Displaying 20 results from an estimated 1400 matches similar to: "built in dial functions?"
2003 Sep 08
2
*78 *72 and sip?
I know *8 kind of works with SIP
but what about the rest should they work
do they work with a zap device?
*0# sends flash
*8# remote call pickup (pickup phone in your group)
*67# disable caller id
*70# no call waiting
*78# do not disturb on
*79# do not disturb off
*72# enable call forwarding
*73# disable call forwarding
*82# enable callerid
2003 Sep 17
3
documentation?
Been learning * now for a couple of weeks and have all basic features
running including VM, MoH, FX lines, iaxtel, and FWD. However, I seem to be
lacking documentation on a lot of technical things and am wondering if I
overlooked something that is obvious to others. (I do have the Handbook,
have been doing a fair amount of google searches, and read the README.*
files.)
Examples,
Where should I
2004 Jan 31
4
rtp sound quality?
pstn -> sip gw -> * -> C7960
When I dial into * via the pstn, I hear the ivr menu just fine (good
quality). I press 3000 (valid extn), and I begin to hear ringing however the
ring back is very very choppy.
I answer the C7960, and speech is clear in both directions. Place the C7960
extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates
both the sip gw
2004 Jun 18
2
C7960 g729 question
I have multiple voiceage g729 licenses installed on a RH9 box, and have
a remote C7960 configured to use it (low bandwidth). In calls like:
Remote C7960 -> g729 -> asterisk -> g711 -> C7960
the audio is oftentimes rather choppy. Changing the remote 7960 to use
g711 seems to eliminate/reduce the choppyness. Any ideas on what might
be behind this?
2006 May 24
2
DHCP configuration for Cisco 7960?
(Apologies to those Toronto Asterisk Users' Group folks who have seen
this message... I figured I'd have more success with a wider audience)
I'm trying to boot a Cisco 7960 from an ISC DHCPD server (3.0.3 on
FreeBSD 4.11), so far unsuccessful, and getting some odd behaviour on
the wire. I wonder if anyone has done this before and therefore can
validate whether or not the traffic I am
2005 Oct 07
1
Distorted VM with iax2 with ilbc and jitterbuffer - bug?
Two asterisk boxes 150 miles apart, both cvs-head as of this morning
(and since Sept 27th), connected via iax2 with low-utilized ds3 internet,
C7960 calls exten on remote system (also C7960), and call goes to VM.
No other calls in either system (eg, no load).
Both boxes have iax config'ed as:
trunk=yes
allow=ilbc
jitterbuffer=yes
Recorded VM messages are very distorted.
Changing only
2006 Mar 18
1
Polycom IP600 - no ring?
Have a strange problem...
When a C7960 calls the Polycom ip600, the ip600's first line button
blinks, the ip600 display shows the proper callerid, but the phone does
not ring at all.
If I call the same ip600 from a bt102, the ip600 rings properly.
If I call the same ip600 from another C7960, the ip600 rings properly.
All phones and asterisk are on the same lan within a few feet.
The
2004 Mar 31
2
C7960 "busy" notification
Using the following defnitions with a C7960:
exten => 3001,1,Dial(SIP/3001,15,r)
exten => 3001,2,Voicemail2(u3001)
exten => 3001,102,Voicemail2(b3001)
exten => 3001,103,Hangup
If someone is on this phone (real conversation) and another call comes in,
the second call goes through the 15 second timeout and is dropped into the
2-priority as "unavailable" (not the 102 busy as
2003 Sep 09
0
Snom200 -> C7960 noisy?
When a Snom 200 (v2.1l) calls a C7960 (v4.4), both using g711u as default,
the conversation is extremely noisy from the Snom to the Cisco, but clear
in the reverse direction. Using a sniffer, I see packets from the Snom to
the Cisco of 87 bytes and Cisco to Snom of 214 bytes. Asterisk is CVS from
Saturday.
The communications between the two was working fine on Saturday, however
something has
2004 Jan 15
1
QoS anyone?
Has anyone played around with QoS or TOS relative to * and sip phones?
I was just doing a little real-time research and noticed our C7960's
mark IP packets with "low delay" and "high throughput" (presumably due
to tos_media: 5 in the SIPDefault config file), and rtp packets flowing
"from" asterisk back to the sip phone are not marked at all.
