That was exactly what I was looking for.
Thanks a lot for that.
That implies special care is needed with rackmount chassis.
Thanks for the tip, anyway.
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Douglas Garstang wrote:
Yes, all the equipment was located at the same physical location. In
hindsight, we could have multi-homed our collocations. Why can't service
providers multi home their edge systems to accept incoming calls from
two physical locations? If a service provider did this,
On Tue, 7 Aug 2007, Olivier wrote:
So no proper logoff between logins, right ?
As I will apply free sitting in school environment, chances are phones would
then remain logged-in several hours or days between another user logs in.
My thoughts are focused on finding the right balance between
--- Mr Shunz [EMAIL PROTECTED] wrote:
Caller ID Scheme as
ETSI-FSK Prior to Ringing with DTAS...
Thank you Daniele.
That seems to work.
I tested it on analog phones without a display.
I had previously experimented with different schemes
because I needed some of our phones to correctly
Hi all,
Can some one tell me how actually the users.conf working with SIP.conf,
IAX.conf, ZAPATA.conf ?
How can I add the SIP user to this file and do I relate this user profile
(in the users.conf) with the sip.conf?
I am a bit confused with this:-(
Can some one help??
Stephen Bosch wrote:
PSTN service still sets the standard.
With infrastructure paid for under a gracious guaranteed-profit monopoly
by ratepayers, now being used as a weapon to stifle competition from
VoIP, cable, and other emerging technologies.
b.
--
This message has been scanned for
Steve Totaro wrote:
What if a train derails and slices through the main fiber connections.
OK, so you have XO, Global Crossing, Verizon, and UCN all for
redundancy. Well guess what? They are all most likely running over
those strands of fiber. You better have a VSAT connection too!
We will be connecting our Asterisk server to ISDN 30 and intend using
the Sangoma A101 card. The install location is in London (UK).
Sangoma card at Voipon
http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA
I would be grateful to hear if this is the
On 8/7/07, Vieri [EMAIL PROTECTED] wrote:
--- Mr Shunz [EMAIL PROTECTED] wrote:
Caller ID Scheme as
ETSI-FSK Prior to Ringing with DTAS...
Thank you Daniele.
That seems to work.
I tested it on analog phones without a display.
Yeah, we had them without display too ...
But, if i
test only. good luck!
james.zhu
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Gordon,
What you described is exactly Follow-me feature : users are always logged
and can be reached somewhere.
By the way, do you introduce special settings so that ringing tones are
different ?
Let me explain this :
If Alice dials its extension and PIN code using Bob's hardphones, Bob and
On Tue, 7 Aug 2007, Olivier wrote:
Gordon,
What you described is exactly Follow-me feature : users are always logged
and can be reached somewhere.
I've heard of some variants of this feature - that's the beauty (and
down-side!) of a programmable system - it's open to different people's
Hmm.. This is what I get:
[EMAIL PROTECTED] ~]# mysql -u root -p
Enter password:
Welcome to the MySQL monitor. Commands end with ; or \g.
Your MySQL connection id is 187143 to server version: 4.1.20
Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
mysql use asteriskcdrdb ;
2007/8/7, Gordon Henderson [EMAIL PROTECTED]:
On Tue, 7 Aug 2007, Olivier wrote:
Gordon,
What you described is exactly Follow-me feature : users are always
logged
and can be reached somewhere.
I've heard of some variants of this feature - that's the beauty (and
down-side!) of a
Have a look at your SIP phones' support for the Alert-Info header (and
Asterisk's support for it, come to that).
I know some hardphone (eg Thomson ST2030) can set ring-tone
according Caller's presence inside phone's directory.
In this case, Asterisk would have
I am writing a cron script to check if certain
extensions are online and if they aren't then Asterisk
creates a couple of .call files to notify another set
of extensions or external numbers.
It works fine except for logging information.
What I'm doing in the script is setting a fake
caller ID
On Tue, 7 Aug 2007, Olivier wrote:
2007/8/7, Gordon Henderson [EMAIL PROTECTED]:
On Tue, 7 Aug 2007, Olivier wrote:
Gordon,
What you described is exactly Follow-me feature : users are always
logged
and can be reached somewhere.
