Hello,
What i am trying to do is . Dail a number and ask if you wana talk to
XXX press 1 and if you dont wana talk press any other key.
For this purpose i am using this
linkhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
.
*I am using this option :- *
*M(**x**)*: Executes the macro (x)
Perhaps slightly O.T.
Does anybody know of (or even better, has experience with)
softphone clients on a blackberry?
Some friends of mine have those devices, but they can only use it for
data, voice has been disabled in their abo, so much of them carry they
business-BB, and their private GSM.
Hello,
We are trying to send faxes by T.38 protocol to a remote SIP proxy from
a local extension. The local extension sends the INVITE, Asterisk sends
the call to the Proxy the call is connected with a regular audio codec.
After a few seconds the remote proxy sends an INVITE with UDPTL and the
On Thu, Dec 03, 2009 at 01:13:13AM +0100, Roger Schreiter wrote:
Hi,
I'm using Sangoma's wanpipe together with dahdi, all
software downloaded today at most recent version.
Hardware is Sangoma A104, a 4xE1 card.
Installation went well.
Anyway, wanrouter status shows a different result
Any one have success with Dial M option, Can some one provide an example?
On Thu, Dec 3, 2009 at 12:57 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Hello,
What i am trying to do is . Dail a number and ask if you wana talk to
XXX press 1 and if you dont wana talk press any other
2009/12/3 Warren Selby wcse...@selbytech.com
Why not do something with Background()? i.e
Playback(you-have)
SayNumber(${numMessages})
Playback(messages)
Background(press-1-or-2)
Then just be sure to record the audio files in the appropriate
directory...
The benefit of using a single
Tzafrir Cohen schrieb:
...
head -n1 /proc/dahdi/*
# head -n1 /proc/dahdi/*
== /proc/dahdi/1 ==
Span 1: WPE1/0 wanpipe1 card 0 (MASTER) HDB3/CCS/CRC4
== /proc/dahdi/2 ==
Span 2: WPE1/1 wanpipe2 card 1 HDB3/CCS/CRC4
== /proc/dahdi/3 ==
Span 3: WPE1/2 wanpipe3 card 2 HDB3/CCS/CRC4
==
Aah the Problem was i am working on 1.4 and in my mind and logic i was
writing code for 1.6.
The example works perfect
On Thu, Dec 3, 2009 at 3:00 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Any one have success with Dial M option, Can some one provide an example?
On Thu, Dec 3,
Hello,
How can you parse a comma separated list using function CUT and AEL ?
I've tried but it displays error message (though is seems to find the
correct value) :
STRING=101,102
VAL=${CUT(STRING,\,,1)};
NoOp(VAL is ${VAL});
Cheers
___
-- Bandwidth
2009/12/3 Olivier oza-4...@myamail.com
Hello,
How can you parse a comma separated list using function CUT and AEL ?
I've tried but it displays error message (though is seems to find the
correct value) :
STRING=101,102
VAL=${CUT(STRING,\,,1)};
NoOp(VAL is ${VAL});
Cheers
Sorry for
Cyprus VoIP wrote:
We set t38pt_udptl=yes in sip.conf and allowed all the codecs to the
local extension and remote Proxy, but it still forces the call to go
back to a voice call.
Define 'internal extension'. Is this a T.38-capable device? If not,
Asterisk doesn't support TDM-to-T.38 FAX
Hello
I have a simple configuration to allow the admins listen to agent calls:
exten = _654,1,ChanSpy(Agent)
exten = _654,2,Hangup()
The problem is... even when the agents hung up... it seems the channels
remain active:
okavango*CLI show channels
SIP/211-b3042018 6...@default:1
Im looking for wifi sip phones that support auto provisioning and work
flawlessly with atserisk. Can anyone suggest me some models?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or
On Dec 3, 2009, at 8:49 AM, Lefteris Zafiris wrote:
Im looking for wifi sip phones that support auto provisioning and work
flawlessly with atserisk. Can anyone suggest me some models?
Don't know of any wifi phone that works flawlessly whatsoever. Best to consider
a DECT style phone.
Fred Posner wrote:
On Dec 3, 2009, at 8:49 AM, Lefteris Zafiris wrote:
Im looking for wifi sip phones that support auto provisioning and work
flawlessly with atserisk. Can anyone suggest me some models?
