On Sun, Jun 07, 2009 at 10:15:42PM -0500, Tilghman Lesher wrote:
IAXmodem is a completely different ball of wax, and I think you would agree
that if the builtin FAX support in spandsp provided excellent support, there
never would have been a reason for IAXmodem to be developed.
Reminder:
You should look on the log for when the sox command is called, if the
invocation makes sense or not.
l.
2009/6/7 Joao Gomes Pereira gomespere...@startel.pt
Hello
I did as you told me, but the problem remains.
Im using Asterisk 1.2.x
and this is my config:
queues.conf
Hey Everyone,
i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM
300,320,360 Devices.
In the combination with asterisk and the patton, there are occuring some
strange behaviour, due to the calling and answering everything works
good, clear voice, great availability.
All the
Tzafrir Cohen wrote:
On Sun, Jun 07, 2009 at 10:15:42PM -0500, Tilghman Lesher wrote:
IAXmodem is a completely different ball of wax, and I think you would agree
that if the builtin FAX support in spandsp provided excellent support, there
never would have been a reason for IAXmodem to be
How do you transfer/move an active call to an external number via a
dialplan using either the app Dial or Transfer or some alternative,
then have Asterisk drop out of the connection. Basically, how can we have
asterisk dial another external number, transfer the caller, then disconnect
and no
Hi!
I have the following problem with Asterisk 1.4.23:
ATA w/ T.38 Asterisk ATA w/o T.38
INVITE
INVITE
---200OK--
---200OK--
A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems
to not time out, or at least have a very long time out.
We have a set up where we can dial two different peers, a primary and a
backup peer. If the first one dies completely, so that no error messages
(no ICMP unreachables or
Hi, i need to use a text to speech in my service.
What do think is the best free project?
Thanks
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I am logging into asterisk manager thru a Java program but not able to
login, if i use PHP I am able to login. I have attached my java code with
this mail. Can someone step me up to go ahead
--
Thank you with regards,
Gopal,
Echoclient.java
Description: Binary data
Hello,
I want to send Media outside Asterisk server, e.g. between peers.
In CLI I see:
. [Jun 8 13:13:58] VERBOSE[19112] logger.c: -- Native bridging
SIP/5060-b7dc5218 and SIP/prov12-09ad3888
. [Jun 8 13:13:58] DEBUG[19112] chan_sip.c: Strict routing enforced for
Benny Amorsen schrieb:
A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems
to not time out, or at least have a very long time out.
We have a set up where we can dial two different peers, a primary and a
backup peer. If the first one dies completely, so that no error
On Sunday 07 June 2009 23:29:30 Lee Howard wrote:
I
only want to clear up any misrepresentations about possible patent
infringements by spandsp to which you alluded.
My understanding wasn't that Steve violated any patents, but that he actively
avoided certain techniques to avoid conflicting
There is a timeout function in the Dial command. The folks who wrote the
command obviously felt that setting a programmatic limit on this would cause
somebody a problem. If you expect a reply from your SIP peer in 30 seconds,
just do Dial(SIP/peer,30) and the line will disconnect in 30 seconds.
Cepstral and Festival are both Free. In Cepstral, you pay a license fee
for the voice you use. In Festival, you tune the mechanical voice the way
you want. So if you want Truly free, choose Festival. If you want a
Human, Professional voice, Cepstral offers a reasonably priced product.
I'm considering implementing an Asterisk PBX for conferencing. Before I get
started, I wanted to make sure that it supports the features that I need.
I plan to use Asterisk as a conference bridge only. I want people to be able
to use my conference to listen live to lectures/etc, without having to
Lee Howard wrote:
Tilghman Lesher wrote:
On Sunday 07 June 2009 19:39:50 Lee Howard wrote:
Tilghman Lesher wrote:
What's the use case for the Digium
driver? Am I missing something by not using it?