I've heard of some variants of this feature - that's the
That's exactly what I was after.
Thanks
Maybe a bit of SIP MESSAGE and it would be perfect.
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Am Montag, den 06.08.2007, 18:09 +0200 schrieb gincantalupo:
Hi,
I'm trying to use a Detewe TA 33-clip but there is no rj11 connector on
it...only a TAE connector.
I'd like to create an adapter so I need to know which TAE pins to
connect to RJ 11 pins.
Is there anybody who knows where I
Hi,
first step is correct
Hmm.. This is what I get:
[EMAIL PROTECTED] ~]# mysql -u root -p
Enter password:
Welcome to the MySQL monitor. Commands end with ; or \g.
Your MySQL connection id is 187143 to server version: 4.1.20
Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
fateme fatah schrieb:
Please every that work with A102d say how about is it?Is it really difficult
to install card for me new in asterisk?
Best regards.
It is not more difficult to install than any other E1 card for Asterisk.
In fact in my opinion it's one of the easier to install.
Christian
Hi List,
Me setup for faxing is
Asterisk (TxFAX) = ATA = FAX Machine
And SIP setting is
Codec uLaw
dtmfmode inband
but I am facing a problem
when I send a FAX of one page from Asterisk to ATA+FAX Machine the FAX
Machine Print two pages (Enlarging the page) but shows it received one
page.
I need help on my zaptel.conf and Zapata.conf for a TE207P
I'd like Span 1 to receive a PRI from the phone company(US PRI).
I'd like Span 2 to interface with a Nortel Phone system as a PRI(acting as the
phone company)
Essentially my asterisk box is a man in the middle intercepting calls from
Hi all,
Anyone have an idea which version of spandsp, libunicall, libmfcr2,
libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the
latest asterisk 1.2?
Would that be the ones listed below?
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz
Brian Capouch wrote:
Stephen Bosch wrote:
PSTN service still sets the standard.
With infrastructure paid for under a gracious guaranteed-profit monopoly by
ratepayers,
At least in the US, this hasn't been done for many a year.
There is no LEGAL monopoly. There is, in many
when executing a NOOP(caller id ${CALLERIDNUM}) in the dialplan
I am getting odd caller id results from a SIP connection. The SIP
Connection is to
a nortel cs 1000.
*4145664222;phonecontext=+1
notice the extra stuff after the number
I am using asterisk 1.2.17
Is there a caller ID issue?
Ola Joao,
tem um modo do Asterisk fazer isso sim.
Entre em contato no meu GTALK por esse e-mail e eu te dou mais informações.
Abs!
Hi List,
The asterisk have one way to do it.
just put one script to discovery if this user is online or offline.
case is offline play one music. if not, call the
On Tue, Aug 07, 2007 at 04:01:54PM +0200, Patrick wrote:
Hi all,
Anyone have an idea which version of spandsp, libunicall, libmfcr2,
libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the
latest asterisk 1.2?
Would that be the ones listed below?
Can You please advice me free softphone which supports SIP registrations ?
Cheers,
Kate
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Hi Alessandro,
Thanks for that.. I'm pretty sure about the user. I used Webmin to
confirm the user configs, but I ran your commands anyway:
mysql use mysql ;
Reading table information for completion of table and column names
You can turn off this feature to get a quicker startup with
On Tue, Aug 07, 2007 at 09:01:56AM -0500, Jeremy Mann wrote:
I need help on my zaptel.conf and Zapata.conf for a TE207P
I'd like Span 1 to receive a PRI from the phone company(US PRI).
I'd like Span 2 to interface with a Nortel Phone system as a PRI(acting as
the phone company)
try using codec gsm or g729
On 8/7/07, Nasir Iqbal [EMAIL PROTECTED] wrote:
Hi List,
Me setup for faxing is
Asterisk (TxFAX) = ATA = FAX Machine
And SIP setting is
Codec uLaw
dtmfmode inband
but I am facing a problem
when I send a FAX of one page from Asterisk to ATA+FAX Machine
Douglas Garstang wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of SIP
Sent: Monday, August 06, 2007 8:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of Service
On Tue, 2007-08-07 at 09:01 -0500, Jeremy Mann wrote:
I’d like Span 1 to receive a PRI from the phone company(US PRI).
