At 08:35 AM 8/24/2010, you wrote:
>The Asterisk Development Team has announced the release of Asterisk
>1.8.0-beta4.
I've now tried all the V1.8 betas including this and I always get a
message telling me to read sip-retransmit.txt when I make a call from
a SIP phone, Aastra480i out a DAHDI line
On Wed, 2010-09-01 at 09:09 -0400, Barry Fawthrop wrote:
> Has any advancement been made to get 3102 operational in either a SIP or
> H323 asterisk environment.
> A post back in time mentioned a downloader service.
> >From the posts and articles I have read, the NCP is acting like a bootp
> and tf
Hi
I know that this topic was on the list maybe dozen of times. But I
have a question regarding the fax support in asterisk, because all the
information I could get does not give me the clear view of if. I read
that Asterisk 1.8 will have strong fax (t.38) support, but I want to
know if these four
Hi,
1. Do you have any experience with receiving incoming SMS on an analog or
ISDN landline ?
How can then you differentiate an SMS call from a voice call ?
>From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems the way to
tell an inbound call is an SMS one is to read the callerid numb
Hi folks,
After a fairly extensive Google trawl, I don't think the following is
possible but would appreciate confirmation from anyone else who has
tried something similar.
I have an AGI (not particularly relevant) which is executed when someone
calls into a specific extension. This AGI finds a
Hello list,
I'm using the following macro when calling an external callphone/GSM
number :
[macro-press1]
exten => s,1,NoOp()
exten => s,n,Playback(/var/lib/asterisk/sounds/prompts/press1)
exten => s,n,Read(INPUT,,1,1,1)
exten => s,n,NoOp(input : ${INPUT})
exten => s,n,GoToIf($["${INPUT}"=="1"]
Hi!
> 1. Do you have any experience with receiving incoming SMS on an analog or
> ISDN landline ? How can then you differentiate an SMS call from a voice
> call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems
> the way to tell an inbound call is an SMS one is to read the caller
2010/9/6 Philipp von Klitzing
> Hi!
>
> > 1. Do you have any experience with receiving incoming SMS on an analog or
> > ISDN landline ? How can then you differentiate an SMS call from a voice
> > call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems
> > the way to tell an inbou
Steve-
> On 09/05/2010 04:08 AM, Vikram Ragukumar wrote:
>> Hello,
>>
>> We are in the process of debugging a voice quality issue for a client of
>> ours that is a VoIP services provider. The client uses a softphone that
>> runs on a pjsip stack.
>>
>> When placing a call using the softphone, it
On 09/06/2010 11:18 PM, Jeff Brower wrote:
> Steve-
>
>>On 09/05/2010 04:08 AM, Vikram Ragukumar wrote:
>>> Hello,
>>>
>>> We are in the process of debugging a voice quality issue for a client of
>>> ours that is a VoIP services provider. The client uses a softphone that
>>> runs on a pjsip s
On Mon, Sep 6, 2010 at 10:02 AM, Barry O'Donovan
wrote:
> Now, ideally, I would be able to act on a 'decision' from a DTMF
> sequence from the agent's handset. I don't think this is possible
> unfortunately. Please correct me if I'm wrong.
>
DYNAMIC_FEATURES within features.conf
--
Paul Belanger
Le 06/09/2010 15:10, Olivier a écrit :
> Hi,
Hello
>
> 1. Do you have any experience with receiving incoming SMS on an analog
> or ISDN landline ?
> How can then you differentiate an SMS call from a voice call ?
> From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems the
> way to tell
Hi,
With a 1.4.35 or 1.6.1.19, I'm facing this behaviour :
- extension 7002 is a SIP hard phone currently configured to forward
incoming calls to extension 7003, when a call is unanswered within a 10s
time frame
- when extension 7001 is calling extension 7002 with a Dial(SIP/7002,20)
statement a
Okay;
So I can use a Digium FXO/FXS type card and use the dial tone to utilize
Vonage with Asterisk. Done it - simple enough.
However.I am wondering if anyone is Cracker-Jack enough to come up
with a way to get SIP credentials? I went as far as asking Vonage
directly and the answer I got
2010/9/6 Administrator TOOTAI
> Le 06/09/2010 15:10, Olivier a écrit :
> > Hi,
> Hello
> >
> > 1. Do you have any experience with receiving incoming SMS on an analog
> > or ISDN landline ?
> > How can then you differentiate an SMS call from a voice call ?
> > From http://www.voip-info.org/wiki/vi
Steve-
We are in the process of debugging a voice quality issue for a client of
ours that is a VoIP services provider. The client uses a softphone that
runs on a pjsip stack.
