Displaying 20 results from an estimated 600 matches similar to: "NewTopic - Asterisk and Cisco AS5300 via E1/PRI"
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
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
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
2003 Dec 03
2
Cisco IAD with MGCP
I repost a message I put a week ago:
I have a Cisco IAD 2431 which has MGCP protocol; I cannot make
it to work againts Asterisk; at least there is some MGCP conversation
between them but when I offhook a phone attached to IAD I get no tone at
all.
As anybody managed to get working Asterisk against an MGCP Cisco
gateway ?
Which MGCP version should I use ?
Also I recently
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",
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.
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
2004 Aug 15
0
how can i config a Cisco IAD 2430 config as a sip client
Hello,
I have a cisco ATA 188 registering both of its lines
to * I can place calls between then an to kphone an
MSN messenger (both registering with * too), a few
days ago a friend lend me a Cisco IAD 2430 and I was
willing to do the same thing with it, since it has 24
ports I was willing to to use 24 analog phones with it
however something really weird happens I can place
calls from my ata,
2009 Jan 28
1
E1 conection to a Cisco2600
Hi
I am trying to connect asterisk with a Cisco GW 2600 with E1 pri using a
Digium, Inc. Wildcard TE210P dual-span T1/E1/J1 card 3.3V (rev 02),
Errors:
[Jan 28 17:32:33] VERBOSE[6182] logger.c: == Primary D-Channel on span 1
up
[Jan 28 17:32:33] WARNING[6182] chan_dahdi.c: PRI Error on span 0: We think
we're the CPE, but they think they're the CPE too.
[Jan 28 17:32:34] NOTICE[6182]
2006 Apr 12
1
ASterisk Back2back
hi All
I need your help , for used Digium Card TE405P, for setting this Board AS E1 ISDN PRI.
1 .Current for make sure my config its rights or no I inform my configurations in Board Jumper T1/E1 is Closed is that rights or no ? for E1 i closed the Jumper.
2. I Want To seeting E1 in ASterisk/PC Back To Back To Cisco E1 AS5300 Use ISDN Signaling,
my configutration :
2005 Mar 19
3
Asterisk and Cisco AS53xx/54xx Access Server Platform
Hello,
I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in
turn is talking to an Asterisk server via SIP for call origination and
termination. Seems simple enough, and it works for the most part,
but:
1) Caller ID name data comes in on the PRI, but doesn't appear to get
handed off to the Asterisk server via SIP, at least not in any
format that Asterisk
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
2006 Jun 12
0
ICLID or CNAM calling name and number through a cisco isdn gateway
All,
I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is....
One pri terminating into a Cisco 2431 router
Sip messages from the Cisco get sent to a asterisk server
linksys ata's a each remote end.
I can receive the calling name if the call originates
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,]
2004 Dec 01
3
Asterisk + AS5300
Is it possible to terminate calls via SIP on a Cisco AS5300? Did anyone do it? How? Do i need an special IOS version?
Ive been trying to compile the OpenH323 channel for the last month, but errors still happens.
Thanks in advance.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
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
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
---------------------------------
Meet your soulmate!
Yahoo! Asia presents Meetic - where millions of singles gather
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
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
2004 Apr 05
1
Extensions.conf sending calls to Cisco AS5300
I have my server configured to send to send all PSTN traffic to my Cisco
AS5300 gateway via SIP. I use the following line in the extensions.conf file
to accomplish this:
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@10.1.1.1,240,T)
Unfortunately, when I removed the T from the end of the statement, the calls
still complete, but they drop as soon as the called party answers the phone.
I thought
2009 Nov 07
1
Asterisk 1.6.1 + Cisco AS5300 + Fax T38 ?
Hi
I have finished the installation of my VoIP basic configuration ...
Actually:
- All calls from my E1 are received by a Cisco AS5300 and sent to my
Asterisk (in G711 by SIP).
- All user are connected by SIP to the Asterisk
- All calls from User are sent by asterisk to the Cisco AS5300
Now, i want see if i can supply T38 Fax Gateway ....
I am search to:
- Cisco Receive all