Is there a * config
2004 Apr 06
3
Problems with IAX2?
Are there open problems/issues with iax2 and jitter (quality)?
Just upgraded to today's dev cvs about an hour ago, and it seems the iax
conversations are lower quality then a month or two ago. iax2 show firmware
says version 13. (Test call originated from C7960 with g711.)
Using the demo as an example,
iax2 show channels
Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter
2003 Sep 09
1
help on MOH config, pretty close?
Trying to test the music on hold function and can't seem to get it to work.
If anyone has it running, could you give me a clue? (I have googled and
found lots of questions, but no real suggestions.)
I downloaded and installed the mpg123 package. From the RH9 console I can
start the executable and hear the music via the speakers. The executable
is located in /usr/bin. (That works!)
I set the
2003 Sep 13
1
Does * machine need a sound board for MOH?
Does anyone know whether the linux machine running * have to have a
sound card on it in order for musiconhold to work for sip phones?
I've tried about everything (including tons of google searching) to get
it to work, and nothing.
When a call is placed on hold between two C7960's, the CLI indicates:
-- Executing Dial("SIP/3002-c418", "SIP/3000|20") in new stack
2004 Jul 25
2
Incoming SIP gateway context?
I just started service with Broadvoice.com and everything seems to work.
However, apparently my understanding of incoming sip contexts is less
then what I thought it was. Could someone point me in the right direction?
(* on a public address, CVS-HEAD-07/12/04, C7960 phones)
In my sip.conf I have:
[general]
port = 5060
bindaddr = 0.0.0.0
allow=ulaw
tos=0x18 ;sets ip tos bits (=lowdelay and
2005 Oct 12
0
Notice message meaning for C7960?
Asterisk cvs-head compiled 2005-10-07 11:
Oct 12 18:35:12 NOTICE[21740]: chan_sip.c:10685 handle_request_register: Registr
ation from 'sip:301495906@204.212.194.101' failed for '208.5.218.28' - Not a lo
cal SIP domain
The sip phone is a Cisco 7960 with one line defined, and registration
with * is occuring just fine. Calls to/from the phone are fine. The
phone is on a distant
2003 Sep 09
2
Has the "allow=all" function changed in sip.conf?
I had posted earlier asking about a Snom200 communicating with a C7960
and lots of noise in one direction. Turned out the problem was created
by me removing the allow=all statement in sip.conf. Someone had suggested
that statement is no longer needed, and using allow=ulaw, etc, had an issue
where one or more deny's had to be used as well.
By adding allow=ulaw in the sip.conf file, the Snom
2005 Oct 12
2
Polycom: Button Remapping, HELP!
I need to find a way to have the Polycom phones automatically park
calls. Right now my users hit #70# (I know the last # is optional but
it speeds it up.) to park a call. Personally I think this is easy, but
my users would like one button to do this for them. The Polycom has
buttons we don't need (Transfer & Services), it would be nice if I could
remap one of those buttons to dial
2005 Aug 28
1
Sip pickup
Hi,
In my office I%u2019m using mixed architecture of Zap and Sip phones, everything works fine but I have got some problems with picking up Sip channels. To be certain I can%u2019t do it at all, after I%u2019m dialing *8 the console says "nothing to pick up" (despite I configure appropriate callgroup and pickupgroup).
Do I need some additional application or Asterisk code
2003 Oct 23
6
Festival on RH9?
I'm about to download Festival source, apply the astrisk diff's, and
initiate basic testing. Thoughts are to download v1.4.3 (latest per
the fesitval website.
If anyone has an existing how-to, install notes, tips, or any suggestions
I'd greatly appreciate it. Direct email is fine if you'd rather not post
them.
Thanks,
Rich
radamson@routers.com
2015 Jul 30
1
Dovecot under Linux with mail-extension and case insesitiv
Am 30.07.2015 um 08:52 schrieb Jost Krieger:
> On Wed Jul 29 22:42:32 2015, Sascha wrote:
>
>> i use doevecot 2.2.18 current. My Problem is with email-extension and
>> case sensitiv folders.
>>
>> Example:
>> user+extenstion will be delivered to the user and subfolder extenstion
>> so this is okay.
>> but user+extenstion will not be delivert to the