Re: [asterisk-users] T.38 client for Linux?
2011/9/21 Ian Pilcher arequip...@gmail.com I am looking for a simple way to send occasional faxes via the FXO port on my SPA3102 -- without having to connect a fax modem to an ATA. In an ideal world, this would be some sort of softfax that runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with T.38. This is one of those things that I thought would be relatively straightforward, but a couple of hours of Googling has left my head spinning. I'm posting here in the hope that there is a (fairly) simple way to do this, and someone can point me in the right direction. Thanks! -- Ian Pilcher arequip...@gmail.com If you're going to shift my paradigm ... at least buy me dinner first. -- _ -- 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 Doesn't Zoiper include some T.38 features ? -- _ -- 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] T.38 client for Linux?
Yes, Zoip support T.38 faxing but It is only client application and you need FOIP gateway (asterisk) to transmit a fax to your FXO port Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Thu, Sep 22, 2011 at 3:20 AM, Olivier oza_4...@yahoo.fr wrote: 2011/9/21 Ian Pilcher arequip...@gmail.com I am looking for a simple way to send occasional faxes via the FXO port on my SPA3102 -- without having to connect a fax modem to an ATA. In an ideal world, this would be some sort of softfax that runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with T.38. This is one of those things that I thought would be relatively straightforward, but a couple of hours of Googling has left my head spinning. I'm posting here in the hope that there is a (fairly) simple way to do this, and someone can point me in the right direction. Thanks! -- Ian Pilcher arequip...@gmail.com If you're going to shift my paradigm ... at least buy me dinner first. -- _ -- 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 Doesn't Zoiper include some T.38 features ? -- _ -- 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] Add PinCode on my dialplan
Hi,
I tried Authenticate where pass codes are stored on the file pass.conf
and it works.
exten = _,1,Authenticate(/etc/asterisk/pass.conf)
Since I have my CDR, I want to have a report wherein I can check which
pass code did the call. How can I achieve it?
Using authenticate through file does not replace ACCOUNT_CODE field with
the pass code entered, it only show *ast_h323 *under the Account_Code field.
Regards,
Malvin
On 9/21/2011 1:01 PM, Sam Govind wrote:
See core show application autheTAB
If passwords are already the same as those of voicemail.conf go for
application VMAuthenticate() - DIA generates a dial-tone which I don't
think is suitable for dialling out from users(insiders)
-= Info about application 'Authenticate' =-
[Synopsis]
Authenticate a user
[Description]
This application asks the caller to enter a given password in order to
continue
dialplan execution.
If the password begins with the '/' character, it is interpreted as a
file
which contains a list of valid passwords, listed 1 password per line
in the
file.
When using a database key, the value associated with the key can be
anything.
Users have three attempts to authenticate before the channel is hung
up.
[Syntax]
Authenticate(password[,options[,maxdigits[,prompt]]])
[Arguments]
password
Password the user should know
options
a: Set the channels' account code to the password that is entered
d: Interpret the given path as database key, not a literal file
m: Interpret the given path as a file which contains a list of account
codes and password hashes delimited with ':', listed one per line
in the
file. When one of the passwords is matched, the channel will have its
account code set to the corresponding account code in the file.
r: Remove the database key upon successful entry (valid with 'd'
only)
maxdigits
maximum acceptable number of digits. Stops reading after maxdigits
have been entered (without requiring the user to press the '#' key).
Defaults to 0 - no limit - wait for the user press the '#' key.
prompt
Override the agent-pass prompt file.
[See Also]
VMAuthenticate(), DISA()
On Wed, Sep 21, 2011 at 9:47 AM, Malvin Rito
mr...@mail.altcladding.com.ph mailto:mr...@mail.altcladding.com.ph
wrote:
Thanks. ?If I want to use unique PIN for every user that dials out
how can I implement it using Authenticate app?
Regards,
Malvin
On 9/21/2011 12:42 PM, Sam Govind wrote:
DISA and DB based Auth could be an overkill.
Kyle showed the very simplistic dial plan if Dial-out pin is
common for the whole system.
