I followed the instructions at http://www.opencall.org/instructions.html
and
http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html
I was able to compile spandsp (./configure ; make ; make install),
manually patched asterisk apps/Makefile (/usr/src/asterisk/apps), as the
Search the mailing lists, this has been answered a million times.
Edit the Makefile and remove the entries for both app_rxfax.o and
app_txfax.o and it will compile fine.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke
Sent: Saturday,
Brian:
Thanks!
I looked through the list and didn't see a
correlation between what I was seeing and those parameters. Must have
missed it.
Thanks for your help.
Andy
- Original Message -
From:
Brian K. West
To: [EMAIL PROTECTED]
Sent: Saturday, June 19, 2004
Hi,
I've compiled h323 from CVS 1.0 of Asterisk. I've copied h323.conf to
/etc/asterisk but still I'm not able to call Asterisk from Netmeeting (I can
call demo of Asterisk witl sip clients, but not Netmeeting as h323 client)
with error:
The person you called cannot accept Netmeeting calls. and
Hi,
so far I had our PRI line connected via Digium TE410P and ZAP channel to
asterisk which worked perfectly. I now have a second line coming in through a
H.323 gateway and chan_oh323.
rxfax() still works and receives faxes with G.711 alaw codec, but I
always get one empty first page via H.323
Hello-
Have you compiled and installed the proper versions of OpenH323 and PWLib?
(Before you compiled the h323 code.)
See the instructions in ~/asterisk/channels/h323/README..
Regards
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London
Hi,
I have asterisk connected to PSTN via H.323 gateway via chan_oh323.
Incoming calls to SIP extensions work, but SIP message 486 busy here from a
busy extension isn't correctly forwarded to H.323.
As a result, a caller from the H.323 side calling a busy SIP extension gets some
rings and then
Hi,
thanks for response. I've followed instructions and just tried ohphone and
it works on simple call to Asterisk. But still cannot call from NetMeeting
with same error.
I still don't know how to setup h323 clients (NetMeeting, OhPhone) to
Asterisk as extensions. I've tried several things but
Hi,
Response below.
In the meantime: I would REALLY appreciate comments from an ATA186 SIP user
who can tell me:
- How to transfer a call without using #-transfer
- Preferably more or less like how we are used to transferring in a classic
pbx system
Noteworthy:
- Which Asterisk version
Hi Folks,
I'm having problem with GS registering in Asterisk.
My setup is the following:
[1755]
type=friend
incominglimit=10
qualify=no
nat=yes
insecure=no
secret=X
dtmfmode=rfc2833
username=1755
host=dynamic
canreinvite=no
defaultip=192.168.0.1
context=sip-incoming
I have dozens of
Hi-
Yes, in the past I've had problems with both Ohphone and NetMeeting, and
H323. Can't remember which is which, but with one program, the handshaking
worked well but no audio, and with the other got audio ok (usually) but
handshaking was flakey.
Now using a Cisco 5300 to call into my asterisk
Where does the date/time stamp from Caller ID come from?
On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30
12:00AM. The Linux time is correct. SayUnixTime return the correct time.
Any Ideas? Does this work?
Thanks!
Eric C. Snowdeal III wrote:
after registering the phones correctly and receiving a 200 o.k.
message i can connect to other registered softphones and pstn
endpoints [ via an voicepulse account ], but after making the initial
connection, i can't hear any sound and i get disconnected after
Is there any way to handle
CDRswith AGI?
Kubat, Philip [EMAIL PROTECTED] wrote:
(Article auto-converted from unnecessary HTML to nice plain text.)
Where does the date/time stamp from Caller ID come from? On my
extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30
12:00AM. The Linux time is correct. SayUnixTime
I've changed the zaptel.conf to set both as internal, and it now seems to
work, which is backwards to the config I thought it should have been, I
would have thought that the Telewest PRI would have been 1 and the GDK 0?
Can somebody confirm that this is the correct definition for timing, if it's
Steve,
your config description (timing) does sound odd. Could you re-post your
revised config files?
Thanks
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 20 June 2004 16:18
To: '[EMAIL
I would also like to know how to insert a pause if possible. A comma is
seen as | not surprisingly.
I have no idea why no quotes.
- Original Message -
From: S. William Schulz [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, June 19, 2004 10:24 PM
Subject: Re: [Asterisk-Users]
They look odd to me for sure, I'm certain (99.9%) that Telewest would not
clock off of us, but as far as I can see, the current config (which allows
the GDK to send and receive faxes) has no external clocking???