While they accomplish the same goal, the
sean darcy wrote:
Tzafrir Cohen wrote:
On Sun, Jun 07, 2009 at 10:15:42PM -0500, Tilghman Lesher wrote:
IAXmodem is a completely different ball of wax, and I think you would agree
that if the builtin FAX support in spandsp provided excellent support, there
never would have been a
2009/6/8 Danny Nicholas da...@debsinc.com
Cepstral and Festival are both “Free”. In Cepstral, you pay a license
fee for the voice you use. In Festival, you tune the mechanical voice the
way you want. So if you want “Truly free”, choose Festival. If you want a
Human, “Professional” voice,
On Mon, Jun 8, 2009 at 9:02 AM, Danny Nicholasda...@debsinc.com wrote:
Cepstral and Festival are both “Free”. In Cepstral, you pay a license fee
for the voice you use. In Festival, you tune the mechanical voice the way
you want. So if you want “Truly free”, choose Festival. If you want a
Hi,
Is anybody picking up emails as attachments on an android phone like
the t-mobile G1?
I had this working a while ago but since I re-installed my asterisk
box to a newer build I am unable to open
the attachments, I just get told it can't handle the format?
I've been through and tried all wav,
On Sat, 2009-06-06 at 21:24 +0200, Philipp Kempgen wrote:
exten = s,n,Set(CALLERID(num)=${IF($[0${CALLERID(num)} 140] ?
${MAINSTUB}${CALLERID(num)}:${MAINNUMBER} )} )
^ ^
remove the trailing spaces
You'll also want
Danny Nicholas schrieb:
There is a timeout function in the Dial command. The folks who wrote the
command obviously felt that setting a programmatic limit on this would cause
somebody a problem. If you expect a reply from your SIP peer in 30 seconds,
just do Dial(SIP/peer,30) and the line
Jared Smith wrote:
On Sat, 2009-06-06 at 21:24 +0200, Philipp Kempgen wrote:
exten = s,n,Set(CALLERID(num)=${IF($[0${CALLERID(num)} 140] ?
${MAINSTUB}${CALLERID(num)}:${MAINNUMBER} )} )
^ ^
remove the trailing spaces
Fellow Asterisk Users,
I'm trying to marry SugarCRM and Asterisk..perhaps starting from elementary
features like a pop-up CRM record upon receipt of inbound call, for
starters.
Anybody who has successfully done this and beyond?
What integration tool are you using?
Which CRM are you using?
What
2009/6/8 Wai-Sun Chia waisun.c...@gmail.com:
Fellow Asterisk Users,
I'm trying to marry SugarCRM and Asterisk..perhaps starting from elementary
features like a pop-up CRM record upon receipt of inbound call, for
starters.
Anybody who has successfully done this and beyond?
What integration
Witch festival version are you talking about?
I need spanish(argentinian) voice...
On Mon, Jun 8, 2009 at 10:29 AM, David Backeberg dbackeb...@gmail.comwrote:
On Mon, Jun 8, 2009 at 9:02 AM, Danny Nicholasda...@debsinc.com wrote:
Cepstral and Festival are both “Free”. In Cepstral, you pay
On Mon, Jun 8, 2009 at 9:18 AM, Christopher
Stamperchristopherstam...@gmail.com wrote:
I'm considering implementing an Asterisk PBX for conferencing. Before I get
started, I wanted to make sure that it supports the features that I need.
I plan to use Asterisk as a conference bridge only. I
On Mon, Jun 8, 2009 at 2:06 PM, Klaus
Darilionklaus.mailingli...@pernau.at wrote:
Hi!
I have the following problem with Asterisk 1.4.23:
ATA w/ T.38 Asterisk ATA w/o T.38
INVITE
INVITE
On Mon, Jun 8, 2009 at 10:51 AM, equis softwareequissoftw...@gmail.com wrote:
Witch festival version are you talking about?
I need spanish(argentinian) voice...
I don't know whether any free programs do spanish TTS. I can tell you that
ATT Natural voices does do TTS en Espanol, and that was
On Mon, 8 Jun 2009, David Backeberg wrote:
On Mon, Jun 8, 2009 at 10:51 AM, equis softwareequissoftw...@gmail.com
wrote:
Witch festival version are you talking about?
I need spanish(argentinian) voice...
I don't know whether any free programs do spanish TTS. I can tell you that
ATT
Steve Underwood wrote:
I've had a kinda-working-but-not-production-ready SIPmodem for ages,
which does allow audio and T.38 from the same HylaFAX system, but I
haven't found the time to complete it.
