Hi I'm new to this list, so please forgive me off-topic or RTFM-questions.
I have an asterisk/elastix driven phone-environment using Polycom
SoundPoint IP 650 as extensions. When adding just one custom ringtone
(~57KB) in a proper format (ML.wav: RIFF (little-endian) data, WAVE
audio, ITU G.711
Hello all.
I have installed AsteriskNow 1.7.1 with all updates.
I'm able to make outbound calls without any problem (the external calls are
made via an analog line, and the receiver see the CID). However I'm unable to
forward incoming calls to the destination I want. What happens is when I make
On 01/21/11 03:19, Tom Rymes wrote:
On 01/19/2011 10:34 PM, Da Rock wrote:
WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
non-existing call leg on other UA. SIP dialog
'481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up.
Have you tried disallowing
Thank you Kevin. That's exactly the answer I was after. I'll see if I
can get it 'stopped' at our server end.
BTW - the reason I asked in here was so that everyone could see the
answer and, hopefully, do the same.
Thanks again!
-Original Message-
From:
Could you please give me a feedback regarding this issue, I'm not sure of the
answer I got browsing the web
Thanks and Best Regards
Le mercredi 19 janvier 2011 09:14:55, Marc Leurent a écrit :
Good morning,
I have a simple question,
Is this problem would affect also an Asterisk 1.4.38 if
Zeeshan Zakaria
Sent: Friday, January 21, 2011 6:11 AM
For a client I am setting up a system which will use T1 PRI from Primus,
who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have
only
used switchtypes euroISDN and National. Although the documentation says
Asterisk does
Hello all.
I have installed AsteriskNow 1.7.1-64bits with freePBX.
The system has 1 DAHDi card with 2 analog FXO ports (to pstn) and 1 FXS port
connected to a FAX machine. I want the every call received on port FXO-2 to be
redirected to the FAX machine. So, what I configured was that every call
In article AANLkTi=dpw1oewcpvhbgf1g2ymgbjo4yaml0gbs_6...@mail.gmail.com,
Zeeshan Zakaria zisha...@gmail.com wrote:
For a client I am setting up a system which will use T1 PRI from Primus, who
offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only
used switchtypes euroISDN
I know that the 'fix' has just been applied
(https://issues.asterisk.org/view.php?id=18262) - but why does it stop
the moh only to start it again? This, also, seems to cause a CDR
problem (see below).
-- Executing [7000@chambers:1] Park(SIP/2000-0008, ) in new
stack
== Parked
On 01/21/2011 8:49 AM, Vitor Carlos Flausino wrote:
The system has 1 DAHDi card with 2 analog FXO ports (to pstn)
[snip]
However, it seams that when the call is received, the trunk does
not inform the DID
This is because FXO ports do not support DID. You need to route the call
based on
On 01/21/2011 08:37 PM, Tom Rymes wrote:
On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote:
A comparison wouldn't be complete without mentioning Hylafax. Hylafax has a
great infrastructure - tools for integrating with Windows clients, and so on.
Neither spandsp or the Digium FAX code can
On 01/21/2011 8:59 AM, Steve Underwood wrote:
On 01/21/2011 08:37 PM, Tom Rymes wrote:
On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote:
[snip]
Its easy to set up some t38modem channels and some iaxmodem channels for
receiving FAXes. Transmit is more problematic. With this split config,
The system has 1 DAHDi card with 2 analog FXO ports (to pstn)
[snip]
However, it seams that when the call is received, the trunk does
not inform the DID
This is because FXO ports do not support DID. You need to route the
call
based on the port it came in on, not based on a DID.
This is how I have done it.
In FreePBX, under the 'Setup' tab, choose 'Zap Channel DIDs'. Assign a
DID (ex. 12345) to the channel(port).
Then go to 'Inbound Routes' and create a route for the DID and set the
destination to the appropriate extension.
On 01/21/2011 08:21 AM, Vitor Carlos
Am 21.01.2011 12:21, schrieb Vitor Carlos Flausino:
Hello all.