Here's the current config:
span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
Here's
Robert Rozman wrote:
Hi,
thanks for response. I've followed instructions and just tried ohphone and
it works on simple call to Asterisk. But still cannot call from NetMeeting
with same error.
I still don't know how to setup h323 clients (NetMeeting, OhPhone) to
Asterisk as extensions. I've tried
shabanip wrote:
Is there any way to handle CDRs with AGI?
There is no need to. Install or create a CDR backend.
Jeremy McNamara
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
I'm trying to switch from one call to another incoming call from PSTN.
When I'm getting a beep I press flash but instead of swithing to the
second call, I'm getting a dial tone. even if I press *0, I cannot connect
to the second call.
Anybody had this problem?
Tx, Bogdan
On Sun, 20 Jun 2004 10:45:00 -0400, Kubat, Philip [EMAIL PROTECTED] wrote:
Where does the date/time stamp from Caller ID come from? On my extensions
ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The
Linux time is correct. SayUnixTime return the correct time.
Hi Steve,
How bizarre, your config doesn't look like it should work too well and
certainly doesn't look like it should improve your fax problem!
I assume that pri_cpe is set for span1 and pri_net for span2 ?
Maybe, just maybe, Telewest reconfigured your PRI to look for clocking from
your CPE
Perhaps I was a little too hasty in my conclusions of dysfunctional fax
on the SPA-2000. It turns out I have a one way audio problem on line
one of my SPA-2000. I have all the correct settings according to the
comments in the wiki, but the problem persists. However, if I do a hook
flash out of
On Sun, 20 Jun 2004, Kevin Walsh wrote:
It's possible that the time/date is also encoded into the Caller*ID
signal. I haven't had cause to look into that. It's possible that
the DECT phones ignore the local time and use the time provided by the
Sipura (if the Caller*ID signal does indeed
Hello.
I have compile asterisk with modifyed prepaid application and populated
the database to! I have fill the card, cardtype, cid, country,
countrycode, reselers. I have make a cid=22 and I have add a user with
username and callerid 22. But I allways get prepaid-no-aaa. Any one could
help me
On Jun 20, 2004, at 9:40 AM, [EMAIL PROTECTED]
wrote:
I'm trying to switch from one call to another incoming call from PSTN.
When I'm getting a beep I press flash but instead of swithing to the
second call, I'm getting a dial tone. even if I press *0, I cannot
connect
to the second call.
You have problems with pgsql. Check it.
Regards,
Gus
- Original Message -
From: Hekuran Doli [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, June 20, 2004 5:27 PM
Subject: [Asterisk-Users] Midifyed-Prepaid-Application
Hello.
I have compile asterisk with modifyed prepaid
Hi,
I posted an error message I was getting while using enum with the latest
CVS, but the problem disappered.
Well, it seems to be intermitten.
The messages below:
Jun 20 15:23:30 WARNING[1218565440]: enum.c:186 parse_naptr: Regex
compilation error (regex = !^+16131234567$).
Jun 20 15:23:30
Your regexp is WRONG
1.1.enum.blah.net naptr = 2 40 u iax2+E2U
!^\\+(.*)$!iax2:[EMAIL PROTECTED]/\\1! .
Thats a valid enum naptr record.
It would translate into iax2:[EMAIL PROTECTED]/11
bkw
- Original Message -
From: Wojciech Tryc [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
anyway now i can use the asterisk with the festival, it seems the problem is
the patch file festival-1.4.3.diff. in the patch file the festival directory
write down as festival-1.4.3 (included the version) but the actual festival
directory is festival (without any version info). so just rename the
[EMAIL PROTECTED] wrote:
Hi,
I posted an error message I was getting while using enum with
the latest CVS, but the problem disappered.
Well, it seems to be intermitten.
The messages below:
Jun 20 15:23:30 WARNING[1218565440]: enum.c:186 parse_naptr:
Regex compilation error (regex =
Try to debug postgresql query's done by app_prepaid...
set log_statement = true in postgresql.conf and watch postgresql
log's ..
maybe it can help you ...
