Steve Totaro wrote:
This link show how to send SMS using HTTP(s) and the format of the
URL.
http://www.kannel.org/download/1.4.1/userguide-1.4.1/userguide.html#AEN4201
The previous link is good news to me. Now I can do anything by
hitting a URL. it is so simple.
I've been asterisk's smsq
Hi Mahesh,
You can use below code for extensions.conf file for time based call flow as
per your requirement.
Please, comment your code in extensions.conf and then use below code. You
can also need recording one sound file non-working-hrs.gsm for non working
hours play file.
[default]
exten =
Thanks for reply sir,
I have one doubt sir , I have hundred DID's like 4578901 to 4578999 .
every DID has 5001 to 5099 extensions.
how can I send the voice mail on it. if I make as with your dialplan i need
to make every DID's point of.
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services
Le 22/06/2011 01:10, ERIC HERRON a écrit :
I know Asterisk 1.8 can send out texts via SMS()
Can I send Asterisk a text via a DID and it do something?
[...]
You can receive SMSs using smsq (at least in 1.4) But be aware that most
of mobile carriers (eg France) send SMSs to landlines number
Hi khalid
thank you for your response i have verify the chapter that talks about
contexts and also menu in asterisk book
and now i can create my menu without issue,if i have any question i will
send an email to this perfect list
thanks evrey one for the help and support
Kind Regards
Thanks for the response.. I just got the opportunity to try this with the
wait time adjusted to 15.. and got the same result...
[2011-06-22 04:42:47] NOTICE[19692]: pbx_spool.c:339 attempt_thread: Call
failed to go through, reason (3) Remote end Ringing
so far I've been unable to identify
Hope following will help you get some idea.
[default]
exten = _45789XX,1,Set(VMNO=${EXTEN:-2})
same = n,GotoIfTime(9:00-19:00,sun-thu,*,*?:NON-WORKING-HRS,s,1)
...
...
[NON-WORKING-HRS]
exten = s,1,Playback(non-working-hrs)
exten = s,n,VoiceMail(50${VMNO})
exten = s,n,Hangup
[SATISH]
Mumbai,
Hi,
I'm reading Digium HA8 manual (page 55).
It lists several parameters that can be used to configure a BRI port.
More precisely, it suggests that :
Term, is the last parameter
NT/TE is the one just before (starting from the end).
If I'm not mistaken, it seems dahdi_genconf generates
On Wed, Jun 22, 2011 at 10:23 AM, Administrator TOOTAI ad...@tootai.net wrote:
Le 22/06/2011 01:10, ERIC HERRON a écrit :
I know Asterisk 1.8 can send out texts via SMS()
Can I send Asterisk a text via a DID and it do something?
To do something with SMS to a DID, I'd recommend you take a
Thanks, that must mean it's not asterisk but the AGI/AMI software we use
along side it.
On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.net wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:*
My Mitel sets are all in Advanced SIP mode (I think that's what the call it),
have you done this? Once you change to Advanced SIP, you can't go back to
basic SIP.
From: vip killa
Sent: Wed 6/22/2011 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
How do you set them to Advanced SIP mode?
On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.net wrote:
My Mitel sets are all in Advanced SIP mode (I think that's what the call
it), have you done this? Once you change to Advanced SIP, you can't go back
to basic SIP.
Once you set it Advanced, you can't see how to do it, so I can't tell you.
It's under one of the menu's, somewhere...
From: vip killa
Sent: Wed 6/22/2011 8:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call paging interrupts call when using
Has anyone ever seen this error before and have any idea why it
happened?
Thanks
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
--
_
-- Bandwidth and Colocation Provided by
http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf
Page 32
From: vip killa
Sent: Wed 6/22/2011 8:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224
How do you set them to Advanced SIP mode?
Does anybody know if the updated licence on iLBC makes it safe to
include in Asterisk when used in a commercial environment again?
https://sites.google.com/site/webrtc/ilbc-freeware
It seems to require that the Google iLBC licence document is on the
box, but that otherwise it is free-to use by
Hmm, could be im on old firmware but i don't see SIP Enhanced Mode and i
followed instructions in that PDF. would you be able to tell me what
firmware you are running?
