Hi ALL
Have installed [EMAIL PROTECTED] 1.0
On FWD DID's, appears that 2 calls are generated to the inbound extention. I
have confirmed this on a number of friends boxes also. Does anyone have a fix
for this ? I set the DID simply to a custom context and it did the same...
Anyone have a way t
> Anybody using a Sipura 3000 for FXO with Asterisk?
>
> Mine is working except for one small nit...
>
> When a call comes in from the PSTN, the Sipura answers it and then
> passes it on to Asterisk, which plays extension ring tone.
>
> I'd prefer for the POTS line to stay on-hook while the e
Hello
We are running Asterisk CVS 22/12/04 and pwlib/oh323 pandora version to work
with our call agent.
Unfortunately **VERY** frequently, asterisk stops responding and goes to 99.9%
CPU. There is no debug output or other information that indicates there is a
problem...
Rather than continually
Hello
I'm interested in setting up a calling card application on asterisk. I
noticed a number in the wiki, both free and commercial. To experiment with,
I'm after a GNU licenced app...Which one would you recommend ?
Regards..Peter
--
Open WebMail Project (http://openwebmail.org)
Hi Mark
Funny you should ask this question, I just spent yesterday integrating
building asterisk with h323 support to connect to a Cisco call agent.I
cant say if it will work for you but it compiles and loads nicely ! I will be
testing this evening
# cd /root
# wget http://www.voxgratia.
Hi Mike
FOr a home solln, $1000 isnt overly cost effective and the payback period for
myself would be too long. The technology is interesting and as everyone says,
its **supposed** to work but I hate being the guinea pig.
The dock-n-talk is interesting and supposedly it does work ok, the signall
> Date: Sun, 06 Mar 2005 20:03:52 +0100
> From: Thomas Trepper <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Cisco 7960
> To: asterisk-users@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> Hi all,
>
> i am new to this list and i
Hi ALL
I'm looking for feedback on how well this unit integrates into asterisk via an
ata. Is the audio quality any good as thats the first thing to upset the wife
if its no good.
I'm looking for a "reasonably priced" GSM gateway 1800mhz for use in Australia
that works with an ata. Quite happy
Hi Mark
I've been involved with IRLP for about 5 years and am one of the original
install team. I've gone through the emmotions of allowing "other" networks
connect to IRLP and I know its caused some lots of headache.
As far as a closed network goes, yes there is LOTS of passion to keep it HAM
o
I just upgraded a 7960 from Call Manager to Sip 7.3
Not having a clue, I couldnt load on 7.3 You will need to load on a version 3
sip image, then load on a verison 5 and then goto a version 7. As the doco
says, its multistage
Anyway, the 7960 works beautifully on the asterisk box, have the dire
I'm trying to configure a 100xp fxo card for the first time but am not able to
get the channel type ZAP recognised
app_dial.c:743 dial_exec: Unable to create channel of type 'Zap'
WHen starting asterisk with -vvvgc i see
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/z
Hi Manny
I have a sipura 3000 connected to asterisk and I must say, its not a bad
sollution. The units seem to be pre-configured to the US phone system and you
need to do some work to get them working properly, namely the hangup tone
detection...
The audio levels aren't too bad, default they cer
Anyone in Australia using these cards ? I've been using a sipura and had to
do a lot of research to get the thing to detect hangup and other subtilties on
the Telstra network.
These boards work OK in Australia ?
Many Thanks Pete
--
Open WebMail Project (http://openwebmail.org)
___
Nathan
If you want more specific information for AUS, drop me a direct mail. My
Sipura 3000 passes the PSTN call (on hook) to the asterisk box and also the
CLIDNUM.
My only problem is that the asterisk box then sends the caller-id to the
handset connected to the sipura, I can get the username bu
Hi
I'm confused by the asterisk WIKI syntax for DISA. I want to only let a CID of
say 1234567 pass through DISA, which calls an extension of 333
In reading the documentation, I thought it should look like this
exten => 333/1234567,1,Authenticate(1234567)
exten => 333/1234567,2,DISA(no-password|m
Set the grandstream to RFC2883 in your phone, this will work with asterisk.
also define DTMFMODE=RFC2883 in sip.conf under the phone definition.
Pete
--
Open WebMail Project (http://openwebmail.org)
-- Original Message ---
From: "Tomas Florian" <[EMAIL PROTECTED]>
To: <[EMAIL P
Richard
Thanks for the response. All DTMF settings are RFC2833. The call through
IPKALL I cannot control **their** DTMF settings so I'm a little confused as to
what the settings should be...My friend and I have been carefull to use
RFC2833.
Any other suggestions ? :-) This is super weirdn
Hello ALL
This is bending my head and I'm hoping someone can help. Call flow as follows
Sipphone -> sipphone.com -> asterisk -> sipura 3000 (PSTN port)
the sipphone is calling to my pstn line on the sipura. Works FANTASTIC, pin
number on sipura entered, DTMF to PSTN decoded and number dialed,
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