I’d like Span 2 to interface with a Nortel Phone system as a
PRI(acting as the phone company)
[snip]
Do I just need to make both PRI signaling? See below:
Your config
Mark Coccimiglio wrote:
Steve Totaro wrote:
What if a train derails and slices through the main fiber connections.
OK, so you have XO, Global Crossing, Verizon, and UCN all for
redundancy. Well guess what? They are all most likely running over
those strands of fiber. You better
On Tue, 2007-08-07 at 19:22 +0500, Rizwan Hisham wrote:
try using codec gsm or g729
No, please don't. I'll be the first do admit I don't know much about
faxing, but I *do* know that you don't want to try to send faxes over a
highly-compressed codec such as gsm or g.729. It will probably only
Hi,
I've compiled rxfax and txfax on my Asterisk box.
I've made a small extensions.conf for test so that when I call a number
with my Idefisk softphone I activate txfax command.
My goal is to try to send a fax using an analog line first and then an
ISDN line but I do not know how to specify the
So would the timing be 0?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith
Sent: Tuesday, August 07, 2007 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TE207P Question
As an added note, you
Hi,
Where can I find relevant information concerning callto:// tags ?
Is it standardized or browser specific ?
How within your browser, can you specify the software and parameters to used
when clicking on such callto:// tags ?
I couldn't find much googling or reading Preferences tab in Firefox.
Active Directory relies on Kerberos for authentication. Kerberos uses
tickets and does not centrally track presence.
You would either have to parse all the Domain Controller logs or use
some other method to determine when a user is logged in, such as setting
a flag with login/logout scripts.
Hi List,
I am using asterisk to test another asterisk/voip software, by
generating user agent on asterisk which can be used to place
calls to unit under test. I have also tests in which I place
calls over PSTN lines and receive them back on PSTN line and
verify the results.
Everything is fine,
On Tue, 2007-08-07 at 09:48 -0500, Jeremy Mann wrote:
So would the timing be 0?
Yes, that's correct if you're supplying timing to the device on the
other end.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Hi,
If I have [myprovider] section with context=something. When I do an
outgoing call by using Dial(SIP/myprovider/464646), does context=...
affect anything? As I understand it, it only affects incoming calls, but
I might be wrong.
Another thing, the setting of context=... on [default] section
Hi, try to login as asteriskcdruser to mysql
# mysql -u asteriskcdruser -p
Enter password: password
Welcome to the MySQL monitor. Commands end with ; or \g.
Your MySQL connection id is 12
--- Jerry Geis [EMAIL PROTECTED] wrote:
when executing a NOOP(caller id ${CALLERIDNUM})
I am using asterisk 1.2.17
I use CALLERID(num) or CALLERID(all) in 1.2+.
I don't know if that can help.
Very good. Sangoma cards are great. Get the a101d though. Nice wee review:
http://www.smithonvoip.com/new-voip-products/sangoma-a101d-single-port-t1e1-card/
Voipon are great guys too. We resell for them.
On 07/08/07, Rory Campbell-Lange [EMAIL PROTECTED] wrote:
We will be connecting our
On Tue, 2007-08-07 at 11:18 -0400, Filipe Brandenburger wrote:
If I have [myprovider] section with context=something. When I do an
outgoing call by using Dial(SIP/myprovider/464646), does context=...
affect anything? As I understand it, it only affects incoming calls, but
I might be wrong.
On Tue, Aug 07, 2007 at 08:16:35PM +0600, Kate Kretz wrote:
Can You please advice me free softphone which supports SIP registrations ?
twinkle? ekiga?
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL
That is wrong on so many levels.
You may want to take the time to install hylafax+iaxmodem, it offers
error correction and has many more features that offset the time
required to install...
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
Hisham
Sent: Tuesday,
sock=/tmp/mysql.sock
Is this path for socket correct ?
