Has anyone else had problems with the Page() application not working
under Asterisk 1.2.3?
We use Cisco 7960 phones and set one of the lines to auto answer. When
someone dials the paging extension it calls the page app and invites all
the lines on the phones that are set to auto answer into a
Does anyone know what application I should place this function in? For
example with the DB function I currently do something like this to add
an entry to the asterisk database:
exten = s,n,Set(DB(AGENT/${MACRO_EXTEN:1})=${CALLERID(num)})
To delete the entries I do something like this:
exten
I'm having trouble building res_snmp (under Asterisk 1.4.0) on a box
running Debian Sarge. res_snmp says its dependencies are netsnmp but
Debian doesn't seem to have a netsnmp package. I've tried installing
pretty much every package available related to snmp and no luck. I'm
just wondering if
I've downloaded:
asterisk-1.4.1
zaptel-1.4.0
I've compiled and installed zaptel. When I go to install asterisk I do:
./configure
make menuselect
I then take a look under the codec selection menu and I see that
codec_zap can not be compiled.
Trying to use:
Asterisk 1.4.2
Zaptel 1.4.0
chan_zap won't compile in asterisk 1.4.2 when used with zaptel 1.4.0.
The changelog has this entry:
* channels/chan_zap.c, configure, configure.ac: If we receive
ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256,
For those of you out there with SPA-942s in production, do any of you
have issue with the sound quality when using g.729 as the preferred
codec? We are noticing terrible sound quality on the other end of a call
made to a spa-942. Everything sounds crystal clear at the SPA itself but
sounds
Does anyone have this working? I have a Cisco 7970 with the 8-0-2-SR1S
firmware loaded on it. I can get the phone to register with * just fine
when I place my asterisk server on the same subnet and do no NAT. When I
give my asterisk server a static public IP and put the phone behind a
NAT to
to make the 7970 phones accept
SIP responses back to the originating port. I wasted several hours but
couldn't figure it out.
-Evan
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Rock River InternetJeremiah Millay
202 W. State St, 8th Floor
I tried upgrading a used Cisco 7970 from the image it shipped with to
SIP 8.0.2 SR1 but didn't have any luck so I followed the procedures to
do a factory reset on the phone. The phone is grabbing an IP and
attempting to grab my term70.default.loads file but not moving any
further. The phone
You need to login as your extension number at your voicemail context.
Find out what context your mailbox is in by checking voicemail.conf.
Should be between the brackets.
For exmaple:
In voicemail.conf:
[mycontext]
1234 = 4321,Joe Example,[EMAIL PROTECTED]
You would log in to the comedian
I'm running SIP between my Lucent TNT acting as a gateway, and an
asterisk server. We have a PRI coming into the Lucent. Basically the
problem I'm having is mostly on inbound calls but some outbound calls as
well. I hear echo and sometimes some weird artifacting on calls coming
in from the
stopped using the tnt.
Carlos Alcantar
Race Technologies, Inc.
101 Haskins Way
South San Francisco, CA 94080
P: 650.246.8900
F: 650.246.8901
E: carlos at race.com
-Original Message-
From: Jeremiah Millay [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 28, 2005 2:50 PM
To: asterisk-users
: text/plain
On Wed, 2005-06-29 at 09:18 -0500, Jeremiah Millay wrote:
No I do not hear any clicking sound. Some calls come in perfect, and others
come in with some echo and sometimes artifacts, which I think might be caused
by jitter. Also it is mostly inbound calls that I have the problem
Does anyone know if it is possible to send variables over an IAX trunk?
Is there a setting in iax.conf that allows this or is there another hack
to allow this?
Thanks in advance.
Jeremiah
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Rock River InternetJeremiah Millay
202 W. State St, 8th Floor [EMAIL PROTECTED]
Rockford, IL 61101 815-968-9888 Ext. 2202
USA fax 968-6888
I'm trying to change the ring tone on my 7960 from the dialplan. I've
tried the example on the wiki but it doesn't seem to work. Something like:
exten = 3010,1,SetVar(ALERT_INFO=Bellcore-dr1) ; selects Ringer
exten = 3010,2,Dial(SIP/3010,15)
I'm not sure what the Bellcore-dr1 ringer is
)
+dirintro = dir-intro-fnln;
else
dirintro = dir-intro-fn;
}
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Rock River InternetJeremiah Millay
202 W. State St, 8th Floor [EMAIL PROTECTED]
Rockford, IL 61101
I ran into this same problem the other day. What you need to do is put all firmware files in the tftp root directory. The trick with the files is you need to match the case of the filename that the phone is looking for. My XmlDefault.cnf.xml needed to have the proper case. If you do a tcpdump on
I'm using chan_sccp module. The phone is using the latest firmware from Cisco. Pretty much everything works great. Somethings that aren't working, are call conferencing, customizable ring tone list, etc. Also I'm not really liking how the sccp channel works from the asterisk command line. Its not
We are successfully using Lucent MAX TNT with Asterisk. Config is essentially the same as the one found on voip-info wiki. Just do a google on asterisk lucent tnt, and it should be one of the first pages to pop up. We run our PRIs into the TNT, then talk SIP from the TNT to our asterisk server.
What version firmware are you running on your Cisco Phones? We are running
Asterisk 1.2 with the 7.4 firmware. The latest is 7.5 but there are some
strange things that happen with this firmware. If I were you I would try a
different firmware on the phones. Hope this helps.
Jeremiah
Help!
.
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Rock River InternetJeremiah Millay
202 W. State St, 8th Floor [EMAIL PROTECTED]
Rockford, IL 61101 815-968-9888 Ext. 2202
USA fax 968-6888
I know this has been added to SVN but I'm looking for the source for the
original module. It used to be at http://redice.krisk.org/ but this page
no longer seems to display anything. I'd like to add it to my 1.2.13
stable install. Does anyone have a copy of the original? I used to have
this
I'm using a mix of Cisco 7960, Linksys SPA-942, Cisco 7961, Cisco 7970
phones in a paging group. I have all the phones set up with an extra
line that auto answers the dial from my paging extension when the
primary line is not in use. All of these are operating correctly however
the 7961/7970s
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