See application
*Authenticate(password[,options[,maxdigits[,prompt]]] *and if
Voicemail PIN are required to be used use application
*MAuthenticate([mailbox][@context][,options] *
Regards,
- Sammy
On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton k...@mocker.org
mailto:k...@mocker.org wrote:
Something like this should work:
exten = _011.,1,Answer
exten = _011.,n,Wait(1)
exten = _011.,n,Read(password,enter-password,5)
exten = _011.,n,GotoIf($[${password} = 12345]?5:9)
exten = _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
exten = _011.,n,Dial(SIP/+${EXTEN:3}@outbound)
exten = _011.,n,Hangup
exten = _011.,n,Playback(invalid)
exten = _011.,n,Hangup
Could be cleaned up (the GotoIf isn't very descriptive about
where it's going), but it's a starting point.
On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:
Hi List,
I currently have a asterisk server running used for
dialing-out for IDD but I want to Put a pincode wherein only
users with the right pin code will be allowed to dial IDD.
Any sample dialplan you can suggest pls?
Thanks,
Malvin
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Re: [asterisk-users] Add PinCode on my dialplan
Dear Malvin,
I see Sam worked hard to post you the whole info about the application where
it clearly states the use of option a - Please change the configuration
line accordingly now and see if it works for you.
Best Regards,
Gohar
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Malvin Rito
Sent: Thursday, September 22, 2011 3:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Add PinCode on my dialplan
Hi,
I tried Authenticate where pass codes are stored on the file pass.conf and
it works.
exten = _,1,Authenticate(/etc/asterisk/pass.conf)
Since I have my CDR, I want to have a report wherein I can check which pass
code did the call. How can I achieve it?
Using authenticate through file does not replace ACCOUNT_CODE field with the
pass code entered, it only show ast_h323 under the Account_Code field.
Regards,
Malvin
On 9/21/2011 1:01 PM, Sam Govind wrote:
See core show application autheTAB
If passwords are already the same as those of voicemail.conf go for
application VMAuthenticate() - DIA generates a dial-tone which I don't think
is suitable for dialling out from users(insiders)
-= Info about application 'Authenticate' =-
[Synopsis]
Authenticate a user
[Description]
This application asks the caller to enter a given password in order to
continue
dialplan execution.
If the password begins with the '/' character, it is interpreted as a file
which contains a list of valid passwords, listed 1 password per line in the
file.
When using a database key, the value associated with the key can be
anything.
Users have three attempts to authenticate before the channel is hung
up.
[Syntax]
Authenticate(password[,options[,maxdigits[,prompt]]])
[Arguments]
password
Password the user should know
options
a: Set the channels' account code to the password that is entered
d: Interpret the given path as database key, not a literal file
m: Interpret the given path as a file which contains a list of account
codes and password hashes delimited with ':', listed one per line in the
file. When one of the passwords is matched, the channel will have its
account code set to the corresponding account code in the file.
r: Remove the database key upon successful entry (valid with 'd'
only)
maxdigits
maximum acceptable number of digits. Stops reading after maxdigits
have been entered (without requiring the user to press the '#' key).
Defaults to 0 - no limit - wait for the user press the '#' key.
prompt
Override the agent-pass prompt file.
[See Also]
VMAuthenticate(), DISA()
On Wed, Sep 21, 2011 at 9:47 AM, Malvin Rito mr...@mail.altcladding.com.ph
wrote:
Thanks. ?If I want to use unique PIN for every user that dials out how can I
implement it using Authenticate app?
Regards,
Malvin
On 9/21/2011 12:42 PM, Sam Govind wrote:
DISA and DB based Auth could be an overkill.
Kyle showed the very simplistic dial plan if Dial-out pin is common for the
whole system.
See application Authenticate(password[,options[,maxdigits[,prompt]]] and if
Voicemail PIN are required to be used use application
MAuthenticate([mailbox][@context][,options]
Regards,
- Sammy
On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton k...@mocker.org wrote:
Something like this should work:
exten = _011.,1,Answer
exten = _011.,n,Wait(1)
exten = _011.,n,Read(password,enter-password,5)
exten = _011.,n,GotoIf($[${password} = 12345]?5:9)
exten = _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
exten = _011.,n,Dial(SIP/+${EXTEN:3}@outbound)
exten = _011.,n,Hangup
exten = _011.,n,Playback(invalid)
exten = _011.,n,Hangup
Could be cleaned up (the GotoIf isn't very descriptive about where it's
going), but it's a starting point.