Alex
On Sun, 2004-06-20 at 20:27 +, Hekuran Doli wrote:
bill my local clients registred to my asterisk box using sip
Why not just use extension logic? (which is far more powerful than
app_prepaid can ever dream of being)
Jeremy McNamara
Alexander Didebulidze wrote:
Try to debug postgresql query's done by app_prepaid...
set log_statement = true in postgresql.conf and watch postgresql
log's ..
maybe it can
The number is 1-613-823-1716 and the enum service is e164.org. The most
interesting part is that this is intermittent problem, sometimes it works
sometimes it doesn't work. Again, any other lookups works just fine.
Thanks,
Wojtek
- Original Message -
From: Aaron J. Angel [EMAIL PROTECTED]
This is only intermittent problem!!?!?
Wojtek
- Original Message -
From: Brian K. West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, June 20, 2004 3:50 PM
Subject: Re: [Asterisk-Users] enum problem with latest cvs
Your regexp is WRONG
1.1.enum.blah.net naptr = 2 40 u
I'm trying to get a Carrier Access Corp. Channel Bank I working with a
Digium T100P without success.
What is stranger is that the status lights on the channel bank and T100P
seem to change almost each time I power cycle the channel bank or reset the
T100P.
The channel bank has three status
Hello, I am unsure if I was reading it wrong but someone once told me
the ATA device Grandstream has supports custom ring tones, I have this
device and have no idea how to implement this.. is it possible?? If so
how do I do it? I am using the latest firmware..
Thanks!
--
Stephen Rosebush,
Hi
folks,
I use safe asterisk
to startup and run asterisk in the background. In Safe_asterisk script, there is
a parameter (right at the top ), CONSOLE which I can set to no or something. If
it is no asterisk startup as asterisk -vvvg , if it is set to something the
asterisk startup as
George,
We have this config working. Please give me a call (yeah, I'm at the office
too) and we can walk through your config together.
-Darren
--
Darren Nickerson
Senior Sales Support Engineer
iFax Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638
+1.215.243.8335 (fax)
-
any sample of extension logic around , just to have an idea?
Best Regards
Hekuran
Why not just use extension logic? (which is far more powerful than
app_prepaid can ever dream of being)
Jeremy McNamara
Alexander Didebulidze wrote:
Try to debug postgresql query's done by
Jeremy,
Please explain little bit more how to use the
extension logic for a prepaid app.
Example would be greatly appreciated.
Thanks
--- Jeremy McNamara [EMAIL PROTECTED] wrote:
Why not just use extension logic? (which is far more
powerful than
app_prepaid can ever dream of being)
I updated from CVS yesterday and now everytime I start asterisk, I get the
following message:
config loader has no config file so nevermind.
What does this mean? It doesn't seem to hurt anything, just a tad annoying
to see everytime I run asterisk.
Ich werde ab 21.06.2004 nicht im Büro sein. Ich kehre zurück am
27.06.2004.
Ich bin vom 20.6. bis 27.6 nicht per Email erreichbar und werde die Emails
sobald als möglich ab dem 27.6 bearbeiten. Dringende Anfragen bitte an
Andreas Widrig/CZWIAN/CH/Ascom machen.
Brian K. West [EMAIL PROTECTED] wrote:
Your regexp is WRONG
The regexp isn't wrong.
1.1.enum.blah.net naptr = 2 40 u iax2+E2U
!^\\+(.*)$!iax2:[EMAIL PROTECTED]/\\1! .
6.1.7.1.3.2.8.3.1.6.1.e164.org. 155 IN NAPTR 100 10 u E2U+IAX2
!^\\+16138231716$!iax2:[EMAIL PROTECTED]/16138231716!
I'm having the same problem...nothing changed...just the CVS version.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Sun, 20 Jun 2004, Aaron J. Angel wrote:
I updated from CVS yesterday and now everytime I start asterisk, I get the
following message:
Title: Message
I was wondering if
this is possible. I have a situation where I am connecting to a third
party voicemail system from asterisk. I know this does not make since to
everyone, but it has to be this way. Basically - I have an application
that runs on the Asterisk system and when
E2U+IAX2 -- thats backwards also.
bkw
- Original Message -
From: Aaron J. Angel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, June 20, 2004 6:13 PM
Subject: RE: [Asterisk-Users] enum problem with latest cvs
Brian K. West [EMAIL PROTECTED] wrote:
Your regexp is WRONG
Hi all,
I have 2 incoming telephone lines connected to 2 X100P FXO Cards.
One line is for my family, the other will be for my home office.
I am new to Asterisk, but though that I would need calls being answered in
different contexts.