On Wed, Jun 22, 2011 at 9:23 AM, Terry Brummell te...@brummell.net wrote:
http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf
R7.2.07.02.00.04
And yes, that is likely the cause.
From: vip killa
Sent: Wed 6/22/2011 9:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224
Hmm, could be im on old firmware but i don't see SIP
i just upgraded to R8.0.08.00.00.04 but i can't find the bootrom upgrade so
i'm still running 02.03.02.02
tested and call is still being interrupted when paging it...
are you running straight asterisk or is something else handling the dialplan
when you test?
On Wed, Jun 22, 2011 at 9:58 AM, Terry
PIAF with * 1.8.3
My bootrom is 2.3.2.2 also.
From: vip killa
Sent: Wed 6/22/2011 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224
i just upgraded to R8.0.08.00.00.04 but i can't find the
Nothing registers on my Asterisk CLI when I attempt to text a DID.
Can I not be texting from a cell phone?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of randulo
Sent: Wednesday, June 22, 2011 8:03 AM
To:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
ERIC HERRON
Sent: Wednesday, June 22, 2011 10:27 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Inbound
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Eric Wieling
Sent: Wednesday, June 22, 2011 10:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inbound SMS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jerry Geis
Sent: Tuesday, June 21, 2011 10:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question on pause
Ahh then it makes sense, FreePBX checking to see if the line is in use, then
sending busy signal instead of interrupting the call
On Wed, Jun 22, 2011 at 10:13 AM, Terry Brummell te...@brummell.net wrote:
PIAF with * 1.8.3
My bootrom is 2.3.2.2 also.
--
*From:*
Do you have BLF working on the Mitel?
On Wed, Jun 22, 2011 at 10:36 AM, vip killa vipki...@gmail.com wrote:
Ahh then it makes sense, FreePBX checking to see if the line is in use,
then sending busy signal instead of interrupting the call
On Wed, Jun 22, 2011 at 10:13 AM, Terry Brummell
Yes.
From: vip killa
Sent: Wed 6/22/2011 10:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224
Do you have BLF working on the Mitel?
On Wed, Jun 22, 2011 at 10:36 AM, vip killa vipki...@gmail.com
Any chance you could send me (off list) you're example provisioning files
(without the SIP credentials and IPs of course)? I can't find them anywhere
online.
On Wed, Jun 22, 2011 at 11:21 AM, Terry Brummell te...@brummell.net wrote:
Yes.
--
*From:* vip killa
Hi,
My setup is:
Asterisk1 w/ HA8+B400M --- Asterisk2 w/ HA8+B400M --- Patton SN4638
Asterisk1 is in TE/PtP (with termination)
Asterisk2 is in NT/PtP( with termination)
Patton is in TE/PtP
The cable between Asterisk boxes is an RJ11M-RJ11M (custom made with pinouts
1 to 1, 2 to 2, ...).
The
Might take a bit, putting out some fires and beating back some aligators here
at work, but I'll get it to you.
From: vip killa
Sent: Wed 6/22/2011 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call paging interrupts call when using Mitel
My setup is:
Asterisk1 w/ HA8+B400M --- Asterisk2 w/ HA8+B400M --- Patton SN4638
Asterisk1 is in TE/PtP (with termination)
Asterisk2 is in NT/PtP( with termination)
Patton is in TE/PtP
The cable between Asterisk boxes is an RJ11M-RJ11M (custom made with
pinouts 1 to 1, 2 to 2, ...).
The
On Wed, Jun 22, 2011 at 01:32:00PM +0200, Olivier wrote:
Hi,
I'm reading Digium HA8 manual (page 55).
It lists several parameters that can be used to configure a BRI port.
More precisely, it suggests that :
Term, is the last parameter
NT/TE is the one just before (starting from the end).
I want to use extension numbers that begin with the # key in my dialplan,
but I can't get my Aastra phone (6731i) to transmit the # key to asterisk.
It works fine for the * key.
