Displaying 20 results from an estimated 100 matches similar to: "zaphfc problem: overlapdial don't work after update bristuff"
2009 Sep 14
0
DAHDI Dial 9 Receiving Setup Acknowledge
I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make
calls from the Toshiba to Asterisk and internal calls from Asterisk to
the Toshiba. What I can't do is make an call with an outside
destination from Asterisk to the Toshiba. The Toshiba is looking for 9
to grab an outside line then it expects to see the 10 digits. In the
FreePBX dial plan I use 9|. which sends 9 plus the 10
2005 Jan 28
1
1.0.2-BRIstuffed-0.2.0-RC2b and '*8' calls dropping
I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a
call with '*8' - the call will drop after about 20 or so seconds. Is
this a general problem with Asterisk 1.0.2?
As this is the latest release that it appears Klaus-Peter Junghanns has
for public consumption - is there anything I can patch for just this
problem - or has Klaus-Peter Junghanns (or anyone else) been
2005 Jan 03
2
agent with queues remain unavailable during transferred call
Hi,
I'm seeing something I'd like suggestions on:
I have a queue with agents that log in using agentcallbacklogin. The
extension that is logged in with is a Local channel. Now, if a call
comes in to the queue and is handled by an agent (in our case using
Cisco 7960 SIP phones) and transferred (attended) to another extension,
the agent remains unavailable during the remains of the call.
2004 Dec 21
4
hint extension and Snom phones - CVS or stable?
Hi,
does the hint extension work together with the Snom phones in stable? I
don't get an error in the dialplan, but it does not work either.
On SIP/26 I want to monitor SIP/22. This is what I do right now:
extension.conf
[incoming]
exten => 955,hint,SIP/22
exten => 955,1,Dial(SIP/22)
sip.conf
[26]
...
subscribecontext=incoming
...
Running Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b
TIA
2005 Jan 23
4
Florz patch for zaphfc
Has anyone had any success using the Florz patch for zaphfc ?
I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN
lines however the users are complaining of crackles on the line which I am
assuming is related to the IRQ issues raised by Florz.
I have tried to use the patch but it errors trying to patch zaphfc.h
Any help would be appreciated.
Regards,
Stuart
--
No virus
2005 Jan 12
3
Bristuff 0.20RC3 loses connectivity after short line interruption?
I installed bristuff0.20-RC3 (which includes * 1.0.3 stable)
It works fine until I disconnect the phone jack for the ISDN line. Even
after plug it back in asterisk still reports that it could not create a
zap channel when I try to dial out and the line gives an engaged tone when
I try to dial.
Re-starting asterisk doesn't solve this, I have to stop asterisk, unload
the modules, reload
2004 Nov 30
0
AW: zaphfc problem
Hi,
I solved the Problem:
the context defined in zaptel.conf:
context = zapincoming
was not in extensions.conf, so nothing worked. ( I assumend this context
is only necessary for calls from zaptel, not to it)
This may be a bug: in the verbose log I found nothing like "[chan_zap]
Could not initialize ..., due to ....", this had helped alot and had
saved alot of hours (-;
2004 Dec 03
0
ZAPHFC, Asterisk does not load with signalling = bri_net_ptmp
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I am trying to get zaphfc running in nt mode with asterisk.
2004 Jun 30
6
zaphfc - hfc pci based ISDN card : point2point & DDI
hello,
anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode?
what kernel ?
and second question mISDN driver .. anyone has working solution with
mISDN and maybe fritz card?
what you suggest for DDI -> point2point mode (card,kernel,chan_..., ...) ?
thank you,
Tomaz
2006 Apr 02
1
Two ISP
Hello all.
First of all, please be a bit indulgent to my poor English :-).
Second, this message is "kinda" BIG, so if you don''t like BIG
messages, simply don''t read it :-).
I''ve read http://shorewall.net/2.0/Shorewall_and_Routing.html
and http://shorewall.net/MultiISP.html, however I still a bit confused how
to organize what I need :-).
