Displaying 20 results from an estimated 20000 matches similar to: "incoming calls on Cisco 7960"
2005 Jun 16
1
Cisco 7960 (SIP) with Asterisk: how to get # to work during a call
Gents,
I've built an Asterisk system to replace our PBX at work and have Cisco
7960 phones (SIP 7.4) running with Asterisk 1.0.7.
How to I get Asterisk to recognise the '#' being pressed during a call?
In sip.conf I have entries likle this:
[2001]
type=friend
context=local-phone
auth=md5
username=2001
secret=xyzzy
callerid=Jack Tubby <2001>
2006 Feb 15
2
Hint priority
Hi All
Has anyone managed to get the hint priority with Swissvoice IP10S phones
working?
I have 2 phones: a Snom 360, setup as the reception phone on extension
11, and a Swissvoice IP10S on extension 12.
When calling each other (tested both ways) I can only ever see the Snom
360 in the Active State from 'show hints'. The Swissvoice stubbornly
remains in the Idle State when on a call!
2004 Jul 29
2
Zultys Zip 4x4
Is anyone successfully using one of these with Asterisk? I cannot get the
phone to register, this message keeps coming up on the Asterisk console:
Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request:
Registration from '"000BEA801CA6" <sip:000BEA801CA6@hcs.net:5060>' failed
for '204.194.36.138'
The telephone LCD says "SIP registation
2004 Sep 21
1
Cisco 7940/7960 and voicemailmain not able to press keys after a hold.
I have noticed a problem with the Cisco 7940/7960 phones where if
you put your voice-mail box on hold using soft keys and come back
you can no longer navigate. I am curious if anyone else can
duplicate this problem. Happens reliably for me with the 7940
phones.
I use rfc2833 for DTMF. I would think it was a Cisco bug, but
for the fact that this did not happen with older version of
2008 Mar 17
1
update_call_counter: Call to peer '2509' rejected due to usage limit of 1?
Hi,
I am using asterisk-1.4.15, My sip configs is like
[2501]
type=friend
username=2501
secret=2501
canreinvite=no
host=dynamic
dtmfmode=rfc2833
context = sip
disallow=all
allow=ulaw
incominglimit=1
nat=1
queue.conf is like
[gen-enq]
joinempty = yes
musiconhold = default
strategy = rrmemory
servicelevel = 60
timeout = 60
retry = 5
wrapuptime=5
announce-frequency = 90
announce-holdtime = yes
2010 May 12
2
asterisk-users Digest, Vol 70, Issue 25
Hi Vardan
I did same as you told and deleted the SIP information in Astdb and
restarted asterisk. but the result was same.
as you said there might be mistake in sip.conf so i am pasting both servers
configuration here..
1- nasir.server.com
[abc]
username=abc
type=friend
secret=mysecret
nat=yes
mailbox=12234568
incominglimit=2
outgoinglimit=2
host=dynamic
dtmfmode=rfc2833
context=payasyougo
2004 Jul 18
4
Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi All
Total noob on the list so all help appreciated....
I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows).
I've plugged in two Cisco 7960 phones....
The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......
2007 Aug 31
1
Cisco 7960 Won'
I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2)
If the call that I'm trying to make goes through, everything works fine. But if
there's any sort of error (like me messing around in my extensions.conf, etc). I
can't get the connection to drop. ie: If I get the conjestion tone and hang up
the phone, I can do a sccp show channels I can see that the
2004 Apr 20
3
Limiting incoming SIP calls & Original CallerID on transfer
I have 2 issues which I need to resolve on our production Asterisk
server:
We are currently using Polycom IP600 VOIP phones for our office which
are capable of handling 2 calls per SIP registration. What we're finding
is when staff are on the phone, Asterisk will pass them a second call
which will show up on their display, and an audible beep is heard over
the phone (regular call waiting). I
2004 Sep 21
3
Uniden uip200
I got a Uniden UIP200 and started to configure it and I am lost....
