Displaying 20 results from an estimated 200 matches similar to: "Cisco 7200 One-Way Audio"
2003 Apr 30
5
PRI Setup
Heh guys, I just received a T400 card, I've been using a T100 for a
little while, and it works fine when using a raw channelized T1. I'm
relocating my asterisk machine, and PRI's will only be available,
haven't found any good config info for PRI's, can someone point me to
PRI config info, or let me know what changes I need to make in order to
bring them up, I imagine,
2008 Jul 25
0
Slightly Off Topic: Cisco & Premisys Slimline
Has anyone got a Premisys Slimline channel bank working with a Cisco
AS5400 or similar?
I'm not sure if my unit is bad, or what. I'm using FXS Loop Start.
Calling the port connects immediately without ringing the attached
phone. If I pick up the phone, it's connected and I can talk to the
caller. Hanging up has no effect. I can see the bit transitions (0101
to 1111 when I go
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
2004 Nov 29
1
Cisco gateway help needed
HI,
I have been pulling my hair out trying to get a Cisco MC3810 to interface my
Asterisk box with a T1.
I am able to make outgoing calls but incoing calls never reach my Asterisk
box. The cisco give a fast busy when I try to call one of the DID's. When
playing around with the dial-peers I can get the cisco to pick up the call,
but then it forwards the call back to the ANI that is dialing.
2005 Oct 02
1
Adit 600 FXO card sound quality
I have an adit 600 with one fxo card connected to a Digium single span T1 card.
CallerID, disconnect supervision work perfect, however the users
complain that they have some sound quality issues, after testing it I
realized that whenever one is in a phone call they get like silence
between the sounds coming from the other party, almost like a cell
phone, in other words if there is no sound coming
2004 Jul 31
0
Trunk doesn't work Adit 600/T100P
Hi !
I am connecting to Adit 600 thru a T100P card
I have configured 1-16 FXS channels and 17-24 FXO.
Everything looks fine on Asterisk side I get a tone
on all FXS channels, but when I try to dialout thru
one of the FXO channels 17-24 it doesn't connect to
the POTS line and echoes back my voice.
I use fxsls and fxols for the T1 channels and ls on
Adit side. Whats wrong here ?
here is my Adit
2005 Oct 13
2
Sample cisco config for cisco 7206
I see a lot of comments but no actual show runs.
Can someone post a 7206 config.
I am having a dickens of a time getting calls to pass.
I currently have the following loaded.
Cisco IOS Software, 7200 Software (C7200-IK9O3S-M), Version 12.3(8)T6,
RELEASE SOFTWARE (fc2)
Thanks !!!
Jerry
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2004 Jan 17
1
Channel Bank Woes...
Hello all,
I have a new Carrier Access Adit 600 and am having a problem dialing out
and receiving incoming calls.
After the initial ring, the calling party will here nothing. There is a
silence, but the SIP phones that is supposed to ring will ring. The
calling party will not hear the sound of the ring that * sends out on
the line. Additionally, if someone picks up the receiver there is no
2016 Oct 04
0
The security id structure is invalid
On 10/4/16 2:40 PM, Rowland Penny via samba wrote:
> On Tue, 4 Oct 2016 14:00:02 -0400
> Ron GarcĂa-Vidal via samba <samba at lists.samba.org> wrote:
>
>> ERROR: incorrect GUID component for member in object CN=Domain
>> Admins,CN=Users,DC=dc1,DC=mydomain,DC=net -
>>
2005 May 12
1
realtime sip show peers no nat
Hello
sip show peers does not mark hosts as NAT even though sip.conf and
sip_peers table has nat=yes.
spitfire*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask
Port Status
voipuser.org/gdsm 216.127.66.119 N 255.255.255.255
5060 Unmonitored
5560/5560 192.168.4.5 D N A 255.255.255.255
5060
2005 Oct 03
2
asterisk, cisco 3640's and DIDs...
I would think I could do this but for some reason I am stymied.
I have a PRI from RCN connected to a cisco 3640 (in my day "cisco" is
all lower case :-)). My config looks something like this on the cisco...
---------------------------------------------------------
voice-card 3
dsp services dspfarm
!
ip cef
!
isdn switch-type primary-5ess
!
controller T1 3/0
framing esf
linecode
2016 Oct 05
0
The security id structure is invalid
Here is some more information that could be helpful. This is the entry
for LDAP User in ldbedit:
# record 253
dn: CN=LDAP User,CN=Users,DC=dc1,DC=mydomain,DC=net
objectClass: top
objectClass: person
objectClass: organizationalPerson
objectClass: user
cn: LDAP User
sn: User
givenName: LDAP
instanceType: 4
whenCreated: 20140106220805.0Z
displayName: LDAP User
uSNCreated: 6218
name: LDAP User
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
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 May 19
0
Re: Asterisk-Users Digest, Vol 10, Issue 154
Send Asterisk-Users mailing list submissions to
asterisk-users@lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
asterisk-users-request@lists.digium.com
You can reach the person managing the list at
asterisk-users-owner@lists.digium.com
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
2004 Dec 01
4
Getting started with Asterisk
Hello ,
I'll just started with asterisk and I would liket to to dial between your
two phones with to cisco ATA 186 , but I have a problem
The two cisco ATA and the server in the same networks and i have the ring in
the phone but i'am not able to place a call
Between the twe phone .
In attachement the sip.conf and a log file
Any suggestement .
Regards
RAbii
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 Aug 27
0
Bad hangup event cause
Hello, i have a problem with the hangup cause received from the AMI in the Hangup events. All calls that arent answered after ringing are returning hangup cause 16 (normal clearing) instead 19.
Im running asterisk 1.4.11, the calls are generated to a SIP peer using the AMI originate command.
This is the 'sip debug' output:
Reliably Transmitting (no NAT) to 192.168.0.70:5060:
INVITE sip:1
2004 Oct 06
2
Cisco router for PRI termination?
If you have a PRI terminated in a Cisco router talking SIP to * and would
be willing to share your Cisco config, please respond. Also, I would be
interested in knowing what version of IOS you are using. We are using an
NM-HDV in a 3640.
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815