On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:
Hi List,
I currently have a asterisk server running used for dialing-out for IDD but
I want to Put a pincode wherein only users with the right pin code will be
allowed to dial IDD. Any sample dialplan you can suggest pls?
Thanks,
Malvin
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[asterisk-users] ForkCDR and asterisk 1.6.1
Hello,
In my 1.6.1.18-powered system, I've got the following dialplan (in
extensions.ael) :
Dial(SIP/foo,15);
if (${DIALSTATUS}=NOANSWER)
Dial(SIP/bar,15);
When SIP/baz dials peer SIP/foo which do not answer, I've got a single CDR
entry like this:
SIP/baz SIP/bar time_when_foo_started_to_ring
time_when_bar_ended_talking ANSWERED
How can I get two CDR entries :
- one for the unanswered call from SIP/baz to SIP/foo
- and one for the answered one from SIP/baz to SIP/bar ?
I've read about ForkCDR app but from reading its embedded doc, I couldn't
find if and how I could get the result I'm after.
What would you suggest ?
Regards
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Re: [asterisk-users] Add PinCode on my dialplan
Hi look at option a. This option put on accountcode field the name on the left
your password file.
Regards
Enviado desde mi iPad
El 22/09/2011, a las 5:49, Malvin Rito mr...@mail.altcladding.com.ph escribió:
Hi,
I tried Authenticate where pass codes are stored on the file pass.conf and it
works.
exten = _,1,Authenticate(/etc/asterisk/pass.conf)
Since I have my CDR, I want to have a report wherein I can check which pass
code did the call. How can I achieve it?
Using authenticate through file does not replace ACCOUNT_CODE field with the
pass code entered, it only show ast_h323 under the Account_Code field.
Regards,
Malvin
On 9/21/2011 1:01 PM, Sam Govind wrote:
See core show application autheTAB
If passwords are already the same as those of voicemail.conf go for
application VMAuthenticate() - DIA generates a dial-tone which I don't think
is suitable for dialling out from users(insiders)
-= Info about application 'Authenticate' =-
[Synopsis]
Authenticate a user
[Description]
This application asks the caller to enter a given password in order to
continue
dialplan execution.
If the password begins with the '/' character, it is interpreted as a file
which contains a list of valid passwords, listed 1 password per line in the
file.
When using a database key, the value associated with the key can be anything.
Users have three attempts to authenticate before the channel is hung
up.
[Syntax]
Authenticate(password[,options[,maxdigits[,prompt]]])
[Arguments]
password
Password the user should know
options
a: Set the channels' account code to the password that is entered
d: Interpret the given path as database key, not a literal file
m: Interpret the given path as a file which contains a list of account
codes and password hashes delimited with ':', listed one per line in the
file. When one of the passwords is matched, the channel will have its
account code set to the corresponding account code in the file.
r: Remove the database key upon successful entry (valid with 'd'
only)
maxdigits
maximum acceptable number of digits. Stops reading after maxdigits
have been entered (without requiring the user to press the '#' key).
Defaults to 0 - no limit - wait for the user press the '#' key.
prompt
Override the agent-pass prompt file.
[See Also]
VMAuthenticate(), DISA()
On Wed, Sep 21, 2011 at 9:47 AM, Malvin Rito mr...@mail.altcladding.com.ph
wrote:
Thanks. ?If I want to use unique PIN for every user that dials out how can I
implement it using Authenticate app?
Regards,
Malvin
On 9/21/2011 12:42 PM, Sam Govind wrote:
DISA and DB based Auth could be an overkill.
Kyle showed the very simplistic dial plan if Dial-out pin is common for the
whole system.
See application Authenticate(password[,options[,maxdigits[,prompt]]] and if
Voicemail PIN are required to be used use application
MAuthenticate([mailbox][@context][,options]
Regards,
- Sammy
On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton k...@mocker.org wrote:
Something like this should work:
exten = _011.,1,Answer
exten = _011.,n,Wait(1)
exten = _011.,n,Read(password,enter-password,5)
exten = _011.,n,GotoIf($[${password} = 12345]?5:9)
exten = _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
exten = _011.,n,Dial(SIP/+${EXTEN:3}@outbound)
exten = _011.,n,Hangup
exten = _011.,n,Playback(invalid)
exten = _011.,n,Hangup
Could be cleaned up (the GotoIf isn't very descriptive about where it's
going), but it's a starting point.
