Displaying 20 results from an estimated 2000 matches similar to: "ztcfg error : TE110p error with " CAS signalling on span 1 conflicts with HDLC with ..."
2007 Sep 06
1
14. Re: ztcfg error : TE110p error with " CAS signalling on span1 conflicts with HDLC with ... (Carlos Chavez)
Hi Carlos/All,
Thanks for your reply. I can remove dchan=16 from zaptel.conf
But according to the documentation of Digium and sangoma they mentioning to
use dchan=16.
Are there any specific reason you have experiance regarding this and I am
confusing that what this is included to the documentations.
Regards,
Vidura.
On Wed, 2007-09-05 at 20:26 +0530, Vidura Senadeera wrote:
> Dear All,
2007 Aug 21
6
Saftware RAID1 or Hardware RAID1 with Asterisk
Dear All,
I would like to get community's feedback with regard to RAID1 ( Software or
Hardware) implementations with asterisk.
This is my setup
Motherboard with SATA RAID1 support
CENT OS 4.4
Asterisk 1.2.19
Libpri/zaptel latest release
2.8 Ghz Intel processor
2 80 GB SATA Hard disks
256 MB RAM
digium PRI/E1 card
Following are the concerns I am having
I'm planing to put this asterisk
2005 Mar 29
1
Avaya Partner ACS system, pre 7.0
Hi all,
I've got an old avaya partner acs <7.0 system here. I'd like to add a
simple voip bridge so I can hook up our remote offices. From my
research, it would seem the pre-7.0 series doesn't have a t1 port, so if
I wanted to do this, I would have to feed the avaya system fxs ports
from the asterisk box.
Does that sound about right? Has anybody ever done this? Does
2009 Jun 29
1
ISP< ->Asterisk <-> ATA <->DIALUP
Hellow,
* I have a problem with dial up signalling. currently I have configured
asterisk server and E1 card to ISP. then other side I am having ATA to PC
for connecting internet through DialUP connection. is it possible and please
send me the procedure how I can do it ?? *
ISP< <-> Asterisk <-> ATA <-> DIALUP
--
Thanks & Regards,
Vidura Senadeera,
Sri Lanka.
2007 Jan 19
1
Re: asterisk-users Digest, Vol 30, Issue 79
>
>
> Hi,
>
>
> I checked by changing to from-zaptel, but no luck yet. Pls guide me on
> this.
>
> Regards,
> vudura senadeera
>
>
> ------------------------------
> >
> > Message: 9
> > Date: Fri, 19 Jan 2007 16:47:18 -0000
> > From: "Robert Jenkins" < raj@jrw.co.uk>
> > Subject: RE: [asterisk-users]
2012 Feb 14
2
Asterisk + Avaya (CM5.2) H.323 trunk Link
Anyone have an H.323 trunk tied between their Avaya and Asterisk box that
works? I am having some issues trying to get the two systems to connect. I
am using the ooh323 channel to try to make the connection between the two
system. I have all my configs if anyone would like to look over them. If I
do a trace on Avaya I get a denial event 1191: Network Failure.
Thanks!
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2006 Apr 26
1
Intergrate Asterisk IP PBX with Legacy PBX, continuing existing funtionality of legacy pbx
Hi All,
I would like to explain the layout that i am trying to achive. I am so
helpless on this regard.
So here is the story ........
" This is with regard to the setup which you can find at the
"Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am
attaching the picture for your information.
Now I am taking a challenging step to of integrate IP PBX with our
2007 Jan 19
1
Integrating asterisk with Toshiba Astrata DK380
Deat all,
I am in middle of integrate Asterisk with Toshiba astrata legacy pbx.
Following is my setup
*Asterisk <-> Digium TE110P <-> E1 card in toshiba pbx <-> Toshiba PBX*
A =============================================> B
C <============================================ D
Asterisk PBX and strata PBX connected using back to back E1 cross cable.
Physicall connectivity
2010 Jul 16
1
IAX endpoints not Registering after upgrage from Asterisk ver 1.4.26.1
Dear All,
I am experiance a issue with my IAX clients. I have upgradeed Asterisk to
1.4.28
After then IAX clients are not working and It's not registering even.
Please help.
