Displaying 20 results from an estimated 400 matches similar to: "ASterisk Back2back"
2009 Oct 15
2
Asterisk with a Cisco AS5300 gateway
Hi
i test a new equipment on my backbone: a Cisco AS5300 with voice dsp
ressource
connected at a E1 Voice Link.
I want that all call incoming on the cisco 5300 are sent to Asterisk and
all Asterisk outgoing
call are sent to Cisco AS5300.
Actually, i configure the AS5300:
isdn switch-type primary-net5
!
voice service voip
sip
!
voice class codec 400
codec preference 1 g711alaw
codec
2007 Jan 24
0
NewTopic - Asterisk and Cisco AS5300 via E1/PRI
Hi,
I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but
only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch
the hangup.
I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like
it such that I call from * via E1/PRI to the cisco, and call out via R2 to
2003 Jun 23
1
Setting up the E100P
Hello,
I have an E100P, and in the zaptel.conf I have:
span=1,1,0,ccs,hdb4,crc4,yellow
fxsks=1-10
the light on the card is green( BTW what do all those states of the card
that zttool reports YELLO, RED, BLUE ..., is there a doc for zttool?, or
for the card?)
in the asterisks` zapata.conf I have:
[channels]
context=default
switchtype=euroisdn
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
2010 May 12
2
select read/write access by user or group?
This should be simple, but I've been pounding my head against it for
several hours, and I'm not getting the result I want.
I'd like to be able to select shares, whether they are the root of a
drive or /home/username and designate who can read and who can read/write.
Does Samba have it's own groups to go with it's own users?
I really can only use "security = user"
2003 Sep 29
2
cisco AS5300 : problem configuration
I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ...
I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly
2005 May 11
2
Asterisk and Cisco AS5300 or 3600
Guys.
I need some advice on some h323 issues. I need to test connectivity from
Asterisk to a Cisco AS5300 that has PSTN lines and to cisco 3600 voip
routers.
H323 needs to be used here but I was wondering if anybody has linked
Asterisk to these Cisco routers before?
Thank you for any pointers.
2010 May 21
1
predict
Hello I am creating a linear model with the command
net5 = lm( X[,1] ~ PrinComp[,1:5]) where my vector PrinComp looks like this
> head(PrinComp[,1:5])
PC1 PC2 PC3 PC4 PC5
[1,] 1.8626055 -3.34190998 -0.5448889 2.8296751 0.3994096
[2,] 3.1124144 -1.68113572 1.7187314 -2.0162583 -0.2935675
[3,] 3.3538049 -0.05471002 -2.9385065 0.6921495 -2.2743761
[4,]
2006 Jan 24
1
need help asterisk and AS5300
hi All
Any body already setup asteriks call routing to Cisco AS5300 with SIP Server ?
i need informations sample config for that, or can show how to route docs .
thanks
Dirgan
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2003 Jul 30
16
Need help
I do part time consulting work. I need to setup an asterisk system to
allow me to record both inbound and outbound calls to clients. I have one
client that is just a PITA. The client has changed their mind three times
so far and we are at step one.
I have a spare slackware box and a seperate phone line for the consulting
work. I have MCI Neighorhood as my carrier.
What I need to know is:
1.
2009 Oct 28
2
Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)
Hi
Now, my Cisco AS5300 sent call to my asterisk, but two problems:
When i call the phone number, i have:
[Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension '0426000000' rejected because extension not found.
[Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension
2003 Aug 06
1
chan_oh323 + dtmf
Hello all,
I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper.
PSTN ---->AS5300 ---->Gatekeeper ---->Asterisk
I set up a conference room on the Asterisk sever (Room No 1234).
I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper.
I manage to get to the start of the conference
2007 Apr 17
1
TM Malaysia E1 PRI signaling
Anyone configured a E1 PRI in Kuala Lampur, Malaysia with TM Malaysia?