On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:
Hi List,
I currently have a asterisk server running used for dialing-out for IDD
but I want to Put a pincode wherein only users with the right pin code
will be allowed to dial IDD. Any sample dialplan you can suggest pls?
Thanks,
Malvin
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[asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)
Apologies for cross posting but some of us aren't on the other list
(vice/versa) and thought both groups would benefit.
For those familiar with the VoIP Abuse Project, no need to explain the
gist of this. I got tired of parsing through the alerts (lists) I
receive via email daily. They're long and sometimes I don't have the
time to post them all. So for now, posting VoIP Abuse addresses straight
to Twitter.
So, anyone trying to compromise a pbx, is now autoposted on an hourly
basis to Twitter. Still working on pulling, have about 4 machines linked
up now, will mop em up during the week.
http://twitter.com/#!/voipabuse
Now, you can concoct a quick script off of it, e.g.:
links -dump http://twitter.com/voipabuse;|awk '/attacker/{print
iptables -A INPUT -s $2 -j DROP| sort -u}'
Will get a quickie soon from my Acme's, nCites, etc. when I have time.
For those NOT familiar with it, please Google it as I don't feel like
typing anymore ;) (sorry)
--
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM
It takes 20 years to build a reputation and five minutes to
ruin it. If you think about that, you'll do things
differently. - Warren Buffett
42B0 5A53 6505 6638 44BB 3943 2BF7 D83F 210A 95AF
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x2BF7D83F210A95AF
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Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)
very cool!
On Thu, Sep 22, 2011 at 10:37 AM, J. Oquendo aster...@tormenting.netwrote:
Apologies for cross posting but some of us aren't on the other list
(vice/versa) and thought both groups would benefit.
For those familiar with the VoIP Abuse Project, no need to explain the
gist of this. I got tired of parsing through the alerts (lists) I
receive via email daily. They're long and sometimes I don't have the
time to post them all. So for now, posting VoIP Abuse addresses straight
to Twitter.
So, anyone trying to compromise a pbx, is now autoposted on an hourly
basis to Twitter. Still working on pulling, have about 4 machines linked
up now, will mop em up during the week.
http://twitter.com/#!/voipabuse
Now, you can concoct a quick script off of it, e.g.:
links -dump http://twitter.com/voipabuse;|awk '/attacker/{print
iptables -A INPUT -s $2 -j DROP| sort -u}'
Will get a quickie soon from my Acme's, nCites, etc. when I have time.
For those NOT familiar with it, please Google it as I don't feel like
typing anymore ;) (sorry)
--
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM
It takes 20 years to build a reputation and five minutes to
ruin it. If you think about that, you'll do things
differently. - Warren Buffett
42B0 5A53 6505 6638 44BB 3943 2BF7 D83F 210A 95AF
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x2BF7D83F210A95AF
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Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)
Sounds like a great idea.. Hopefully the page/account never gets hacked and
bad IP's published.. I could see a great hack of
127.0.0.1
192.168.0.0/16
10.0.0.0/8
getting up there somehow and next thing you know - BAM!
But I haven't RTFM - I'm guessing there is probably a white list that
supersedes the naughty list.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, September 22, 2011 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)
very cool!
On Thu, Sep 22, 2011 at 10:37 AM, J. Oquendo aster...@tormenting.net
wrote:
Apologies for cross posting but some of us aren't on the other list
(vice/versa) and thought both groups would benefit.
For those familiar with the VoIP Abuse Project, no need to explain the
gist of this. I got tired of parsing through the alerts (lists) I
receive via email daily. They're long and sometimes I don't have the
time to post them all. So for now, posting VoIP Abuse addresses straight
to Twitter.
So, anyone trying to compromise a pbx, is now autoposted on an hourly
basis to Twitter. Still working on pulling, have about 4 machines linked
up now, will mop em up during the week.
http://twitter.com/#!/voipabuse
Now, you can concoct a quick script off of it, e.g.:
links -dump http://twitter.com/voipabuse;|awk '/attacker/{print
iptables -A INPUT -s $2 -j DROP| sort -u}'
Will get a quickie soon from my Acme's, nCites, etc. when I have time.
