Hi:
I want to install mpg123-0.59r on my asterisk server.I downloaded it in
/usr/src then untared it and I typed these command :
#cd /usr/src/mpg123-0.59r
#make linux
after run make linux ,I saw 2 errors in terminal:
make CC=gcc LDFLAGS= \
OBJECTS='decode_i386.o
fateme fatah wrote:
Hi:
I want to install mpg123-0.59r on my asterisk server.I downloaded it
in /usr/src then untared it and I typed these command :
#cd /usr/src/mpg123-0.59r
#make linux
after run make linux ,I saw 2 errors in terminal:
make CC=gcc LDFLAGS= \
OBJECTS='decode_i386.o
On Sun, Jun 22, 2008 at 12:24:22AM -0700, fateme fatah wrote:
Hi:
I want to install mpg123-0.59r on my asterisk server.I downloaded it in
/usr/src then untared it and I typed these command :
BTW: why do you want to use mpg123?
--
Tzafrir Cohen
icq#16849755
On Sat, Jun 21, 2008 at 05:30:51AM -0700, Vieri wrote:
Hi,
I'm having trouble connecting two Asterisk boxes via a IAX2 friend trunk.
iax2 show peers on both boxes seem to show that all's fine (Status OK on
qualify=yes peer).
voip1 is an Asterisk 1.2.27 production server.
voip2 is an
Hi :
asterisk didn't send voice message to my mail([EMAIL PROTECTED]).My main
configured files are:
extensions.conf:
[from-pstn]
exten = 9711315,1,Dial(SIP/3000,30)
exten = 9711315,2,VoiceMail([EMAIL PROTECTED])
exten = 9711315,3,PlayBack(vm-goodbye)
exten = 9711315,4,HangUp()
sip.conf:
[3000]
Hi :
asterisk didn't send voice message to my mail([EMAIL PROTECTED]).My main
configured files are:
extensions.conf:
[from-pstn]
exten = 9711315,1,Dial(SIP/3000,30)
exten = 9711315,2,VoiceMail([EMAIL PROTECTED])
exten = 9711315,3,PlayBack(vm-goodbye)
exten = 9711315,4,HangUp()
sip.conf:
[3000]
Hi,
Is there a way to setup software loop on one of E1 ports on Sangoma
A104d card? I need to setup loop for my telco operator, but Asterisk is
located 300km from my place.
Cheers,
Marcin
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Junghanns have PCI cards with GSM modules and the drivers
it works great
Best Regards
Thierry
_
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de broadband
Voice
Envoyé : samedi 21 juin 2008 23:39
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet :
Have you configured and tested sendmail?
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I have a small Portech GSM gateway. It works well. It's GSMSIP which
seems to me a better solution than FXO/FXS type interfaces. They make
gateways up to 32 port for E-1 interconnect.
Michael
--Original Message Text---
From: broadband Voice
Date: Sat, 21 Jun 2008 17:38:50 -0400
I am thinking
I posted at almost the exact moment you did Michael! Taking a break
from the move.
On Sun, Jun 22, 2008 at 3:45 PM, Michael Graves [EMAIL PROTECTED] wrote:
I have a small Portech GSM gateway. It works well. It's GSMSIP which seems
to me a better solution than FXO/FXS type interfaces. They make
Take a look here
http://www.smallnetbuilder.com/content/view/30428/82/
Michael Graves wrote a great 2 part article about this.
On Sun, Jun 22, 2008 at 2:13 PM, WideVOIP [EMAIL PROTECTED] wrote:
Junghanns have PCI cards with GSM modules and the drivers
it works great
Best Regards
Thierry
Someone on last Friday's VoIP Users Conference mentioned this and I
agree I'd like to hear it discussed:
Have you done an installation where there are multiple asterisk boxes
where one is doing only vmail, one conferencing etc? If you have,
would you consider talking about what you did on an
When you say GSM gateway, do you mean using cell phones as FXOs or do
you mean using Asterisk coupled with a BTS and having your own little
cell?
Thanks,
Steve Totaro
On Sun, Jun 22, 2008 at 8:13 AM, WideVOIP [EMAIL PROTECTED] wrote:
Junghanns have PCI cards with GSM modules and the drivers
Ok, so now that it's possible to implement SIP over TCP instead of UDP
why would I want to do this? Beyond simply integration with M$ OCS.
And what are the implications for management of QoS? I would expect
that lost packets would be less of a factor.
Thanks,
Michael
--
Michael Graves
Junction has certainly done this as they use Asterisk for some
functions. VM I think. They now use OpenSER for their conference
bridges.
Michael
On Sun, 22 Jun 2008 16:00:48 +0200, randulo wrote:
Someone on last Friday's VoIP Users Conference mentioned this and I
agree I'd like to hear it
C F wrote:
Looks promising will try it and let you know.
Thank you
On Thu, May 29, 2008 at 9:22 AM, Jerry Jones [EMAIL PROTECTED] wrote:
Just noticed end of thread here. Not sure if it was mentioned but the
newer - less than a few years - have a backdoor that works. Have not
been able
Bart,
Did you try the method of inband along with changing the frequencies
at the same time?
