Hi,
I'm unable to configure SS7 (surely my bad because it's my first try).
I get this error:
ERROR[15475] chan_dahdi.c: Unknown signalling method 'ss7'
My system has:
asterisk 1.4.31
zaptel 1.4.12.1
libpri 1.4.11.5
libss7 1.0.1
(installed from source)
I can't upgrade this server to Dahdi and
2011/12/7, Jamie A. Stapleton jstaple...@computer-business.com:
Yes, we are using it. Most of the docs on the Internet are for 1.4.
However, we now have it working with 1.8 (after some work).
What do you imply by after some work ?
Did you have to modify asterisk code ? asterisk config ? both ?
On 12/07/2011 06:15 AM, Vieri wrote:
I can't upgrade this server to Dahdi and latest asterisk version...
In any case, according to the libss7 README, it should work with my software
versions.
What makes you think that? There is no support for SS7 in Asterisk 1.4.
--
Kevin P. Fleming
Digium,
I spoke with the Asterisk Pre-sales team and they said that SS7
support isn't technically supported, but it is there (e.g. talk to the
OS community about this) so here's my question:
I'm trying to interface an Asterisk Softswitch to a Nortel DMS100.
If I get a dual-span card, can I run SS7
I am using AMI to call a phone and play a wave file. That works fine
to SIP/401.
Now I am trying to redirect that call that is ringing to another phone
(SIP/404).
When I do it the other phone rings but the first phone continues to ring
also.
Then when I answer on SIP/404, I get a ring not
- Original Message -
I spoke with the Asterisk Pre-sales team and they said that SS7
support isn't technically supported, but it is there (e.g. talk to the
OS community about this) so here's my question:
I'm trying to interface an Asterisk Softswitch to a Nortel DMS100.
If I get
[Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:821 pgsql_reconnect:
Postgresql RealTime: Everything is fine.
[Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:201 realtime_pgsql:
Postgresql RealTime: Result=0x12e4edd0 Query: SELECT * FROM sippeers
WHERE name = '105680' AND host = 'dynamic'
[Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer
'105680' is trying to register, but not configured as host=dynamic
Going on this, I'd say you probably tried to specify the host with a
static IP address or a host name. If that's the case, you can't
register, because that
On Tue, Dec 6, 2011 at 4:05 PM, white hat whitehat...@gmail.com wrote:
Would you be willing to post sanitized versions of your jabber.conf,
gtalk.conf and details regarding the context you're using and how your
inbound route is configured in your dial plan?
Are you using STUN? Is Asterisk
No secrets :)
SELECT * FROM sippeers WHERE name = '105680' AND host = 'dynamic'
name|type|username|secret|fromuser|fromdomain|nat|context|canreinvite|disallow|allow|host|insecure|port|ipaddr|outboundproxy
I am running an Asterisk 1.4.34 installation. I am trying to separate
several SIP phones into two separate spygroups. These phones are making
external calls as opposed to receiving incoming calls. Is there a place
to assign a phone to a Spygroup other than when the call is initiated.
I am
Just a thought - make your normal phone use context default and your
others use context spyonme
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
jeremy.hellst...@synovate.com
Sent: Wednesday, December 07, 2011 2:11 PM
To:
I am running Asterisk 1.6.2.20 with dahdi 2.5.0.2, Oslec from kernel
sources. Hardware is 2 X100P Wildcards. Everything seems to be working
OK but my logs are filling up with this message:
Dec 7 14:25:06 servername kernel: FXO PCI Master abort
The messages just pour in constantly until the
Check this post - it sounds like exactly what is happening to you.
http://osdir.com/ml/debian.packages.voip.devel/2009-01/msg00046.html
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson
Sent:
On 12/7/2011 2:35 PM, Danny Nicholas wrote:
Check this post - it sounds like exactly what is happening to you.
http://osdir.com/ml/debian.packages.voip.devel/2009-01/msg00046.html
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Wed, Dec 07, 2011 at 02:46:51PM -0600, Brent Davidson wrote:
On 12/7/2011 2:35 PM, Danny Nicholas wrote:
Check this post - it sounds like exactly what is happening to you.
http://osdir.com/ml/debian.packages.voip.devel/2009-01/msg00046.html
Yes, that appears to be what is happening to
Hi,
A telco has recently installed a new line in our building and I need to connect
it to my Asterisk server with a Digium PRI card.
It's not the first time I set up and configure a PRI link but I'm failing to
make this one work.
The only information I got from the telco is:
Line Coding
On Wed, 7 Dec 2011, Vieri wrote:
A telco has recently installed a new line in our building and I need to
connect it to my Asterisk server with a Digium PRI card.
It's not the first time I set up and configure a PRI link but I'm
failing to make this one work.
chan_dahdi.c: No D-channels
On 12/07/2011 04:15 PM, Steve Edwards wrote:
On Wed, 7 Dec 2011, Vieri wrote:
A telco has recently installed a new line in our building and I need
to connect it to my Asterisk server with a Digium PRI card.
It's not the first time I set up and configure a PRI link but I'm
failing to make this
--- On Wed, 12/7/11, Steve Edwards asterisk@sedwards.com wrote:
A telco has recently installed a new line in our
building and I need to connect it to my Asterisk server with
a Digium PRI card.
It's not the first time I set up and configure a PRI
link but I'm failing to make this
--- On Wed, 12/7/11, Kevin P. Fleming kpflem...@digium.com wrote:
Vieri: You aren't even far enough along to worry about
D-channel assignments or anything like that. Your span is in
RED alarm; that means it can't see the far end at all. Until
you get that cured (layer 1 - physical layer)
On 12/07/2011 04:51 PM, Vieri wrote:
--- On Wed, 12/7/11, Kevin P. Flemingkpflem...@digium.com wrote:
Vieri: You aren't even far enough along to worry about
D-channel assignments or anything like that. Your span is in
RED alarm; that means it can't see the far end at all. Until
you get that
and maybe more but right now I don't recall any loopback device although I
won't be sure until I go to the site.
Can a loopback device be bought seperately?
Sure, we use the below device all the time:
http://www.amazon.com/Cables-Unlimited-TST-LOOP-003-Smartronix-Superlooper/dp/B000V6Y7IY
Hi,
I am making confrence application.
In sip.conf
[phone1]
type=friend
host=dynamic
Takes an alphanumeric string.
context= employees
[phone2]
type=friend
host=dynamic
context= employees
[phone3]
type=friend
host=dynamic
context= employees
In extension.conf
[employees]
On Thu, 8 Dec 2011, Durgesh Mishra wrote:
I am making confrence application.
[Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No
application 'MeetMe' for extension (employees, 777,
1)
You don't have app_meetme.so loaded.
What happens if you enter 'module load
I check in CLI
[Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No
application 'MeetMe' for extension (employees, 777, 1)
== Spawn extension (employees, 777, 1) exited non-zero on
'SIP/phone1-'
Plz tell me , where i am wrong in configuration.
Chances are you
Hello List,
Can anyone please confirm whether Asterisk can be installed in Oracle
Enterprise Linux6 OS.
Thanks,
Rekha
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Hi
This is due to module app_meetme.so is not loaded.
Execute below command in asterisk cli and check the cli logger.
module load app_meetme.so
If you are installed asterisk in a linux system without any analog
interface this meetme application will not work. You have use
application
In make menuselect =application=XXX app_meetme . I am doing confrence call
using sip softphone.
I checked It Depends on: dahdi(E) .
How I can do app_meetme enable?
Thanks
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal
Sent: Tuesday, December 06, 2011 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Talk detection in meetme
Hi,
I create Chat
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