When placing a call using the softphone, it negotiates the use of G729
codec with the remote
Hi!
>> Yes, typically there is only one SMSC that can send you SMS on a fixed
>> line; look at its Caller ID to identify a SMS call.
>
> Even when the call is coming from a cellphone ?
A SMS is not really a call (at least not in the mobile world), and the
cellphone cannot directly send a SMS
Le 06/09/2010 17:39, Olivier a écrit :
>
>
> 2010/9/6 Administrator TOOTAI mailto:ad...@tootai.net>>
>
> Le 06/09/2010 15:10, Olivier a écrit :
> > Hi,
> Hello
> >
> > 1. Do you have any experience with receiving incoming SMS on an
> analog
> > or ISDN landline ?
> >
I have a very difficult to diagnose problem. We are running Asterisk
1.6.2.11, DAHDI 2.4.0, FreePBX 2.8 on a Centos 5.5 server (Xeon quad
core 4gb). Last week we started having a problem where the server will
randomly stop sending and receiving calls. Asterisk does not die or
crash. You
Hi,
How are shared variables destroyed (the one set with function SHARED) ?
Shall I care about that or are those variables destroyed whenever associated
channel is destroyed ?
Regards
--
_
-- Bandwidth and Colocation Provided by
On Mon, Sep 6, 2010 at 5:24 PM, Administrator TOOTAI wrote:
> As stated by Philipp, SMSC is unique. However -in France at least- SMS
> sended to landlines are altered and sended as voice messages by the
> operators. For messages from Orange you will recognize that's a SMS as
> the callerID is the
On 09/06/2010 07:45 AM, Andrew Nowrot wrote:
> Hi
>
> I know that this topic was on the list maybe dozen of times. But I
> have a question regarding the fax support in asterisk, because all the
> information I could get does not give me the clear view of if. I read
> that Asterisk 1.8 will have st
Le 06/09/2010 19:31, Randy R a écrit :
> [...]
> Some of this may have changed, but when I has asterks and a fixed-line
> SMS service from France Télécom, that's the way it worked.
>
End of 2009 SMS sended to landlines where easy to treat, we even setup
an SMS2Mail gw. Those days, we only trea
After upgrading my small test system from Debian Etch->Lenny via a
complete reinstall, I find my g729 hostid has changed. Same machine,
same CPU, same NIC! It doesn't seem reasonable that I have to burn
my one "no-hassle" re-registration for a simple OS upgrade.
The README only says that hostid
Hi Group,
i have a MeetMe Question.
I use "MeetMe(,Ms)" in the Dialplan and if a Conference Room does't exist
Asterisk play (conf-invalid.slin)
If i use "MeetMe(${room},Ms)" (value from DTMF Read) and the Conference Room
doesn't exist Asterisk don't play (conf-invalid.slin) and Asterisk Han
>
> I use "MeetMe(,Ms)" in the Dialplan and if a Conference Room does't exist
> Asterisk play (conf-invalid.slin)
> If i use "MeetMe(${room},Ms)" (value from DTMF Read) and the Conference
> Room doesn't exist Asterisk don't play (conf-invalid.slin) and Asterisk
> Hangup the Call.
>
Use the "i"
Tilghman Lesher wrote:
> On Thursday 02 September 2010 01:13:35 cov...@ccs.covici.com wrote:
> > Matt Riddell wrote:
> > > On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote:
> > > > Hi. I have a soft phone -- expresstalk-- on a computer in my network
> > > > and I use the internal ip address of
Tilghman Lesher wrote:
> On Thursday 02 September 2010 01:13:35 cov...@ccs.covici.com wrote:
> > Matt Riddell wrote:
> > > On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote:
> > > > Hi. I have a soft phone -- expresstalk-- on a computer in my network
> > > > and I use the internal ip address of
Dial with M option
On Mon, Sep 6, 2010 at 10:02 AM, Barry O'Donovan
wrote:
>
> Hi folks,
>
> After a fairly extensive Google trawl, I don't think the following is
> possible but would appreciate confirmation from anyone else who has
> tried something similar.
>
> I have an AGI (not particularly r
Hi Danny,
I don't think this is the issue as I get the same problem when I divert one of
my SIP handsets to that extension, and dial internally. The connection happens
instantly. I can see the file playing on the asterisk console whilst I am
getting dead air.
aF
On 01/09/2010, at 7:54 AM, Dan
Hi Paul,
No cigar unfortunately. I also tried encoding the message as gsm, ulaw and alaw
with no success.
The ISDN interface is alaw and the SIP phones I was testing with are definately
alaw.
Not sure what to do from here. I might just need to bypass the issue using some
alternate way to put
31 matches
Mail list logo