In some distro it is /var/lib/mysql/mysql.sock . Type locate mysql.sock in
shell . Also remove uncomment port=3306 if using socket to connect .
On 07/08/07, Alessandro Russo [EMAIL PROTECTED] wrote:
Hi, try to login as asteriskcdruser
When you make calls then context=xxx of the peer you are using ( your
extension ) will matter , the context=yyy line of your trunk wont matter .
If you dont specify a context= for a peer then it is considered to be in
[default] context .
On 07/08/07, Jared Smith [EMAIL PROTECTED] wrote:
On
Hi,
this is my cdr_mysql.conf
[global]
hostname=localhost
dbname=asterisk
table=cdr
password= mypassword
user=asteriskcdr
;port=3306
;sock=/tmp/mysql.sock
;userfield=1
as you can see, the last three are comment out, try to comment these rows,
your system could be able to retrieve the sock in
Either your configs are wrong or the card is faulty. You should have
crystal clear calls.
Thanks,
Steve
Manish Sapariya wrote:
Hi List,
I am using asterisk to test another asterisk/voip software, by
generating user agent on asterisk which can be used to place
calls to unit under test. I
Anthony Francis wrote:
Douglas Garstang wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of SIP
Sent: Monday, August 06, 2007 8:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Hi Gavin
Many thanks for the note. For what reason do you recommend the old a101
though?
Regards
Rory
On 07/08/07, Gavin Henry ([EMAIL PROTECTED]) wrote:
Very good. Sangoma cards are great. Get the a101d though. Nice wee review:
Christian Victor wrote:
fateme fatah schrieb:
Please every that work with A102d say how about is it?Is it really difficult
to install card for me new in asterisk?
Best regards.
It is not more difficult to install than any other E1 card for Asterisk.
In fact in my opinion it's one
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Anthony Francis
Sent: Tuesday, August 07, 2007 7:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of Service
Douglas
Brian Capouch wrote:
Stephen Bosch wrote:
PSTN service still sets the standard.
With infrastructure paid for under a gracious guaranteed-profit monopoly
by ratepayers,
In a regulated marketplace with legislated minimum service levels.
In Canada, most of the phone systems were
Jason,
What type of phones are you using? I originally started getting this
error when I got the Cisco 7961Gs (prior to dumping them and going with
all Polycoms). It turned out to be some setting in the XML provisioning
boot file (although I can't remember which one). Once I went to a
Douglas Garstang wrote:
So you've never gotten a dropped call or dead air on a PSTN call? Put it
in a little perspective.
I can count on one hand the number of outages of this kind that I've had
on PSTN in my lifetime.
Your mileage may vary.
-Stephen-
Mark Coccimiglio wrote:
Single point of failure should NEVER completely disable your company.
Yes outages happen and backhoe's cut fibre all the time. From within
this stuff can make one's life rather difficult, but from the outside it
should be almost unnoticed. When was the last time
Sangoma is working on it with me just for information here is what
appears in /var/log/messages when the PRI goes down:
Aug 7 06:11:48 EMSPBX kernel: wanpipe1: Critical: Echo Canceller Chip
Security Compromised: Disabling Driver!
Aug 7 06:11:48 EMSPBX kernel: wanpipe1: Please call Sangoma
On 8/6/07, Erik Anderson [EMAIL PROTECTED] wrote:
On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
Yeah you are sending the SABME's because you think you are the master,
they are not replaying with a UA because they think they are the master,
you should def be pri_cpe.
Tried it...no
On 8/7/07, Andrew Joakimsen [EMAIL PROTECTED] wrote:
In your wanpipe1.conf see if you have
TDMV_DCHAN = 0
Nope. I have it set to 24.
-erik
--
Erik Anderson
http://andersonfam.org
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Stephen Bosch wrote:
Of course not -- but how many hundreds of millions have been invested in
their infrastructure?
