss_thread: CID is '16', flags 0 [Sep 14 08:21:14] DEBUG[9315]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for DAHDI - 1 [Sep 14 08:21:14] DEBUG[11186]: pbx.c:3239 ast_str_retrieve_variable: Result of 'EXTEN' is 's' [Sep 14 08:21:14] DEBUG[11186]: pbx.c:4230 pbx_extension_helper: Launching 'NoOp' [Sep 14 08:21:14] DEBUG[9315]: devicestate.c:458
[asterisk-users] Need to record user voice while play background music
Hi All; I was wondering if anyone has any experience for recording user voice while play background music? *My test case is :* When user enter in IVRS he is listen message for record your voice dial some digit after that user listen some background music and he also able to sing a song. A recorded user voice with background music within one file. After that user is able to listen his recorded file. Please, suggest me possible way for achieve this task. Thanks in Advance. -- Best Regards, Rajni Vanza Consultant Technology --- Working On Linux,C/C++,VoIP Technology -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need to record user voice while play background music
On Fri, 14 Sep 2012, RAJNI VANZA wrote: I was wondering if anyone has any experience for recording user voice while play background music? What methods have you tried? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