I've tried numerous Local Dial Plan patterns in the aastra web configuration
but none of them worked. My current Local
On 06/22/2011 03:32 PM, Steve Davies wrote:
Does anybody know if the updated licence on iLBC makes it safe to
include in Asterisk when used in a commercial environment again?
https://sites.google.com/site/webrtc/ilbc-freeware
It seems to require that the Google iLBC licence document is on
On 22 June 2011 17:14, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:
On 06/22/2011 03:32 PM, Steve Davies wrote:
Does anybody know if the updated licence on iLBC makes it safe to
include in Asterisk when used in a commercial environment again?
On 22 June 2011 17:09, marvin horst fivehor...@gmail.com wrote:
I want to use extension numbers that begin with the # key in my dialplan,
but I can't get my Aastra phone (6731i) to transmit the # key to asterisk.
It works fine for the * key.
I've tried numerous Local Dial Plan patterns in the
2011/6/22 Richard Mudgett rmudg...@digium.com
My setup is:
Asterisk1 w/ HA8+B400M --- Asterisk2 w/ HA8+B400M --- Patton SN4638
Asterisk1 is in TE/PtP (with termination)
Asterisk2 is in NT/PtP( with termination)
Patton is in TE/PtP
The cable between Asterisk boxes is an RJ11M-RJ11M
We ran into this a few years ago. Polycoms and Grandstreams worked fine with
#xxx extensions, but Aastra's would not. Could not dial extensions beginning
with #
We chased Aastra tech support for 2 weeks. They acknowledge the bug, and we
were told they would fix this in their next firmware
If you check the archives you might find the original messages on this topic
from a few years ago...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
[davies...@gmail.com]
Sent: Wednesday,
I am running Asterisk 1.8 on Slackware linux. The system is running fine we
just added a new SIP trunk from XO and I am having problems getting it to
place outbound call.
I am working with XO support but getting little to no help.
Has anyone gotten SIP Trunking working with XO SIP Trunking.
2011/6/22 Richard Mudgett rmudg...@digium.com
My setup is:
Asterisk1 w/ HA8+B400M --- Asterisk2 w/ HA8+B400M --- Patton SN4638
Asterisk1 is in TE/PtP (with termination)
Asterisk2 is in NT/PtP( with termination)
Patton is in TE/PtP
The cable between Asterisk boxes is an RJ11M-RJ11M
2011/6/22 Shaun Ruffell sruff...@digium.com
On Wed, Jun 22, 2011 at 01:32:00PM +0200, Olivier wrote:
Hi,
I'm reading Digium HA8 manual (page 55).
It lists several parameters that can be used to configure a BRI port.
More precisely, it suggests that :
Term, is the last parameter
Thanks, that worked. I was looking in the aastra documentation and the caret
(^) symbol is not mentioned as a dial plan character. Do you know what the
caret in the dial plan signifies?
The following works here:
sip dial plan terminator: 1
sip digit timeout: 99
sip dial plan: X+^
I believe
The link between the Asterisk boxes must not be terminated on both
ends. Termination resistors on both ends is practically guaranteed
to cause link issues.
Very interesting but I'm afraid I still don't get it.
For various reasons, my goal is build an asterisk + patton solution
that
On Wed, Jun 22, 2011 at 1:14 PM, RAJNIKANT VANZA rajniva...@gmail.com wrote:
You can also need recording one sound file non-working-hrs.gsm for non
working hours play file.
The recording should be available for all expected codecs.
On Wed, 22 Jun 2011, mahesh katta wrote:
I have one doubt
Next week I'll be in the hot city of Madrid doing Asterisk/Kamailio training -
The SIP master class.
Maybe we can organize a voip nerd dinner on Thursday evening? If you're
interested, please e-mail me off list and I'll send out more details later.
Greetings
/Olle
PS. E-mail off list :-)--
Try setting echotraining=no in chan_dahdi.conf.
IIRC that pause is for the echotraining to train the echo canceler.
Eric,
Is there another setting ? This had no effect.
This is happening while I am entering the number to dial, I have not
even hit the Dial button
yet on the phone. I hit
I can't help with IP Phone pauses.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jerry Geis
Sent: Wednesday, June 22, 2011 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi All,
I have experiancing strenge issue with my production Asterisk system.
I'm using asterisk vertion 1.4.28 installed cent OS 5.
Issue decription.
I have SIP trunk from local carrier to their hosted PBX( broadsoft). Out
going calls over this trunk working fine and I can make a
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