For those NOT familiar with it, please Google it as I don't feel like
typing anymore ;) (sorry)
--
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM
It takes 20 years to build a reputation and five minutes to
ruin it. If you think about that, you'll do things
differently. - Warren Buffett
42B0 5A53 6505 6638 44BB 3943 2BF7 D83F 210A 95AF
http://pgp.mit.edu:11371/pks/lookup?op=get
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x2BF7D83F210A95AF
search=0x2BF7D83F210A95AF
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Re: [asterisk-users] Fax from FXS to PRI
On 09/20/2011 08:57 PM, Don Kelly wrote: This is a scary answer—you’re saying that what should be simple “TDM” FXS to PRI does not work? There is no TDM connection on a PCI or PCI-Express bus. Transferring data between two cards in the system either requires a direct connection between them (there are older, more expensive systems out there that use H.100 for this purpose) or the data has to go through the computer and its software. This means the data has to be packetized (even DAHDI's 1ms chunks are still packets), delivered to the host memory, an interrupt sent and acknowledged, and some code on the CPU then has to copy the data to another place in host memory for it to be sent out to the other card. Modern systems are of course fast enough to do this, but there are a lot of variables here, and it's possible for data to be occasionally delayed on its way between cards. With a voice call, this will likely not be noticed at all, but with a modem call or FAX call, it can be catastrophic. Are you suggesting this is an Asterisk problem or a Digium hardware problem? It's not a 'problem' with the hardware or the software; it's the nature of the beast. If you want 100% guaranteed TDM reliability, you have to use TDM connections, not packetized connections. Packetized connections can strive for 100% reliability, but most of them will never achieve it. For many people, with modern CPUs, current versions of DAHDI and Asterisk, and appropriate configuration (using the faxbuffers option in chan_dahdi.conf, for example), such a system can be setup to work very, very close to 100% of the time. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] Fax from FXS to PRI
On 09/20/2011 03:43 PM, Adam Moffett wrote: If I have a 4 port Digium FXS card and a single port PRI card on the same asterisk box, is it expected that I'd be able to plug a fax machine into the analog FXS port and have no problems sending or receiving faxes? Our connection to the Telco is on the PRI obviously. I don't recall the specific card models that we have, but I can check if it matters. Does the version of asterisk or Zaptel matter? You need to be using DAHDI, not Zaptel, as there have been a lot of improvements in DAHDI as a result of Digium spending a great deal of time testing the exact scenario you outlined (and related ones). In addition, you need to be using a recent enough version of Asterisk to have the 'faxbuffers' configuration option available in chan_dahdi.conf, so that channels transferring FAXes can be configured with extra buffering to provide an insurance policy against timing slips. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] T.38 client for Linux?
On 09/22/2011 02:20 AM, Olivier wrote: Doesn't Zoiper include some T.38 features ? It does, but the free version adds watermarks. It also doesn't appear to have been packaged for recent versions of Fedora (or any 64-bit version). -- Ian Pilcher arequip...@gmail.com If you're going to shift my paradigm ... at least buy me dinner first. -- _ -- 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 from FXS to PRI
September 22, 2011 11:20 AM Kevin P. Fleming wrote: For many people, with modern CPUs, current versions of DAHDI and Asterisk, and appropriate configuration (using the faxbuffers option in chan_dahdi.conf, for example), such a system can be setup to work very, very close to 100% of the time. Thanks for the explanation. I think that most of us can be happy with very, very close to 100%. --Don -- _ -- 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] T.38 client for Linux?