You missed the point. The standard formula I use is 5 days out or
more precisely 2% of gross revenues each year. For google its still a
kings ransom, but for a small
I've got 4 SIP phone lines with a call-limit of 2 for each. I've written a
handy macro to allow my users to dial a phone number and the macro will
figure out the next available line to use by first checking if the GROUP()
is over 2 and then checking to see if ChanIsAvail() as a backup, and if it
On Tue, 2007-08-07 at 11:13 -0700, Nicholas Blasgen wrote:
My question is this. Is it possible to tell Asterisk to execute part
of a macro as a block without allowing any other commands to be
processed during that time?
You'll want to check out the MacroExclusive() application. It does
Mark Coccimiglio wrote:
Stephen Bosch wrote:
Of course not -- but how many hundreds of millions have been invested in
their infrastructure?
You missed the point. The standard formula I use is 5 days out or
more precisely 2% of gross revenues each year. For google its still a
set your own mutex using astdb? It may just be atomic enough for you to
get by.
Nicholas Blasgen wrote:
I've got 4 SIP phone lines with a call-limit of 2 for each. I've
written a handy macro to allow my users to dial a phone number and the
macro will figure out the next available line to
I'm going to try it out, but I'm not very hopefull although it's exactly
what's needed. My macro contains a Dial() command and my concern is that
the dialplan isn't considered done untill Dial() returns. But I'm going to
try it. Will report back shortly.
On 8/7/07, Jared Smith [EMAIL
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stephen Bosch
Sent: Tuesday, August 07, 2007 10:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of Service
Brian Capouch
Steve Totaro wrote:
Setup those cell phones to use chan_mobile and you have a very nice
solution. Unless the phones are assigned to people who use them as
their own. You could possibly add some lines on a family plan $10/mo
extra w/T-Mobile and use those strictly as PSTN fialover lines.
Am Dienstag, den 07.08.2007, 16:51 +0200 schrieb Olivier:
Hi,
Where can I find relevant information concerning callto:// tags ?
Is it standardized or browser specific ?
How within your browser, can you specify the software and parameters
to used when clicking on such callto:// tags ?
I
Hello all. I am just getting back into Asterisk and I am setting up my
Linksys SPA3102. I have incoming calls working fine, as is the phone
plugged into the unit. My problem is I cannot get the SPA3102 to dial
a phone number automatically. I can call the extention of the PSTN and
I get a
Not specific to the SPA3102, but just normal outbound dialing is as follows:
exten = _1NXXNXX,1,Dial(trunk type/name/${EXTEN})
or if you want to require people to dial 9, then:
exten = _91NXXNXX,1,Dial(trunk type/name/${EXTEN})
or if you're like me and you're used to a cell phone and
On Aug 6, 2007, at 10:42 AM, Stephen Bosch wrote:
Eric ManxPower Wieling wrote:
Douglas Garstang wrote:
Let's assume for a moment that it's impossible. That does not
mean adding additional servers and additional networking
equipment does not add value, or is a worthless endeavour.
I
Tim Johnson wrote:
Hello all. I am just getting back into Asterisk and I am setting up my
Linksys SPA3102. I have incoming calls working fine, as is the phone
plugged into the unit. My problem is I cannot get the SPA3102 to dial
a phone number automatically. I can call the extention of
In Zapata.conf, if my PRI is NI-2 configured, do I still use
switchtype=national ?
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for
On 8/7/07, Jeremy Mann [EMAIL PROTECTED] wrote:
In Zapata.conf, if my PRI is NI-2 configured, do I still use
switchtype=national ?
Yup:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf#ISDNPRISwitchConfiguration
___
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Tim,
If the Asterisk stuff below doesn't fix it, try the docs at
http://www.jmgtechnology.com.au/spa_3000_guide.pdf
Ensure you enable VoIP to PSTN gateway mode and that PSTN Line is
registered with Asterisk. This is probably OK as you appear to get
dialtone back from the SPA. If you are
Replying to myself, I got this : http://en.wikipedia.org/wiki/URI_scheme
Anyway, I'm still wondering which is the best way to go, for standard and
usage compliance.
It seems that callto: was initialially used by netmeeting before being by
Skype software.
I could find a tab in XP Internet Option
Ollvier,
You could use the Firefox plug-in for Snap. It will auto detect
numbers on a webpage and make them dialable.