What signaling did they provide, framing, formatting?
primary-4ess Lucent 4ESS switch type for the U.S.
primary-5ess Lucent 5ESS switch type for the U.S.
primary-dms100 Northern Telecom DMS-100 switch type for the U.S.
primary-dpnss DPNSS switch type for Europe
primary-net5 NET5 switch type for UK,
2005 Jun 01
0
Segmentation Fautl / Core Dump with G.729
Hello,
Has anyone experienced a segmentation fault in asterisk for using G729
against an AS5300 in SIP ?
I'm having this problem. Also, any other codec I use gives me incompatible
media (can be read in SIP DEBUG messages).
AS5300 configured for multiple codecs, so is Asterisk.
Tried G711u/A G723 and G.729. Any clues ?
Regards,
Jorge A.
Info:
Asterisk ver 1.0.7 stable
Using AMPortal
2013 Mar 07
2
Asterisk 1.6 + Cisco AS5300
Hello,
I have a Cisco AS5300 connected to Asterisk (1.6.2.9)
Between 15-16 minutes, the call is disconnected without reason.
Here is what is displayed in the debug:
Received an SDES from 10.4.0.10:17399
-- Got SIP response 420 "Bad Extension" back from 10.4.0.10
-- Stopped music on hold on SIP/as5300-1-0000004d
== Spawn extension (dialin, 065939191, 2) exited non-zero on
2003 Jul 25
0
7940 & AS5300 codec issues/questions G.729 & G.711
I've previously been using G711alaw on both the AS5300 and the phones but feel the need for a less bandwidth hungry codec for those users that are connected behind ADSL and so was investigating G.729 but ..
Firstly I found that on my AS5300 I have either G.729r8 or G.729br8 and on the 7940 phones I have G.729a, I'm not sure which interoperate the best with each other and so was wondering
2004 Sep 08
1
OH323 Ignoring PROGRESS indication
Good time of day all!
1)
I am trying to use as5300 and asterisk. As5300 sends calls to me. I
get the following in
* console:
-- IAX2/magrathea/6 is making progress passing it to OH323/R27464
Sep 8 10:57:59 NOTICE[1140046640]: chan_oh323.c:1159 oh323_indicate:
Ignoring PROGRESS indication.
As5300 user does not hear anything, just silense instead of dial tones.
My config is oh323.conf
2004 Jan 19
3
Residential services
Hi folks,
The obligatory newbie disclaimer:
"Hi, I'm new to Asterisk and I have a couple questions..."
OK, now that that's over with:
I've just started working for a small CLEC, and I'm trying to sell * to
my boss as a replacement for some expensive/inflexible/closed-source
software he's been using to provide residential dialtone with for a
couple years now.
2004 Jan 16
2
NO DTMF detection in the Outgoing call with GW Cisco5300
Hello all,
When I generate an out-going call from *, the DTMF detection is not
working ? ASTERISK --> GW AS5300 --> PSTN
But the DTMF is working correctly when it's an incoming call.
PSTN - -> GW AS5300 - -> ASTERISK
Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info,
no way !!!
Is it normal that asterisk try to setup the outgoing-call using ULAW ?
if I
2004 Jun 08
4
AS5300 and Asterisk
Hey all,
I have an as5300 I use for dial in customers, we have 4 PRIs on it.
We have a few free channels on it. I'm wondering if I setup SIP on the
as5300 I can have asterisk use the free channels for dial out.
I'd still have to use my TDM04B for incoming calls, but at least I can
expand my outgoing.
Anyone done anything like this before? I've never messed with VoIP on
Cisco
2003 Aug 01
0
Cisco AS5300 -- Not hearing anything
Hi to all!
I have this config,
PSTN <--> AS5300 <--> ASTERISK
I am using the Cisco as5300 to receive incoming calls
and routing them to Asterisk for IVR.
When I ran asterisk this is what I get when calling
the voicemail demo.
*CLI> -- Executing Playback("SIP/-081058b8", "transfer|skip") in new
stack
-- Executing Macro("SIP/-081058b8",