I have a tftp server setup on my * server and have the files unidencom.txt
and uniden<mac>.txt there. But it doesn't quite work yet. It registers as
a sip phone (sip show peers), but I cann't dial it and the display shows #1
disconnected all the time. It has firmware version BS4.59a in it.
I have no idea if I
2008 Feb 08
1
(no subject)
Hi,
I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also.
But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized
2003 Nov 28
4
call waiting disable in sip
Hello,
is there a way to disable call waiting in sip? I`m using grandstream 101
and even when the phone is in use I hear ringing in the headset. It is
pretty annoying , is there some way to disable this? I cant find
anything like it in the grandstream docs.
Thanks
--
Anton Yurchenko<phila@dg.net.ua>
Digital Generation
2006 Apr 12
1
Cisco 7960 won't dial (sccp)
I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk
working fine for sip clients, and can call the 7960's just fine, but
I can't seem to dial out on them.
As soon as I enter the first digit, the phone attempts to dial it without
waiting for the rest. I've changed timeout settings, etc but can't seem to
get it to work. Any ideas?
Asterisk
2009 Jul 28
3
CIsco 7960 + asterisk: hepl needed
Dear All,
I'm trying to configure my new phone Cisco 7960 to work with asterisk.
I followed http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html
and I got into the point where I can see on the the display line indication showing
"55 <phone icon with x>" so it looks like the phone is not registered.
The phone and the asterisk are in the same local
2007 Jun 20
1
DTMF doesn't work between Asterisk and Cisco SIP Proxy
Hi buddies,
I encountered DTMF issue when I tried to place call from x-lite to a
sip conference serice,here is the diagram.
X-lite---->Asterisk--->Cisco SIP proxy---->SIP Conference service
The Call can be established,and I can hear from x-lite the prompt of
the conference,but when I input any digits,nothing happened,the
conference service did not recognize my input.At the same time,in
2004 May 04
1
Asterisk and windows h.323 gatekeeper calling problems...
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi there, i have a working Microsoft ISA firewall with buildin H.323
Gatekeeper....
So Far, i got registerd the asterisk on the M$ Gatekeeper...
here is the h.323 configuration:
; Open H.323 driver configuration
;
[general]
port = 1720
bindaddr = 0.0.0.0
allow=all ; turns on all installed codecs
dtmfmode=rfc2833
gatekeeper =
2005 Jan 25
5
Polycom and call waiting again..
I searched and read all the relevant posts, but I still don't have a
solution to my problem..
I've got a small queue for tech support calls using AddQueueMember. The
agents are using IP300's from polycom.
In my example, only one agent is logged int.
When a call comes into the queue, asterisk sends the call to the one agent
logged in. The agent answers and is talking to the
2004 Apr 08
1
Two operators, 10 rollover lines, Cisco 7960G chanisavail problem
Here's my situation.
I have two receptionists that answer incoming lines. Each has a 7960G with
5 incoming lines each. I'm trying to set this up so each line on each phone
doesn't utilize call waiting. My problem seems to be that
ChanisAvail(Sip/cisco1&Sip/cisco2&Sip/cisco3&Sip/cisco4&Sip/cisco5) always
returns cisco1.
Here are the sip.conf entries: (mind you,
2005 Sep 21
1
Call getting disconnected in queue
Hi,
I have a small call center with 4 Zap lines and 4 agents. Agents login
using sip phones with AgentCallbackLogin. I occasionally gets a
complaint that when customers call the call center, after the initial
greeting is over the call gets cut after playing the thank you message.
I started investigating and found that that happens when the call gets
transferred to an agent who is making an
2005 Feb 07
3
incoming calls in h323 do not come to right dialplan
Hello,
I am moving topic from asterisk-dev list to asterisk-users list. Did anyone
succeed receive incoming calls in h323 and orient them to right context based
on "host" identification?
To summarise, I have quintum Gateway sending call to Asterisk box, and I would
like to use asterisk as a protocol converter h323 --> sip.
in h323.conf, I have
[quintum_gw1]
type=user