I''ve a
2011 Jan 17
0
Sangoma A104d / overlapdial=yes / dial with audio one-way issue
Hi,
I am facing an audio-problem with the dial application and I (!) think,
that it is connected to the dahdi parameter "overlapdial=yes". Sangoma
support does not see any connection between this. But when enabling this
option I face with some(!) dial-partners a audio one-way issue (the
called party can not be heard). Only using PSTNS (germany e1 trunk) - no
voip.
Is there any
2010 Sep 15
1
One way audio when overlapdial is set to yes
Hi Group,
I am currently facing a dead end and any help will be much appreciated.
I have an a104d installed in an asterisk box, two of which is configured on ISDN
pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am
getting one way audio when a local on the hipath tries to make a pstn call but
no issue on incoming calls from pstn going to the hipath locals.
local
2004 Nov 20
2
zaphfc sound problems
hi list.
after my unsuccessfull experiences with mISDN i tried again to implement
a zaphfc based solution.
problem is: sound on calls via capi is stuttering/broken and therefore
unuseable.
my conf:
- cel 1300, 256mb ram
- avm b1 via capi connected to my outgoing ISDN
- acer surf pci via zaphafc and crosslink cable with termination as
internal bus
- siemens gigaset 3035 as internal phone
-
2005 Apr 06
2
OpenGroupware.Org
Hi all!
I'm thinking on installing OGO on CentOS 4. I've been to
download.opengroupware.org but I didn't find any CentOS/RHEL 4 specific
rpm packages. I'm wondering, since RHEL4/CentOS4 is based on Fedora Core
3, can I use these packages safely? Has anyone ever tried installing OGO
on CentOS 4? Has anyone ever used ogoall-...rpm ?
Thankx in advance.
Best regards,
Pedro
2009 Jun 18
2
dahdi and overlapdial problem
Hi there,
we have a problem with dahdi and overlapdial. We are running an E1 in
Germany and are in need of overlapdial. The E1 is connected to a Sangoma
A101.
As soon as overlapdial is set to "yes" we have problems with incoming
audio on the dahdi channels. When set to "no" all audio is fine.
Basically we can choose between being able to receive calls or to place
calls
2007 Oct 25
3
Realtime on Asterisk 1.2.24
Does realtime work reliably on Asterisk 1.2.24?
Are there any definitive guides, I can only find bits and pieces here
and there. Any accurate howtos would be of great help.
I am missing func_realtime.so. Where does this file come from?
Asterisk or asterisk-addons? I saw in one of the howtos that it is
needed. Is it needed for 1.2.X or 1.4.X. Also, what about the switch
lines in the
2005 Jan 31
5
Q: PRI leading 0 (area access code) or 00 (country access code) missing on incoming callerid
hi,
on our incoming E1-PRI from german telco Arcor the leading 0 for the
(area access code in europe) and the 00 (country accescode in europe)
are missing on incoming callerids.
only prepending a single 0 is not the solution as suggested by some
writers on this list, because there is no way to differ between national
and international callerids and it's not possible to make the decission
2005 Apr 08
0
OpenGroupware.org Tests
After the initial question on this list about OGO and CentOS 4, here's
what I've done.
I installed CentOS 4 on a qemu image and setup a yum repo pointing to
download.opengroupware.org/packages/fedora-core3/releases/ . I did this
based on the assumption that FC3 packages work on RHEL4/CentOS4.
yum install ogoall
With this command, OGO user and database was created (Postgresql was
2004 Dec 27
1
Make error installing bristuff-0.2.0-rc2b
Hi list!
I'm trying to install bristuff on a newly installed box. The box is
running a rebuild of RedHat Enterprise linux with the latest kernel. The
architecture of the box is x86_64 (Athlon 64), the os is 64 bit with 32
bit compat libs installed. Also the kernel source is installed as well as
the (kernel) development tools.
The bristuff install.sh script downloads asterisk+zaptel 1.0.2
2003 Oct 27
4
Groups in *
Hi list!
I have a little question about groups and Asterisk ... is there anyone out there that can say if Asterisk can do any of this;
We have a customer that want call handling we cant give him with a traditional PBX, and I'm running Asterisk @home so I thought I could give it a try ...
The customer wants that incoming call should go to one group with some phones in it, if the group is