Thanks,
Steve T
On Sat, Jun 21, 2008 at 3:29 PM, Barton Fisher [EMAIL PROTECTED] wrote:
OK, tried changing DTMF tone as described on URL and no difference
Bart
Steve, I fooled with dtmf mode and it
Also, when you tried inband, did you set it on the phone as well as sip.conf?
Thanks,
Steve T
On Sun, Jun 22, 2008 at 10:35 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
Bart,
Did you try the method of inband along with changing the frequencies
at the same time?
Thanks,
Steve T
On Sat, Jun
On 6/22/08, Michael Graves [EMAIL PROTECTED] wrote:
Ok, so now that it's possible to implement SIP over TCP instead of UDP
why would I want to do this? Beyond simply integration with M$ OCS.
And what are the implications for management of QoS? I would expect
that lost packets would be less
On Sat, Jun 21, 2008 at 01:47:47PM -0500, Lyle Giese wrote:
I agree with using used chan banks off of Ebay, but I would not touch a
Zhone. I had one and sold it as soon as I could. They are a real PITA to
program and don't pass caller id.
Yeah, I gather they're a bit touchy to
We have a number of clients who have replaced their cell carrier voicemail
with Asterisk (call forward no answer to * box).
One feature they miss is that the cell carriers send the phone a message
showing # voicemails waiting. Can Asterisk do the same somehow?
MD
On Sun, Jun 22, 2008 at 11:23 AM, OCG Technical Support [EMAIL PROTECTED]
wrote:
We have a number of clients who have replaced their cell carrier voicemail
with Asterisk (call forward no answer to * box).
One feature they miss is that the cell carriers send the phone a message
showing #
Yep - tried both and combination thereof - The bad effect of inband mode was
audio went one way after first press
My test app reads back the ANI DNIS at answer (which works), then prompts
for more digits.
It's suppose to read back whatever is heard. I can see it reading back
something, back I
Well, I realize that there must be some proprietary protocol between the
carrier and the phone, since they have a dedicate spot on the cell screen
for # VM waiting...
As for an SMS message, is there a module/app which allows easy SMS
messaging? (I looked a couple of years ago but only found
I just send an email with the voicemail message details.
Cool part about this is the attachment can be downloaded and played on
the phone via windows media player.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent:
We already do thatbut:
If users turn their phone on after being out of range/server/power off, the
carrier sends a VM notification. Also, not all phones have POP client
capabilities...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
OCG Technical Support wrote:
We already do thatbut:
If users turn their phone on after being out of range/server/power off, the
carrier sends a VM notification. Also, not all phones have POP client
capabilities...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On June 22, 2008 11:32:56 am OCG Technical Support wrote:
Well, I realize that there must be some proprietary protocol between the
carrier and the phone, since they have a dedicate spot on the cell screen
for # VM waiting...
As for an SMS message, is there a module/app which allows easy SMS
Yeah, it gets a bit confusing with all the scenario possible - Regardless,
you are right I should stay on 1.2 until 1.4 is ready for prime time, but
now that 1.6 is out, I'm sure I'm in for a long wait. I reposted my bug
again, since I think I may have listed it wrong - it's now
I've been trying to get Telco MWI to work with asterisk 1.6. I'm
currently using the beta9 version. I haven't been able to get this new
feature working on any 1.6 beta release version.
I've turned on the full asterisk debug:
logger.conf:
full = notice,warning,error,debug,verbose
The only
On Sun, 22 Jun 2008, Michael Graves wrote:
I have a small Portech GSM gateway. It works well. It's GSMSIP which
seems to me a better solution than FXO/FXS type interfaces. They make
gateways up to 32 port for E-1 interconnect.
I've used a 2-channel one too. They also do 4-channels ones -
Matt Watson wrote:
On June 22, 2008 11:32:56 am OCG Technical Support wrote:
Well, I realize that there must be some proprietary protocol between the
carrier and the phone, since they have a dedicate spot on the cell screen
for # VM waiting...
As for an SMS message, is there a module/app
On Sun, 22 Jun 2008 18:08:04 +0100 (BST), Gordon Henderson wrote:
On Sun, 22 Jun 2008, Michael Graves wrote:
I have a small Portech GSM gateway. It works well. It's GSMSIP which
seems to me a better solution than FXO/FXS type interfaces. They make
gateways up to 32 port for E-1 interconnect.
Joseph L. Casale wrote:
I'm in Alberta, thanks for the clarification. Did you guys get a Whitepages
listing by chance?
I am contacting Superpages now.
Yes, in BC it's Superpages who publishes the phone book. If I recall
their charge includes both a white and yellow pages listing.
Trevor
Jim Duda wrote:
My Telco service is Verizon FIOS. I know that MWI is working, because
if I pick up an analog phone set attached to the line, I can hear the
stutter tone.
The MWI detection in chan_zap/chan_dahdi is not for stutter tone; it
supports either FSK MWI pulses (slimier to Caller
I'm guessing you're using gentoo. Your LDFLAGS ended up being
backslash. Don't do that, and it will probably build. I'm making an
educated guess, based on your 'omit-frame-pointer'. Don't omit your
frame pointer either, if you ever want help debugging. Go into
/etc/make.conf, find the gentoo docs,
Hello. It's been a while since I last posted (probably because my * works
just fine). I'm working on something to replace call queues in my own
application-specific way and I'm using MeetMe rooms to bridge agents and
clients and do other things.
When an agent needs to be bridged with a client
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