On 09/21/2011 09:07 PM, Nasir Iqbal wrote: You can use ictfax HTTP://www.ictfax.org web interface to send faxes, Ictfax is pure foip software based on t.38 as compared to hylafax I took a look at ictfax. It's massive overkill for what I need, and the setup looks far from simple. (And that's before I got to the part about changing the Apache user ID!) -- Ian Pilcher arequip...@gmail.com If you're going to shift my paradigm ... at least buy me dinner first. -- _ -- 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 from FXS to PRI
On 09/20/2011 09:57 PM, Jeff LaCoursiere wrote: Like most faxing issues, at it's root it is a timing problem IMO. Sangoma makes a special timing cable to link their cards so you can do exactly what you are asking to do. I've never purchased it, but last I looked into the issue, that is what they suggested. Digium sells such cables as well, but they cannot resolve this problem completely. They solve a related, but different, problem. There are really two issues at work here: the first is that the nominal 8kHz clock used on all TDM telephony boards is just that... 'nominal'. Every board has its own clock, and digital boards can be configured to derive a clock from the spans they are connected to. All of these clocks are built with varying degrees of accuracy and tolerance; PSTN-derived clocks will generally be of greater accuracy than anything you'd be able to afford putting into a PC, so that's why they are always the preferred option when they are available. If two ports in a system are using clocks that differ slightly in frequency, then over time they will 'drift'; one port will be producing (and consuming) data a slightly higher rate than the other port. The percentage of frequency difference will determine how often there will be a 'slip'; a 1% difference in an 8kHz clock is 80Hz. With that difference, one port will produce/consume data at a rate of 80 samples per second faster than the other port; at that rate, slips will occur fairly often. When slips occur, data will be lost, or will be replaced with silence... this is unavoidable. Timing cables are designed to reduce the drift/slippage problem; when a timing cable is connected between two cards, and at most one of those cards is connected to an external connection that provides a clock, then a single clock can be distributed between the cards. This produces a 'clock domain', and the ports will produce and consume data at the same rate. Timing cables do *not*, though, transfer data between the cards; even though a timing cable will result in the clocks on the cards being in sync, the data from one card still needs to be transferred to the other card *on time*. If the data arrives late, silence (or noise) will be sent in its place. This means that the entire system (hardware and software) must be able to reliably transfer the data between the two cards without it ever arriving late. Software running on a CPU, with other software operating at the same time, is subject to CPU scheduling delays and lots of other potential reasons why it might fail to deliver the data on time. The only way to guarantee that the data will not ever be late is to directly connect the TDM busses on the cards, or failing that, to run a single application on the host CPU with appropriate real-time scheduling so that it will *always* have access to CPU time slices when they are needed. Traditional PBXes avoided this problem (because users would not have been willing to accept it) by having direct TDM connectivity between all of their ports, and having the software only control the connections, not transfer the data. This, of course, produces significantly more expensive hardware, and this sort of connectivity isn't possible with packet networks. As the world has moved to lower-cost solutions and Voice-over-IP, some of these long-standing 'problems' have reappeared. For voice calls, most users are willing to accept them in return for lower costs, increased flexibility and other benefits... but modems and FAX machines can't tolerate these problems without help (which is where T.38 and V.150 enter the picture). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] T.38 client for Linux?
On 09/22/2011 11:29 AM, Ian Pilcher wrote: On 09/21/2011 09:07 PM, Nasir Iqbal wrote: You can use ictfax HTTP://www.ictfax.org web interface to send faxes, Ictfax is pure foip software based on t.38 as compared to hylafax I took a look at ictfax. It's massive overkill for what I need, and the setup looks far from simple. (And that's before I got to the part about changing the Apache user ID!) You can do this with Asterisk itself, starting with Asterisk 1.8. There are SendFAX and ReceiveFAX applications that support both traditional and T.38 FAX transmission, using either SpanDSP or Digium's Fax for Asterisk backends. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] bounty for ASTERISK-17474 streaming MusicOnHold bug
Hi all- This is my first post to this list so please don't flog me if this is not the appropriate place to post this. I've had an issue for over a year affecting MOH + DAHDI timers, I reported it at: https://issues.asterisk.org/jira/browse/ASTERISK-17474 Basically the MOH channel goes dead after a 'moh reload' command is issued (these are streaming sources that use mpg123) and will not start back up unless you do a full asterisk stop/start. The issue hasn't caught anyone's attention yet. So I thought I'd try to put together a bounty to spark some interest in fixing this issue. Is anyone else interested in putting up something to get this bug fixed? I'd be willing to put up $100 myself. If there's any support I would create the bounty on http://www.voip-info.org/wiki/view/Asterisk+bounty -- is that the correct way to do it? thanks, Luke -- _ -- 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] RTP stream when * and Xlite are both behind Nat devices.