Cheers,
Mitchel
On 8/7/07, Olivier [EMAIL PROTECTED] wrote:
Replying to myself, I got this :
http://en.wikipedia.org/wiki/URI_scheme
Anyway, I'm still wondering which is the
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andres Paglayan
Sent: Tuesday, August 07, 2007 1:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of Service
On Aug 6,
Have both Sip and Callto (I'd use skype)
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent:
Asterisk Project Security Advisory - ASA-2007-019
++
| Product | Asterisk |
You can try using NetBIOS and see who is logged it at a machine, given
you know which machine a user loggs into, you also have to make sure
that NetBIOS is running on that machine/s. Or you could write a
logon/logoff script that will login/logout that user from Asterisk.
On 8/7/07, Olivier
Mitchel, he's not looking for a click to dial solution - he wants to
implement some form of click on his website so people can call him.
At the end of the day most people aren't going to have it configured
correctly etc and you should really use web page based softphone.
Regards,
Dean Collins
This is the postmaster at the list and I am notifying you that your
message failed.
On 8/7/07, zhu lizhong [EMAIL PROTECTED] wrote:
test only. good luck!
james.zhu
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The Asterisk development team has released Asterisk versions 1.2.24 and 1.4.10.
Version 1.2.24 is the final 1.2 release that contains normal bug fixes. The 1.2
branch will only be maintained with security fix releases from now until it is
completely deprecated.
Version 1.4.10 contains numerous
Is there a clean way to disable music on hold for a specific user sip
user?
I have seen one example that creates a class called [none] that points
to an empty directory, which creates log errors that are annoying (but
not harmful?)
___
--Bandwidth
Damon Estep wrote:
Is there a clean way to disable music on hold for a specific user sip
user?
I have seen one example that creates a class called [none] that points
to an empty directory, which creates log errors that are annoying (but
not harmful?)
How about using the same method
You could set a variable in the users sip.conf details like:
setvar=PlayMOH=NO
or
setvar=PlayMOH=NO
Then in your extensions.conf setup a GoToIf() which reads the variable
PlayMOH and either sets the m or the r in the dial command..
This should work fine and I know it will work in 1.4.x but
*** resent cause the first email never showed up on the list for me ***
Hi all,
Anyone have an idea which version of spandsp, libunicall, libmfcr2,
libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the
latest asterisk 1.2?
Would that be the ones listed below?
I received the original message at 7:01 AM today
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And I thought my sense of humour was poor
PaulH
On Tue, 2007-08-07 at 19:22 +0500, Rizwan Hisham wrote:
try using codec gsm or g729
On 8/7/07, Nasir Iqbal [EMAIL PROTECTED] wrote:
Hi List,
Me setup for faxing is
Asterisk (TxFAX) = ATA =
On Tue, 2007-08-07 at 10:33 -0400, Jared Smith wrote:
On Tue, 2007-08-07 at 19:22 +0500, Rizwan Hisham wrote:
try using codec gsm or g729
No, please don't. I'll be the first do admit I don't know much about
faxing, but I *do* know that you don't want to try to send faxes over a
Hi, I am still haveing the problem
its not a NAT issue i can hear the music on hold or even Echo is
working. only problem is VoiceMail and Playback doesn't work with
asterisk sound file.It just wait forever in Playing a file. I Did rtp
debug , i see only got Packet no send send packet, but for
Anyone in the bay area with strong unified messaging experience?
Respond off list at: [EMAIL PROTECTED]
Justin
Yahoo! oneSearch: Finally, mobile search
that gives answers, not web links.
Hi,
can anybody help me?.
Hi List,
Thanks for all replies.
IAX Modem + HylaFAX
T.38 Modem + HylaFAX
T.38 (using Callweaver)
all is ok
but please help me that
what is wrong with my setting?
I think there is speed difference between Asterisk and FAX Machine due
to improper Negotiation
sorry, I meant RFC 3856, sip presence, not sip regitration
On 8/7/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Aug 07, 2007 at 08:16:35PM +0600, Kate Kretz wrote:
Can You please advice me free softphone which supports SIP registrations
?
twinkle? ekiga?
--
Tzafrir
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