Displaying 20 results from an estimated 1000 matches similar to: "ISP< ->Asterisk <-> ATA <->DIALUP"
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
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 Sep 05
4
ztcfg error : TE110p error with " CAS signalling on span 1 conflicts with HDLC with ...
Dear All,
I'm integrating avaya commuication manager difinity ver 1.0 with asterisk
using B2B E1. following are the details of my H/W, zaptel configs and
software installed.
Digium TE110p
asterisk 1.2.19
cent OS 4.4
zaptel 1.2.18
libpri 1.2.4
etc/zaptel.conf
span=1,0,0,cas,hdb3
bchan=1-15,17-31
dchan=16
when i ztcfg -vvv im having this error message and the E1 is not getting up.
"cas
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
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
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 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 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]
2007 Nov 27
10
Asterisk behind a PIX firewall?
Is there anything special that anyone here has had to do to get an Aastra
phone (on the Internet) to talk to Asterisk behind a PIX firewall?
Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk
server. The Asterisk server's RTP.CONF is set to use 10000-20000. The
phone registers, and will place AND receive calls, however, no audio is
passed. The phone is an Aastra
2007 Dec 03
1
Subject: Newb Question
Hi,
Use orecx, voip call recording and monitoring.
www.orecx.com
Thanks & Regards,
Vidura Senadeera,
Sri Lanka.
Tel - +94114520001
Mobile - +94777766596
yahoo/skype Ids - vidurased
> ------------------------------
>
> Message: 17
> Date: Fri, 30 Nov 2007 08:58:41 +0530
> From: ram <talk2ram at gmail.com>
> Subject: Re: [asterisk-users] Newb Question
> To:
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 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
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).
2007 Mar 07
1
Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping
Hi steve and All,
I'm attaching cat proc/interrupts out put, lspci -bv out put, zapa.conf,
zaptel.conf for your information
Thanks so much for the feedback and I do accordingly. Hope to get rid off
this isue any how.
To day also reported 10 call drops within 2 hours of period.
fook forward to have your support on this regard.
Thanks & Regards,
Vidura Senadeera,
Network Engineer,
2007 Dec 13
0
Didnt get a frame from Channel and call gets
Hi,
Let us know more information about your setup.
Hardware/software details details such as.
server configuration
PSTN cards you are using?? ( E1 or FXO card)
sip.conf, zapata.cons, zaptel.conf config details??
Thanks & Regards,
Vidura Senadeera,
Sri Lanka.
Tel - +94114520001
Mobile - +94777766596
yahoo/skype Ids - vidurased
==================
Message: 5
Date: Mon, 10 Dec 2007
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 Jun 12
2
[LLVMdev] Problems with the tools
I just built LLVM 2.0 on Windows with Visual C++ 2005. After a few tweaks, it produced several nice .exe files.
I tried to compile a simple Hello-World program as given in the documentation:
--------------------------------------------------------------------------------------
; Declare the string constant as a global constant...
%.LC0 = internal constant [13 x i8 ] c"hello
2010 Mar 11
1
argument is of length zero
I am new to R, so I beg your indulgence.
> a='hello world'
> if (grep('hellow',a)) {
print("wow")
} else {
print("wow2")
}
Error in if (grep("hellow", a)) { : argument is of length zero
What is the proper syntax to ensure that "wow2" gets printed?
Thank you for your help!
Dan
[[alternative HTML version deleted]]
2007 Mar 07
1
Back to back E1 - asterisk <=> toshiba pbx - Call droping issue
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