How can I direct one line to a given context and the other
Brian K. West wrote:
E2U+IAX2 -- thats backwards also.
actually later RFC's specify it in that format...
http://www.faqs.org/rfcs/rfc3762.html
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
Aaron J. Angel wrote:
6.1.7.1.3.2.8.3.1.6.1.e164.org. 155 IN NAPTR 100 10 u E2U+IAX2
!^\\+16138231716$!iax2:[EMAIL PROTECTED]/16138231716! .
So why would asterisk log an error with just !^+16138231716$?
I've probed all our name servers and they're all responding correctly,
however someone else
Adam Goryachev wrote:
On Sat, 2004-06-19 at 06:13, Stephen R. Besch wrote:
So, if someone could brief me on the GPL issue, and (perhaps someone
else) offer a distribution point, it's free for the asking, VB sources
and all.
Stephen R. Besch
Alright, I've waited a long time before offering
This question isn't entirely Asterisk related, but I'm hoping that
someone here may have the knowledge to respond to me anyway. I'm using
Asterisk with several Sipura SPA-2000 SIP devices as FXS adapters. I
would like to have my SPA's automatically provisioned through http or
tftp, but I can't
Andrew Yager wrote:
Hi,
I'm trying to make a dialup internet connection through my asterisk PBX.
When I bipass the Asterisk box, I can get 51600bps. When I run through
the asterisk box, I'm limited to about 21600bps.
Modem connections over 33.6K require that there be only a single
No its backwards from what I have read... I got most of my info from many
sources and E2U+SIP vs SIP+E2U
but then again what do I know.. I have only been using enum for about a year
now.
bkw
- Original Message -
From: Duane [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, June 20,
The sip+E2U specifies a service to be contacted by SIP through the use of
an E.164 to URI (E2U) translation.
Thats in one of the documents that I have.. it depends on the direction of
the translation.
bkw
- Original Message -
From: Brian K. West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
3762 is for h323 only:
RFC 3762 - Telephone Number Mapping (ENUM) Service Registration for H.323
2916 is a bit more general:
RFC 2916 - E.164 number and DNS
Then we have:
RFC 2915 - The Naming Authority Pointer (NAPTR) DNS Resource Record
I was pointing out that E2U+IAX2 was backwards..
Can someone email me pached file festival/lib/tts.scm and
festival/src/arch/festival/wave.cc (for festival version 1.4.3)? my mail is
[EMAIL PROTECTED] Thanks in advance.
Best Regards,
Freddy Setiawan
___
Asterisk-Users mailing list
[EMAIL
You have to do more than that.. patch and compile it manually.
gentoo users:
export USE=+asterisk
emerge festival
bkw
- Original Message -
From: Freddy Setiawan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, June 20, 2004 10:52 PM
Subject: [Asterisk-Users] please mail me
Try RFC3761. It specifies E2U+spec under section 2.4.2. It obsoletes
RFC2916, and nothing has superseded it yet.
Brian K. West [EMAIL PROTECTED] wrote:
3762 is for h323 only:
RFC 3762 - Telephone Number Mapping (ENUM) Service
Registration for H.323
2916 is a bit more general:
RFC
i'm using RH9. Yes, after you send the file, i'll recompile the festival
again.
I think something wrong with the patch file its in the
festival-1.4.3.diff it said:
diff -u -r festival-1.4.3/lib/tts.scm festival-1.4.3-asterisk/lib/tts.scm
--- festival-1.4.3/lib/tts.scm 2003-01-09
Brian K. West wrote:
I was pointing out that E2U+IAX2 was backwards.. but then again asterisk
doesn't really care about that... at this point.
Considering the voip-info.org site has it as that, and that's before we
go on to mention the fact IAX2 isn't in any rfc, and that it's ENUM
Aaron J. Angel wrote:
Try RFC3761. It specifies E2U+spec under section 2.4.2. It obsoletes
RFC2916, and nothing has superseded it yet.
Damn always seem to get these out by 1 errors ;)
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think
so my question changes to:
- How to create a CDR backend?
- I want to run my own scripts.
shabanip
shabanip wrote:
Is there any way to handle CDRs with AGI?
There is no need to. Install or create a CDR backend.
Jeremy McNamara
___
Hi all,
I've compiled and start the modified prepaid with postgresql. I would like to ask if
anyone can give me a sample account to be populate in the database.
I also want to confirm if this correct. I created a database prepaid (createdb
prepaid) and from the asterisk-prepaid(current
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