Regards,
Steve
It's good to know that it's not been completely shelved, we are all
sean darcy escribió:
Jared Smith wrote:
On Sat, 2009-06-06 at 21:24 +0200, Philipp Kempgen wrote:
exten = s,n,Set(CALLERID(num)=${IF($[0${CALLERID(num)} 140] ?
${MAINSTUB}${CALLERID(num)}:${MAINNUMBER} )} )
^ ^
I had the same issue with my Windows Mobile phone for a couple of years. I
finally realized that if I had the phone use IMAP instead of POP3, I could
open the attachments. No clue why as I received lots of attachments on the
phone and they always worked. It was only * attachments that didn't
Atis Lezdins schrieb:
On Mon, Jun 8, 2009 at 2:06 PM, Klaus
Darilionklaus.mailingli...@pernau.at wrote:
Hi!
I have the following problem with Asterisk 1.4.23:
ATA w/ T.38 Asterisk ATA w/o T.38
INVITE
On Mon, Jun 8, 2009 at 7:00 PM, Klaus
Darilionklaus.mailingli...@pernau.at wrote:
Atis Lezdins schrieb:
On Mon, Jun 8, 2009 at 2:06 PM, Klaus
Darilionklaus.mailingli...@pernau.at wrote:
Hi!
I have the following problem with Asterisk 1.4.23:
ATA w/ T.38 Asterisk ATA
Maybe it's just me, but I get the impression that Grandstream is
quite uncooperative.
We (and others) have asked them multiple times to make the call-
pickup code (**) configurable but either they don't understand
the request or they're unwilling to do anything about it.
Hi,
From Asterisk 1.6.1 embedded doc, Dial app G option is :
G(context^exten^pri) - If the call is answered, transfer the calling
party to
the specified priority and the called party to the specified
priority+1.
Optionally, an extension, or extension and context may be
Klaus Darilion wrote:
Atis Lezdins schrieb:
On Mon, Jun 8, 2009 at 2:06 PM, Klaus
Darilionklaus.mailingli...@pernau.at wrote:
Hi!
I have the following problem with Asterisk 1.4.23:
ATA w/ T.38 Asterisk ATA w/o T.38
INVITE
On Mon, Jun 8, 2009 at 11:08 AM, Jeff LaCoursierej...@jeff.net wrote:
The quality of TTS these days is truly amazing. May I ask what kind of
cost was involved with ATT?
All of that was setup before I worked here. It's possible that at the
time ATT won against Cepstral for price, or I'm not
Decent product, but their support and development are horrible. I showed
them that their SIP over TCP implementation was broken and their reply was
use udp
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Once again you prove your wisdom. I'm going to look into the AMI think, but
this is a good working solution. My original code was copied from an early
daemon I wrote in PERL, thus the bad problems.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi,
Following advice in voip-info.org, I could successfully send text to a
remote SIP endpoint using sipsak and this command :
# sipsak -M -v -s sip:7...@192.168.100.123 sip%3a7...@192.168.100.123 -B
Lunch time
warning: ignoring -i option when in usrloc mode
timeout after 500 ms
timeout after
Ran after problem(problem was over weekend, and this was ran 2 days
after it started) sorry I forgot about
the -v, I was doing the command at home:
#dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.999% 99.996% 99.998% 99.999% 99.998% 99.999% 99.999% 99.999%
--- Results after 8
Just out of curiosity, how are you planning to use it? (Reading email,
etc?)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software
Sent: Monday, June 08, 2009 7:58 AM
To: Asterisk Users List
Subject: [asterisk-users]
Hi to all,
I recently upgraded a production machine to asterisk 1.4.25. It seems
quite stable but after ~5 days of normal operation it core dumped with
this result:
(gdb) bt
#0 0x00516402 in __kernel_vsyscall ()
#1 0x005b3d20 in raise () from /lib/libc.so.6
#2 0x005b5631 in abort () from
I need to imlplement an IVR service where customers call and put a telephone
number, then I reproduce the name and address.
On Mon, Jun 8, 2009 at 3:57 PM, Michelle Dupuis supp...@ocg.ca wrote:
Just out of curiosity, how are you planning to use it? (Reading email,
etc?)