Asterisk previous version - 1.4.26.1 ( for this worked fine)
FreePBX version - freepbx-2.5.2
--
Thanks & Regards,
Vidura Senadeera,
Sri Lanka.
msn/yahoo/skype Ids - vidurased
-------------- next part
2009 Apr 16
1
Can Asterisk bridge between a SIP client and a Cisco Call
>
> Hi,
You can achieve this by integrate CCM and asterisk using SIP trunk.
In CCM you can create SIP trunk, After creating SIP trunk in between CCM and
asterisk, you have to configure dialplan on CCM to pass the calls to
asterisk.
One the caller id comes to Asterisk you have to use extension.conf to route
the calls.
You can also try with freepbx GUI to configure inbound route, it makes
2007 Feb 14
1
CAS signalling on span 2 conflicts with HDLC with FCS check on channel 20
hello my friends,
when i make a genzaptelconf i get this message
********************
CAS signalling on span 2 conflicts with HDLC with FCS check on channel
*******************
Any idea Please?
I m installing zaptel 1.4
i checked in "http://bugs.digium.com/view.php?id=7860" that it's a bug
but beacause i m a newbie in asterisk i can't undrestand what exactly mean
Thank You
2007 Aug 24
2
TE210P digim card PRI problem
Dear all
I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back and second E1 card on Direct Telcom for outgoing for outside now i got this error when i call on avaya PRI
asterisk think PRI_CPE and remote end also CPE
i have configure /etc/zaptel.conf
span=1,1,0,ccs,hdb3
2007 Mar 07
1
Back to back E1 - asterisk <=> toshiba pbx - Call droping issue
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2003 Nov 13
1
asterisk solution.
Hi All,
We have a Avia Difinity G3R to a Avia EPN connected through 2 dedicated
T1 dsics, can I use an Asterisk solution to replace the T1's? If so can
anyone give me feedback on what components I should be looking into to
help me get started?
Thanks in advance,
Steve.
2007 Jul 04
2
Upgrade Asterisk
Hi!
Just ashort question - obviously I am too stupid too find the answer on
the net. :-)
I want to upgrade a running Asterisk 1.4.2 to 1.4.6 - what will I have
to do? Just install it over the existing version? Do I need to backup
the configuration? Will I need to reconfigure the source or will the new
version "import" my old settings? Will I need to update Zaptel and
Libpri too?
2007 Aug 21
0
Saftware RAID1 or Hardware RAID1 with Asterisk (Vidura Senadeera)
>
>
Dear all,
Thanks for the greate explanation regaing Software/H/W Raid. This details
better but on voip-info.org/wiki pages.
Thanks lot agian.
Regs,
Vidura Senadeera.
======================================
Dear All,
> >
> > I would like to get community's feedback with regard to RAID1 ( Software
> or
> > Hardware) implementations with asterisk.
> >
2007 Mar 15
0
Re: busy/hangup/answer detection in PRI E1 channels (Vidura Senadeera)
> Hi Gareth Blades & Doug,
>
> Thanks so much for for the feedback. I have searched on lot of documents
> but couldn't able to find clear answer regarding it.
>
> I hope you guys replies are very much help all in aterisk community.
>
>
> Thanks & Regards,
>
> Vidura Senadeera,
>
> Network Engineer,
>
> Debug Solutions
>
> Sri Lanka .
2007 Mar 09
0
Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping issue [ ref:00D36mPe.50032wycQ:ref ]
Hi All,
Thanks for every one who helped me on this regard. I think i was able to
rictify the problem.
what i did is remove
callprogress=yes
usecallinpres=yes
and restart asterisk. Today i didn't report any drop calls.
Many thanks for Eric. :)
I hope this situation will continue.
Regards,
Vidura.
On 3/8/07, Vidura Senadeera <vidurased@gmail.com> wrote:
>
> Hi,
>
>
2007 Mar 08
0
Fwd: Back to back E1 - asterisk <=> toshiba pbx -Call droping
Before studying your configs, what have you tried so far?
Did you change this?
Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4
to span=2,0,0,ccs,hdb3,crc4.
Here is the documentation on voip-info for why it may be the cause of
your issues
http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax
span definition format:
2007 Mar 07
0
Back to back E1 - asterisk <=> toshiba pbx - Calldroping issue
As these problems are very time sensitive and frustrating, I suggest you
document each change you make and do them one at a time so you can
actually know what the problem was and not introduce new problems in the
process.
Find someone who is on the phone quite a bit and will give you an honest
evaluation of the call dropping situation (unless you yourself are
experiencing this issue too).