Don't know of any wifi phone that works flawlessly whatsoever. Best to
consider a
With a little familiarity of how SayNumber works, this isn't hard at all.
All that Saynumbers does is parse out the number into 1, 2 or 3 digits and
then plays files from /var/lib/asterisk/sounds/digits. In the example the
OP posted, ${numMessages} had a value of 23, so 20.gsm and 3.gsm would be
On 3 Dec 2009, at 13:55, Fred Posner wrote:
On Dec 3, 2009, at 8:49 AM, Lefteris Zafiris wrote:
Im looking for wifi sip phones that support auto provisioning and
work
flawlessly with atserisk. Can anyone suggest me some models?
Don't know of any wifi phone that works flawlessly
Cyprus VoIP wrote:
We set t38pt_udptl=yes in sip.conf and allowed all the codecs to the
local extension and remote Proxy, but it still forces the call to go
back to a voice call.
That's correct behavior if T.38 cannot autonegotiate.
What happens in the reverse direction, trying to send faxes
We set t38pt_udptl=yes in sip.conf and allowed all the codecs to the
local extension and remote Proxy, but it still forces the call to go
back to a voice call.
Define 'internal extension'. Is this a T.38-capable device? If not,
Asterisk doesn't support TDM-to-T.38 FAX relay (yet). If it
The Ascom i75 isn't really an 'auto-provision' out of the box WiFi
phone, but it has a fairly painless USB cradle for programming. Works
well with Asterisk DD-WRT.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Cyprus VoIP wrote:
Thank you for your answer. The 'internal extension' is indeed a T.38
capable device that works perfectly when connected directly to the
Proxy/ITSP.
As you said, the key to debugging/resolving this issue is the logger. I
wasn't aware of this file. this is what I have
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: Thursday, December 03, 2009 06:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Wi-Fi sip phones
Hi,
Im looking for wifi sip phones that support auto provisioning and work
flawlessly with atserisk. Can anyone suggest me some models?
I have not yet tested it with provisioning because right now the
provisioning server does not support this model.
But I use a Snom 820 and 870 with an extra
I want to use an asterisk box to provide a voip service to a number of
separate companies.
I have a VOIP provider who I want to trunk with. As far as I can see
it there are 2 options
1. Have 1 SIP trunk to one account at the provider who gives me
multiple incoming numbers; this is less than
I've recently decided to spend idle cycles while waiting for various Astlinux
platform builds to complete on making the contents of asterisk/config a little
more complete, a little more useful, a little more real-world...
I started looking at the possibility of taking JTodd's ISN Freenum
Hi all,
I installed a Linux-HA-cluster with DRBD and Asterisk 1.4 on it.
Actually it might work quite good, failover etc. works, even if this is
not a 0-downtime solution, because current calls are dropped and the
phones not are reachable until they reregister at the Asterisk.
It might work
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Thorolf Godawa
Sent: Thursday, December 03, 2009 2:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Source-IP on
Thorolf,
In the [general] section of sip.conf, set the 'bindaddr' parameter to
the cluster IP. If Asterisk is only bound to the floating interface, it
will respond only from that source IP.
-- Alex
Thorolf Godawa wrote:
Hi all,
I installed a Linux-HA-cluster with DRBD and Asterisk 1.4
- John Taylor j...@vetsurgeon.org.uk wrote:
I want to use an asterisk box to provide a voip service to a number
of
separate companies.
I have a VOIP provider who I want to trunk with. As far as I can see
it there are 2 options
1. Have 1 SIP trunk to one account at the provider who
Set 'canreinvite=no' on all applicable peers?
Cyprus VoIP wrote:
Hello,
We are trying to send faxes by T.38 protocol to a remote SIP proxy from
a local extension. The local extension sends the INVITE, Asterisk sends
the call to the Proxy the call is connected with a regular audio codec.
On 4/12/09 9:28 AM, Scott L. Lykens wrote:
Apologize for not directly answering your questions, however, I'm
considering playing with Remus and Xen in the future to deal with high
availability without dropping calls.
See http://dsg.cs.ubc.ca/remus/ for some details.
