Displaying 20 results from an estimated 300 matches similar to: "ICLID or CNAM calling name and number through a cisco isdn gateway"
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
2008 Jun 20
1
Voice only works from one way.
Hello, everyone.
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to
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.
2007 Aug 29
0
Cisco FXS Issue...
Im sure this has been thrown around this list 1,000 times, and Im sure its been around the net too.. But I have done everything, and cannot seem to get inward calls to be processed on my asterisk box..
First, Let me tell you what works:
1) Softphones (ZoIPer using IAX2 Protocol) Can make calls behind a Natted Firewall to the FXS Port, and it rings, and calls work full duplex.
2) Soyo IP Phone
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
2003 May 13
4
app_transfer
I've added an important new application: app_transfer. This application
is designed to allow Asterisk to request the transfer of an incoming call
to a different extension. Consider the following diagram:
Caller -> [ PBX1 ] -> SIP or IAX2 -> [PBX2] -> Transfer App
A caller calls an extension on PBX1 which forwards to PBX2. PBX2 executes
app_transfer, which requests that hte
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
2008 Dec 29
0
SIP host=dynamic help needed for CCME
Hi,
I'm trying to get a remote Cisco Call Manager Express (CME) system behind
a dynamic IP address routing both inbound and outbound calls via SIP to my
local asterisk server. I've got a local CME system working fine on the LAN,
where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't
figure out how to get it working with host=dynamic, even locally on a test
2004 Nov 27
0
Failed to WWW-authenticate on INVITE
I'm having trouble connecting a asterisk server to a SIP Express router.
Inbound calls to my asterisk server works just fine, but when i try to
make outbound calls I get the following error message:
Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to
WWW-authenticate on INVITE to '"username"
<sip:username@mysipprovider>;tag=as5399a078'
I'm
2004 Apr 30
3
Asterisk <--> Cisco router
What codec should be used to connect a * box to
a cisco router which has a t1 with 24 trunks coming in?
My router voip dial plan looks like this:
dial-peer voice 2 voip
destination-pattern [1,2,,3,5,8]..
session protocol sipv2
session target ipv4:10.x.x.x
dtmf-relay cisco-rtp
codec g711ulaw
no vad
!
The problem I have is when more than one call is on it,
sometimes the quality gets very
2004 Jul 14
1
Questing regardning dialplans on a Cisco 5350
Hi.
If I use a Cisco as a PSTN termination GW and need to route all incoming
isdn calls to my asterisk and all outgoing calls from asterisk via the
cisco out to pstn, how do I do that ?
in the cisco I have this:
dial-peer voice 1 pots
destination-pattern [0-9]T
no digit-strip
direct-inward-dial
port 3/0:D
!
dial-peer voice 50 voip
destination-pattern [0-9]
voice-class codec 1
session
2003 Jun 08
1
anyone seen this error when running asterisk!
Hi all -
I'm making gradual progress implementing asterisk on my box! Now, when I
type asterisk it dies at this point. Does anyone have any idea why this is
happening! It have checked everything but running out of options!
[app_voicemail2.so] => (Comedian Mail (Voicemail System))
== Parsing '/etc/asterisk/voicemail.conf': Found
== Registered application 'VoiceMail2'
2004 Nov 30
5
cisco dial-peer voip
I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over
pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y
So far so good.
But I want to setup VOIP sessions with local carrier. I added dial-peer
40 for this. Session target x.x.x.x But calls will always get routed to
the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried.
My situation:
PSTN
2013 May 07
1
passing '302 moved temporarily' back to the SIP provider
Hello,
I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device
back to the SIP provider.
Here is the setup:
Some SIP phones are connected to an Asterisk System version 1.8.
External connection to the public network is also done via SIP to a VoIP provider.
Phone A has a CFW all calls to a phone number in public network (Mobile Phone)
incoming call to
2003 Nov 20
1
Cisco DTMF Issue
We're having an issue with connecting a Cisco ITS installation to * such that DTMF tones are passed to *. DTMF tones aren't passed to voicemail or to any of the interfaces behind *.
On the Cisco Side:
dial-peer voice 8 voip
destination-pattern 9999$
session protocol sipv2
session target ipv4:172.16.1.249
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
We have also
2004 Feb 02
6
Transfer
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
As I've been unable to get app_transfer to work, could someone explain how it
is supposed to work?
Currently I have two Asterisk boxes. A call comes in via zaptel to ast1. ast1
dials ast2 using iax2 and gets instructed to transfer the call to a different
extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing
happens
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
2011 Apr 01
1
The SIP channel driver - I'm giving up.
Friends,
After having spent many years working with the Asterisk SIP channel driver and the SIPv2 protocol, I have finally realized that this is a dead end. It's getting nowhere and it's way too complicated to set up, run and support in working code.
After realizing this, I started a new standardization project together with my friends in Canada, Simon and Marc, to develop a working
2005 Jul 18
0
why $cdr{'CALLERID'} and $cdr{'DNID'} are empty in perl agi connected with asterisk manager
hello perl experts
i am working with "ast-rad-acc.pl" from
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
i dont know why $cdr{'DNID'} and $cdr{'CALLERID'}
under 'sub send_acc {' are empty. i m successfully
connected with asterisk manager and when call i hangup
my perl application is getting that all other thing
are ok but i dont know why only
2004 Apr 07
1
PSTN calls do NOT hang up
Hi all,
In my Asterisk setup, incoming calls through Cisco PSTN gateway to Asterisk extensions sounds work fine. All calls can be terminated properly after hangup. However, when calls were forwarded to voicemail, after recording & hangup the PSTN calls and cisco FXO port remained connected unless cisco port was manually shut/no shut. # key used to hang up the call did NOT help. Did anyone