Displaying 20 results from an estimated 2000 matches similar to: "Asterisk and Cisco Call Manager Express (CME)"
2009 May 20
3
Asterisk CCM, CME Integration
Hi All,
I'm just posting this questions to both forums as its related to both. In
hope to get some help on below issue:
Asterisk 1.4.x
CCM = 4.x
CME = 4.x
codec = g711ulaw
Here is my setup:
600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME
-----> 461X Phones
461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X
Phones
so in
2008 Oct 15
1
Cisco 7960 not always receiving incoming calls
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone will receive the next inbound call. The phone
will not receive another inbound call until a call out again from it
first. Any ideas?
I am using SIP and am using the latest
2008 Oct 19
6
adding a second extension
I'm trying to add a second extension to my setup. The second device is
able to successfully connect to the Asterisk server. I am unable to
contact extension 101 from 102 and vise-versa. Also are my context
setup logically or is there a better fashion to organize them? My
error is at the bottom.
Here is the extension.conf
[default]
;
; By default we include the demo. In a production system,
2006 May 31
9
Unable to use 'valid users' from Active Directory
I am able to return users and groups using wbinfo -g and -u. Samaba will
even allow users to connect that are in our domain. The problem exist
while trying to narrow down permissions to a share.
[public]
comment = Public Stuff
path = /home/
public = yes
read only = no
valid users = @"UFAD\_IFAS-FRE-USERS_autoGS"
This does not work. It prompts the end user for a
2008 Oct 07
2
Cisco 7906g & SIP
Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk
1.2.26.
I have uploaded in my tftp server the firmware
'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in
SEPmacaddress.cnf.xml I have:
<loadInformation>SIP11.8-0-4SR1S</loadInformation>
..but in tftp log server I have:
Oct 07 11:56:22 asterisk1.local
2010 Oct 24
5
Integrating Asterisk 1.8 with Google Talk and Google Voice
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
2008 Oct 04
5
Vitelity Asterisk configuration help
I have a Asterisk server setup and I am able to connect to the server
using a soft client 'x-lite' and call and leave a message on my second
extension 102. I have setup a Vitelity account and add what I believe
to be the correct information to my sip.conf and extension.conf. I
would like to setup incoming and outgoing calls with voicemail
support. I've searched all over but many of the
2008 Oct 19
2
Latency woes, qos the fix?
My latency is kind of high and the voice delay is noticeable.
The Asterisk server is on a dedicated host outside of the network. I
am performing PAT/NAT using a Cisco router.
ns1*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored
vitel-inbound/rsreese 64.2.142.116
2005 Jul 26
0
RE: VM on * for CME Install - Solved
I found with some more testing that you have to setup a 5 digit number (or
something longer than your phone extensions) to make the voicemail work.
Now the trick is making the MWI work.
Rick
-----Original Message-----
From: Lull, Rick
Sent: Friday, July 15, 2005 3:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: VM on * for CME Install
Hi folks-
I've
2007 Feb 14
0
Asterisk & CME integration using h323
Hi all, I'm trying to trunk my asterisk with a cisco cme 3.3 using h323.
Cisco conf:
dial-peer voice 8 voip
destination-pattern 2...
session target ipv4:<asterisk ip>
codec g711alaw
no vad
h323.conf
[general]
port = 1720
bindaddr = 0.0.0.0
;tos=lowdelay
;
disallow=all
allow=alaw
allow=ulaw
allow=gsm
context=from-internal
extension.conf
[from-internal]
exten =>
2006 Mar 01
1
Cisco Callmanager integration with asterisk
Hello
We have integrated cisco callmanager 4.1 with asterisk and we can dial from
cisco to asterisk but we're getting an error if we call from asterisk to
callmanager. This is the error I'm getting
anybody can help me?
Verbosity is at least 3
-- Executing Dial("SIP/2234-e084", "SIP/cme-pbx/4455") in new stack
-- Called cme-pbx/4455
-- SIP/cme-pbx-25ae is
2005 Jul 15
0
VM on * for CME Install
Hi folks-
I've got to the point of trying to configure voice mail on the * box
for the SCCP/CME phones. The phone can call the voicemail number (8500) and
I can hear Allison's voice. Attempts to punch in a voicemail box number or
password don't seem to register; keypad presses don't seem to be heard by
the * box. The CME configuration has the 'dtmf-relay rtp-nte' command
2005 Jun 13
1
about timeouts
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi folks,
I've this infrastructure:
|voip services| -- |*| -- |cme| -- |isdn|
the voip services are logged on my *, then forwarded to number 601 on
cme. The isdn calls too are forwarded to 601. On cme I've a timeout X
for call-forward noan (no answer) to a specific number on * (5901) that
is my x-lite software client. If 5901 is
2005 Jul 14
0
Cisco CME Integration - IOS Version known to work?
Hi folks-
I'm working on getting a test Call Manager Express system working
with Asterisk. My plan is to have * support all the voicemail boxes for the
CME/SCCP phones.
Right now, I can call from a SIP phone to a SCCP phone and back
fine. Calls go from Phone->CME->*->Phone and the reverse. Voicemail works
for my SIP phones, but does not work for the SCCP phones.
I tried to follow
2005 Mar 16
2
[Possible SPAM] : about sip, asterisk and cisco ccme
I am starting to work on a similar solution, but with full call manager
rather than CME. I am going to use Asterisk to accept POTS calls
through PCI FXO ports (winmodems) and then forward the calls through to
call manager via SIP. I don't have my FXO cards yet (waiting for UPS
man!!) but I have * talking to the CM through SIP just fine. I am
testing with the Cisco softphone, connected as a
2009 Jun 05
1
DTMF Problem w/ MeetMe
First, the scenarios:
Call placed from Boston to locally configured Asterisk Meetme extension:
Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Asterisk(Boston)
Call placed from Boston to European Asterisk Meetme extension:
Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco
2821(CME,Europe) <-SIP-> Asterisk(Boston)
In the 1st scenario, everything works
2009 Feb 19
0
Asterisk BLF to Cisco CME
Hi all, I'm searching for a way to inform my Cisco CME that a number on
Asterisk server is busy.
I have a SIP trunk between Cisco and Asterisk and some Cisco ip phone
have a speed dial with a number registered on Asterisk.
How can I exchange busy information between two PBX?
Thanks Enrico.
2010 Dec 22
1
Simplifying dial-plan
Is there a way to include:
_NXXNXXXXXX
_NXXXXXX
_011.
_911
into my current plan:
2005 Jan 12
2
Call Manager or Asterisk
Hello list.
No intention to start a flamewar here but I would really like opinions
from those who know both the Cisco and Asterisk system. I'm working for
a company with 15 offices in 11 countries, offices are relatively small
(3-20 people each) and most of them have a Cisco 1760 Router installed
with Call manager express (CME) and 1-3 ISDN lines (2-6 simultaneous calls).
We
2009 Jan 07
1
CISCO 7940 United_States/7960-tones.xml
I have a smartnet contract for this phone, and have searched high and
low for this file on the Cisco website.
I need:
United_States/7960-tones.xml
English_United_States/7960-font.xml
Every road seems to lead to the Call manager express downloads... I
don't have a CME, so that's basically useles.
Can anyone point me in the right direction?
Mikel