Displaying 20 results from an estimated 2000 matches similar to: "How to make * don't strip the leading 0"
2004 Jul 12
1
R: How to make * don't strip the leading 0
> Is it possible to tell asterisk not to strip the leading 0
> of *incoming* MSNs? I use asterisk with i4l and whenever
> I get a call from an long-distance party, the leading 0, which
> should be there according the german numbering, is not.
Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't stripped already by the phone company? I think the easiest
2004 Nov 23
1
Fax over SIP Problems (sorry for this topic ...)
Hello everyone!
I tried to send a fax over SIP with an Asterisk Server in the middle (no
Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway
is external). Whenever I start sending a Fax to a PSTN destination, the
Call gets answered and asterisk tries to build a native bridging:
-- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and
SIP/xxx-3ef8
Then the following
2006 Sep 14
3
One way audio problem on gateway to PSTN after some time, no NAT involved
Hello everyone,
since some weeks I experience strange problems on my gateways to the
PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that
SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN
What happens is, that after a while (uptime was a least two days) the
gateway starts to not transmit audio to the PSTN on outgoing calls, but
the caller can still hear the called
2004 Jun 01
0
Call Transfer over Fritz!-ISDN Card with i4l does not work
Hello everybody!
After checking the complete wiki and the mailinglist archives I still
haven't really found out why the following constellation does not work.
We have an asterisk-System with some SIP-Phones and an old ISA
Fritz-ISDN-Card used with i4l. The whole system is integrated in out
(ISDN-)PBX for testing.
The ISDN-Card is properly configured, as we are able to phone out,
receive
2006 Mar 31
1
transcoding g723 or g729 on asterisk
Kai,
Thank you for the reply.
I didn't want to bother the list too much. However, after reading I discover
I don?t have a clear cut way of doing transcoding.
Can somebody direct me to where I can get document to get this transcoding
done.
My set up
>From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same
asterisk) g711 to chan_ss7] -----> [pstn]
And vice versa.
I
2006 Mar 31
0
Transcoding on asterisk
Hi all,
Thank you for the reply.
I didn't want to bother the list too much. However, after reading I discover
I don?t have a clear cut way of doing transcoding.
Can somebody direct me to where I can get document to get this transcoding
done.
My set up
>From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same
asterisk) g711 to chan_ss7] -----> [pstn]
And vice
2004 Sep 06
0
SIP-Channels cannot be created after a while of running asterisk ...
Hi list!
I've got a strange phenomen running asterisk for a while. After about
two or three days without restarts, asterisk is not able to create
SIP-Channels anymore, but gives me messages like
Sep 4 00:12:06 WARNING[7175]: Unable to allocate channel structure
Sep 4 00:12:06 NOTICE[7175]: Unable to create/find channel
A reason this happens could be "hanging" SIP-Channels,
2002 Jul 28
1
"For ethernet, no packet uses less than 64 bytes" - why?
Hi
Well, subject says all. In Chapter 9.2.2.1, TBF, the parameter mpu
or "minimum packet size" is explained as:
> A zero-sized packet does not use zero bandwidth. For ethernet, no packet
> uses less than 64 bytes. The Minimum Packet Unit determines the minimal
> token usage for a packet.
In my understanding an ethernet packet needs at least 14 (2*6+2) bytes or
54 bytes if
2004 Jun 22
3
Asterisk answering only one (dialed-) Number on a PTMP (German "Mehrgeräteanschluss")?
Hi,
please excuse my poor englisch.
Is it possible to connect a (privat Test-Asterisk) to my privat ISDN and allow him to only answer one dialed number?
We have 3 up to 10 Numbers on each (Euro-)ISDN (2 b-chanels), it cant't be done by the last Digits cause the numbers are completely different.
For Example:
I have 3 Numbers (641717, 928752....)
Is it possible to tell Asterisk (in
2005 Jan 31
5
Announcement to caller when called party has picked up - without initial Answer()?