Stefan Schmidt s...@sil.at writes:
What kind of client cant handle one packet per minute without getting a
higher load?
It isn't a client. It handles thousands of connected devices, so it'll
be handling perhaps 50 OPTIONS packets every second if I go the qualify
route.
What your are
Klaus Darilion klaus.mailingli...@pernau.at writes:
Asterisk does not forward the 488 back to the caller, but hangs up the
callee's call leg. Further, the caller's call leg will not be hung up.
Is somebody aware of this problem and a fix?
This should be fixed in 1.6.x. At least I had pretty
It works! :D
Thanks
CS
On Sun, Jun 7, 2009 at 8:57 PM, Philipp Kempgen
philipp.kemp...@amooma.dewrote:
César Sequeira schrieb:
I try to connect Qutecom in my Asterisk Server but without success.
What field I need to complete?
Username;
Password;
Realm (asterisk IP Address);
Benny Amorsen schrieb:
Stefan Schmidt s...@sil.at writes:
What kind of client cant handle one packet per minute without getting a
higher load?
It isn't a client. It handles thousands of connected devices, so it'll
be handling perhaps 50 OPTIONS packets every second if I go the qualify
Miguel Molina wrote:
I recently upgraded a production machine to asterisk 1.4.25. It seems
quite stable but after ~5 days of normal operation it core dumped with
this result:
(gdb) bt
#0 0x00516402 in __kernel_vsyscall ()
#1 0x005b3d20 in raise () from /lib/libc.so.6
#2 0x005b5631 in
Gopalakrishnan A.N schrieb:
I am logging into asterisk manager thru a Java program but not able to
login, if i use PHP I am able to login. I have attached my java code with
this mail. Can someone step me up to go ahead
What does the manager interface respond?
What does the CLI say?
I would recommend you to use Asterisk-Java library has support for manager,
agi, etc.
http://asterisk-java.org/
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: lunes, 08 de junio de
Matthew J. Roth escribió:
Miguel Molina wrote:
I recently upgraded a production machine to asterisk 1.4.25. It seems
quite stable but after ~5 days of normal operation it core dumped with
this result:
(gdb) bt
#0 0x00516402 in __kernel_vsyscall ()
#1 0x005b3d20 in raise () from
On Mon, Jun 8, 2009 at 10:54 AM, Marc Charbonneau
timebandit...@gmail.comwrote:
On Mon, Jun 8, 2009 at 9:18 AM, Christopher
Stamperchristopherstam...@gmail.com wrote:
I'm considering implementing an Asterisk PBX for conferencing. Before I
get
started, I wanted to make sure that it supports
On Mon, Jun 8, 2009 at 10:54 AM, Marc Charbonneau
timebandit...@gmail.comwrote:
On Mon, Jun 8, 2009 at 9:18 AM, Christopher
Stamperchristopherstam...@gmail.com wrote:
I'm considering implementing an Asterisk PBX for conferencing. Before I
get
started, I wanted to make sure that it supports
guys...any opinions on the below?
--- On Mon, 6/8/09, Jay Ray jonty...@yahoo.com wrote:
From: Jay Ray jonty...@yahoo.com
Subject: [asterisk-users] Achoring MEdia
To: asterisk-users@lists.digium.com
Date: Monday, June 8, 2009, 1:43 AM
I have 2 hosts that Asterisk is in between of...and
I have setup asterisk alsa.conf to read the null device for ALSA and
console/dsp input.
asound.conf is
pcm.nullpcm {
type null
}
alsa.conf has
input_device=plug:nullpcm
Then when I call into the Console/dsp I get very choppy audio.
I dont need and data from the microphone. I just want
Hi,
I am a newbie to Asterisk; need help understanding three-way
conferencing
call-transfer features implemented over standard extensions i.e. on a
TDM800P card (4 FXO + 4FXS)
In Asterisk I have observed that if an extension is already
participating in
an active call
What is the magic to compile chan_dahdi.so in asterisk 1.6.1.x? I'm
on centos 5.3.
Also, asterisk 1.4.25 cannot compile chan_dahdi as well while the
previous versions do. What changed or what am i missing?
There probably isn't magic. If you post the errors you got during the
compile we'll
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