Hi rcswebb, I had a problem like yours : Asterisk -NAT - internet - NAT - 3CX phone Without modifiyng Astrisk conf I could start a call from the client but without hearing a sound. The solution for me was to force Asterisk to modify the outgoing udp packet to insert it's public ip and not the private IP behind the NAT . So in your sip.conf I modified : [general] *externip=YouEternalIP* NAT=Yes Hope that'll help :-) Message: 11 Date: Wed, 21 Sep 2011 10:52:08 +0100 From: Richard Webb rcsw...@gmail.com Subject: [asterisk-users] RTP stream when * and Xlite are both behind Nat devices. To: asterisk-users@lists.digium.com Message-ID: cam39q6ol6ytw4skhq0vmrouxx_4mkr7p6zo4shbdh8hvqod...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk - Nat - Internet - Nat - Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp stream to be sent to the public facing address of the softphone? Cheers, -- _ -- 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] bounty for ASTERISK-17474 streaming MusicOnHold bug
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luke Hamburg Sent: Thursday, September 22, 2011 3:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] bounty for ASTERISK-17474 streaming MusicOnHold bug Hi all- This is my first post to this list so please don't flog me if this is not the appropriate place to post this. I've had an issue for over a year affecting MOH + DAHDI timers, I reported it at: https://issues.asterisk.org/jira/browse/ASTERISK-17474 Basically the MOH channel goes dead after a 'moh reload' command is issued (these are streaming sources that use mpg123) and will not start back up unless you do a full asterisk stop/start. The issue hasn't caught anyone's attention yet. So I thought I'd try to put together a bounty to spark some interest in fixing this issue. Is anyone else interested in putting up something to get this bug fixed? I'd be willing to put up $100 myself. If there's any support I would create the bounty on http://www.voip-info.org/wiki/view/Asterisk+bounty -- is that the correct way to do it? thanks, Luke My .02 1. This is a 1.8.X issue, so anything I say is purely conjecture, since I am a 1.4 hardliner 2. Don't know if moving to 10.x would help you, but since that is still considered beta, that's probably not an option anyhow. 3. My understanding is that bounties need to be posted on the asterisk-dev list. 4. With those caveats, have you tried this: Copy the load_module and unload_module routines from res_timing_pthread.c to res_timing_dahdi.c (you'll probably need some includes to make it compile right). IMO the final answer for you will be a hybrid res_timing module that combines these two. -- _ -- 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] Problem with multiple sip-peers against the same host
Dear list, We are switching to a new provider for SIP-trunks. We have 20 numbers, each defined as a separate SIP peer. With the old provider everything works. When switching to the new provider's account data, it only works as long as I only define one of the accounts. If multiple accounts are defined, I can only place outgoing calls on one of them, for the other(s) authentication fails, FORBIDDEN. It is almost like Asterisk is using just one of the defined passwords to authenticate all peers on that host. Any input is very appreciated. Regards David Björkevik, Engineer signature.asc Description: OpenPGP digital signature -- _ -- 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] Problem with multiple sip-peers against the same host
Can you please post: 1. Relevant sip.conf 2. sip debug when trying to make a call? On Thu, Sep 22, 2011 at 7:26 PM, David Björkevik da...@dynamore.se wrote: Dear list, We are switching to a new provider for SIP-trunks. We have 20 numbers, each defined as a separate SIP peer. With the old provider everything works. When switching to the new provider's account data, it only works as long as I only define one of the accounts. If multiple accounts are defined, I can only place outgoing calls on one of them, for the other(s) authentication fails, FORBIDDEN. It is almost like Asterisk is using just one of the defined passwords to authenticate all peers on that host. Any input is very appreciated. Regards David Björkevik, Engineer -- _ -- 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] dahdi_dummy required?
I'm running Asterisk 1.8.4.4; I'm new to asterisk, and have been reading the Asterisk Definitive Guide, and it mentions that dahdi_dummy should be used to provide an interface for Asterisk to get kernel timing. - espescially if using timing-dependant modules. I have a minor question: is dahdi_dummy necessary or useful anymore - espescially for users who don't have DAHDI hardware? I ask because I just checked out dahdi 2.5 from svn built (against the Linux kernel 3.0) I noticed that dahdi_dummy didn't seem to be built; when I poked around in the changelog, I saw: * README: README: Remove references to dahdi_dummy. Since dahdi_dummy is no longer required remove the references from README. (issue #17959) Reported by: glen201 Origin: http://svnview.digium.com/svn/dahdi?view=revrev=9308 So am I correct in assuming dahdi_dummy isn't needed/useful anymore? -- Troy Telford -- _ -- 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