I have no idea if it
On Dec 3, 2009, at 5:05 PM, Matt Riddell wrote:
On 4/12/09 9:28 AM, Scott L. Lykens wrote:
Apologize for not directly answering your questions, however, I'm
considering playing with Remus and Xen in the future to deal with high
availability without dropping calls.
See
Fred Posner wrote:
If you're using just SIP to SIP, a better option would be a pure sip proxy,
ala Kamailio/SER, etc. They can survive a failover without a drop.
Agreed. Even if using transaction-stateful relay mode, as long as a
dialog is nailed up, sequential in-dialog messages
Hello -
I am upgrading from asterisk v1.2 to v1.6 and I am seeing a problem with
recording CDRs using MySQL. Unlike all of the other postings and web
pages I have found on this issue, my installation successfully stores
the -first- CDR, but nothing after that.
As background info, I will note
Hello,
Has someone successfully used this QUEUE_VARIABLES() function (in 1.6.2-rc7)
?
I tried to use it as I'm using SIPPEER() but without success.
A previous question about it remainded unanswered (
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466).
Regards
2009/12/3 Danny Nicholas da...@debsinc.com
With a little familiarity of how SayNumber works, this isn't hard at all.
All that Saynumbers does is parse out the number into 1, 2 or 3 digits and
then plays files from /var/lib/asterisk/sounds/digits. In the example the
OP posted, ${numMessages}
Hi guys I seem to be having a problem, I don't know if it's a bug or whether
I'm just doing it incorrectly.
I want to set about 3 channel variables when I originate a call via AMI.
All the documentation I have found says to do it like this:
Variable:
I do this:
Action: Originate
Channel: Local/dial_num...@cfmc_cdi_private
Exten: queue_answer
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=CallAndQueue
Variable: CfMC_QueueToUse=tqe
Variable: CfMC_AgentToUse=1001
Variable: CfMC_DialInfo=SIP/GXP280_18
Variable: CfMC_RingTimeout=30
Thanks Jim.
We're using the 1.6.2 rc's. I'll try it on 1.6.0 and see if it works. If it
does then I guess its a bug in the new releases.
Problem is we need features of the 1.6.2.
On Fri, Dec 4, 2009 at 11:03 AM, Jim Dickenson dicken...@cfmc.com wrote:
I do this:
Action: Originate
Channel:
Hi Alex,
In the [general] section of sip.conf, set the 'bindaddr' parameter to
the cluster IP. If Asterisk is only bound to the floating interface,
yeah, that's it :-)
I have not tested failover right now, but registring of the phones now
works!
Thanks a lot,
--
Chau y hasta luego,
Hi Scott,
Apologize for not directly answering your questions, however, I'm
considering playing with Remus and Xen in the future to deal with
high availability without dropping calls.
thanks a lot for the link, it looks quite interesting.
Unfortunately I think (and this is also my
Thorolf Godawa wrote:
Unfortunately I think (and this is also my experience), a virtualized
Asterisk server will not work on higher load and might loose
UDP-VoIP-pakets what will result in a bad voice quality!
Much has been said of this topic. In general, you are correct; the
effects of
On Thursday 03 December 2009 18:22:19 Matthew Edmondson wrote:
Hi guys I seem to be having a problem, I don't know if it's a bug or
whether I'm just doing it incorrectly.
I want to set about 3 channel variables when I originate a call via AMI.
All the documentation I have found says to do it
hello
all,
i found this error on asterisk CLI while try to play file on SIP channel.
scenario is, manager send a command in meetme and meetme will play file to
user who connected via SIP channel.
chan_sip.c:5041 sip_write: Asked to transmit frame type 64, while native
formats is 0x2 (gsm)(2)
hy
Hope everyone is fine, I have one issue coming in asterisk , What i am doing
is i am generating a callback if some one calls at a specif access number on
asterisk,
Asterisk sends a busy signal to the calling party that he received a request
from party and then sends the call back to the person
exten = 111,1,Set(CallerID(num)=123456)
On Fri, Dec 4, 2009 at 8:32 AM, Masood Ahmed masoo...@gmail.com wrote:
hy
Hope everyone is fine, I have one issue coming in asterisk , What i am
doing
is i am generating a callback if some one calls at a specif access number
on
asterisk,
Asterisk
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