This is super easy to do. All you need to do is to put that announcement in
a MP3 and set a musiconhold class for that incoming Zap channel. Then
basically when ever that PSTN number rings, Asterisk will play the MP3
stream "Your call may be monitored or recorded, please hangup if you do not
agree...etc" in a loop until the line is answered. Caller doesn't pay a
single dime to
2005 Sep 28
2
chan_capi-cm, Euro ISDN bus: 2 extensions on same BRI port not working
Hello,
I am using a system with an AVM ISDN PCI card (fcpci) and asterisk with
chan_capi-cm-0.6. The hardware is connected to a Siemens Hipath 3550 PBX. As
a BRI connection has 2 channels and allows 2 simultaneous calls,
numbers/MSNs 6391 and 6392 were for provisioned for each channel. The system
is working (partly, read on), the trick is the correct cable wiring and
setup the PBX's port
2006 Jan 20
2
AVM C4, asterisk-1.0.8, /etc/asterisk/capi.conf
Has anyone a working /etc/asterisk/capi.conf example for Germany or
Switzerland using the AVM C4 - ISDN Card.
I try to connect asterisk to 3 wires BRI-ISDN (Swisscom).
I appreciate your help and it would save me a lot of time, figuring
it out by myself.
regards, Claudio
2004 Dec 02
2
more than 3 msns with chan_capi
I have 10 msns on my isdn bri incoming (AVM Fritz, chan_capi 3.5)
When i set in capi.conf
[interfaces]
msn=1,2,3,4,5,6,7,8,9,10
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=tcom-in
echocancel=yes
devices=2
Only msn 1,2,3 are working.
What can i do?
Thanks
Jens
2004 Feb 17
5
chan_capi problem
Hi to all
I've mada up my mind and i tried to change from i4l to chan_capi, following some councelling from the gurus.
I compiled it up, and when i try to load it in modules.conf, i get that wonderful message and Asterisk does not start:
[chan_capi.so]Feb 17 09:21:40 WARNING[16384]: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group
Feb
2004 Sep 22
1
(euro)ISDN: complete silence / can't hear a word.
Hello,
I just got my isdn-card working together with i4l and asterisk.
Everything seems to be working fine: I can accept calls coming from the
outside and I can dial out. Even setting the msn works like charm but my
problem is that I cannot hear a word. There's complete silence in both
directions.
Any idea what could be the cause?
Thanks for your help,
Gunther
Lspci:
0000:01:07.0 Network
2005 Sep 19
1
i4l ring indication problem, again...
I can't find solution anywhere. I googled and find people with the same
problem but there was no answers on how to fix this.
I have W6692 based PCI cards that uses hisax driver (card type=36).
Card is working fine under asterisk with i4l modem driver for incoming
calls. If I want to dial out using some sip phone I don't get ring
indication. Phone is ringing and I hear only silence until
2003 Jul 23
3
2 B channels for ISDN cards
Hi,
Is it possible to use 2 B channels simultaneously
with either I4L or CAPI drivers?
We use AVM A1 (Fritz) PCMCIA with I4L driver and
AVM B1 PCMCIA with CAPI driver.
Thanks,
Michael.
2004 May 12
1
Multiple ISDN controllers & Capi
Hi,
how can I set up multiple ISDN controllers with chan_capi, so
that every controller has its own configuration (MSNs to
listen etc...) ?
I know I can configure "controllers=1,2" in chan_capi's capi.conf
but then the controllers have the identical configuration
and context within asterisk.
My setup: I have 2 BRI-controllers (in one AVM C2).
1. BRI: point-to-point, Number:
2004 May 05
7
* & ISDN-BRI-PTP & DID & ISDN4Linux does not show incoming number
Hi,
I am using Asterisk on a DSS1 ISDN-BRI with ISDN4Linux (and a Fritz Card PnP).
The ISDN-BRI is in PTP-Mode (Point to Point "german: Anlagenanschluss")
which is enabled within I4L with "hisaxctrl fcpcipnp0 7 1".
Everything works fine except that I can not see the called number/MSN
of incoming calls within Asterisk and because of this I can not route
incoming calls
2005 Sep 21
2
Submitting ISDN-MSN from a SIP-Phone
Hello,
i wonder why i didn't find a solution for this problem yet, because it
seems very common:
I have an asterisk server with an AVM (Fritz) ISDN-Card (BRI), and some SIP-Softphones which i can call from outside by calling the phonenumber of the
Asterisk-Server and then dialing the number of the SIP-Phone.
If I make a call from a SIP-Phone into PSTN, only the MSN of the asterisk-server
is