Displaying 20 results from an estimated 1100 matches similar to: "Small noise every 3 seconds"
2006 Nov 16
0
call from cisco router to asterisk gets auto attendant
Folks,
I have a NEC 2400 pbx(non-voip) behind a Cisco 3725, connected via
standard wic-t1 card. The NEC needs to call two different asterisk
servers with 4 digits. I have two way calling working with the one * box,
but the other is perplexing me.
Here's the layout
* <--> Cisco 2811(192.168.13.1) <--> 1.54 point to point <- Cisco
3725(192.168.8.1)<-> NEC 2400.
The
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
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
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
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.
2006 Oct 20
2
Clicking Noise on Pure Voip Calls
Setup:
Asterisk server in NY.
Cisco 7960 IP Phones in NY and London.
Dedicated T1 from NY to Ldn.
T1:
Latency - 100ms
Qos applied
No errors
Default codec on Ldn IP Phones = g711alaw
Default codec on NY IP Phones = g711ulaw
Both codecs allowed on each phone.
Issue:
Calls on IP Phones from NY to London hear clicking
noise on NY end.
Anyone experienced something similar or can offer some
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
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
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
2013 Apr 19
0
To enhance the voice quality of the SIP trunk
Hello;
I have a SIP trunk with a service provider, the caller from landline or mobile is hearing us very well, but the agent who is sitting on the handset is not hearing well, the voice at the agent is not crystal (like he is talking from well or far deep place). Although the IP Phones are cisco 7942G and the used codec is g711ulaw (actually it gave better quality than g711alaw).
If we increase
2003 Jul 08
0
codec problems with asterisk
We appear to be having a problem with our asterisk setup.
We have a cisco AS5300 with pri lines coming in and passing the calls onto
asterisk then too the sip phones.
the phone call from the sip phones (7960's) appears to be ok nice and clear
including the user who has called in.
but if your the user who has called in its all crackley sounds really bad
when they speak.
i believe this
2006 Mar 17
0
Critical Problem with asterisk
I am testing asterisk-1.2.1-15 on RedHat 9(i386) for SIP-to-SIP call and i found a strange problem.
When an extension gets a ring and it picks up the call a "tick" sound comes at start. This happens on both sides. I tried Xten's softphones and also hardphones.
A thing which was common in both(soft/hard phones) was the selected codec. When i used g711ulaw on both soft/hard phones
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,
2005 Feb 22
3
Call Manager Express Peer
I have the following configuration and am obviously missing something
small that is causing * not to work as expected.
I have the following defined in sip.conf
[ccme-in]
type=peer
host=10.0.9.1
context=devel_in
disallow=all
allow=alaw
nat=no
canreinvite=yes
qualify=yes
and [devel_in] is defined in extentions.conf
However when I try to call via the dial peer I have configured on the
cisco
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
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 Dec 16
0
FW: Cisco 7960 (SIP) hold problems
ala cisco 7960
-----Original Message-----
From: Matt Schulte
Sent: Thursday, December 16, 2004 10:34 AM
To: 'Paul A Brown'
Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems
Sure thing, the biggest problem I had was getting the SIP filenames
working correctly for updating the firmware (blah, I love Cisco but
these phones are a joke for support). This works for me! Good luck.
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
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 Jul 26
2
7960 SIP Firmware Upgrade Strange Problem
Hi,
I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go
up to 7.5
However in my first attempt to go from V.5.1 to 6.0 this is hat happens:
- The phone reboots
- The phone then reads the file OS79XX.TXT from the TFP server. In the file
I added the version "P0S3-06-0-00"
- It starts upgrading firmware
- Then I get the following message: (Upgrade Failed -