Can you help me find where the CDR's are being stored?
The result of cdr show status is:
Call Detail Record (CDR) settings
--
Logging: Enabled
Mode: Simple
Log unanswered calls: No
*
Thank you for the confirmation
Best Regards,
Le vendredi 21 janvier 2011 14:17:20, Kevin P. Fleming a écrit :
On 01/21/2011 05:59 AM, Marc Leurent wrote:
Could you please give me a feedback regarding this issue, I'm not sure of
the answer I got browsing the web
Thanks and Best Regards
Yes, it does. Bell provides the same as well and it works with Asterisk.
-Bruce
On Fri, Jan 21, 2011 at 7:11 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
Hi list,
For a client I am setting up a system which will use T1 PRI from Primus,
who offer only NI-1 and NI-2 protocols for D-Channels.
Thanks a lot guys for your answers. I'll go ahead with NI-2. I didn't know
its the same thing as National.
Thanks again,
Zeeshan A Zakaria
--
www.visionvoip.com
www.ilovetovoip.com
www.pbxforall.com
On 2011-01-21 10:36 AM, Bruce B bruceb...@gmail.com wrote:
Yes, it does. Bell provides the
On 11-01-21 08:52 AM, Andrew Thomas wrote:
I know that the 'fix' has just been applied
(https://issues.asterisk.org/view.php?id=18262) - but why does it stop
the moh only to start it again? This, also, seems to cause a CDR
problem (see below).
After speaking with Shaun and Russell, this is
People,
I'm trying to force dahdi to load the modules I need to get spans working.
I have two cards, an Digium TE210E (PCI-e) and a Yeastar TDM1600 FXO (PCI)
Actually it is loading just the first of them, it is (wtc4xxp) for TE210E,
but doesn't load
the second module as specified at
Hi Bryant,
The 1.4 box has two interfaces one with 202 ip and the other with
172 ip , the audio code has 172 ip and the ast 1.6 has only 172 ip. Any
ideas ? Both the trunks have t.38 enabled on it. And the way we use fax
is fax machine connected to ata which supports t.38 in both the ends.
I'm setting up a queue for two independent operator phones that are capable of
answering multiple calls at once. It's currently working with the following
settings and Asterisk 1.4:
queues.conf:
[telefonistas]
strategy=roundrobin
;strategy=leastrecent
music=default
timeout=60
retry=0
maxlen=0
Hi,
I'm working with a SIP phone (Thomson ST2030) that support a
multi-registration feature.
It works this way:
- the phone has a main id
- some feature keys can be configured to be tied to supplementary ids (with
a specific id username and password)
- when the phone boots, it will successively
Hello all.
I have a dahdi card with 2 FXO and 1 FXS. I'm able to make calls without any
problem. However, when I have an incoming call, I see the following message on
the asterisk console:
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [s@DID_trunk_1:1] ExecIf(DAHDI/1-1,
On Fri, Jan 21, 2011 at 12:53 PM, Olivier oza_4...@yahoo.fr wrote:
Hi,
Is it possible for a SIP to register twice to the same Asterisk server
using 2 different ids ?
Consulting this list archives gives mixed answers.
Yes, I do this with my Polycom 550, and I've done it with other phones
Hi all.
I have a customer who does some automatic messaging. Back when we were
running Asterisk 1.4.x, they used waitforsilence after amd, to wait for an
answering machine greeting to finish before leaving their message. We've
upgraded to 1.6, and made no other changes and things don't work.
At 10:33 AM 1/18/2011, you wrote:
While that's a useful data point, it's not relevant to the problem. A
significant portion of the SIP stack was re-implemented in 1.8, and Polycom
phones are on the desktops of nearly every Asterisk developer. Since you
aren't using a Polycom, the SIP stack on
Where can I get the latest stable version of spandsp. That will work with
res_fax_spandsp.so. The link listed on the voip-info website is dead. Any
other location for download?
http://www.soft-switch.org/
Thanks
Bryant Zimmerman
--
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