Displaying 20 results from an estimated 2000 matches similar to: "Is answer() necessary ?"
2009 Nov 23
1
Is Answer really needed
Hi
All my incoming dial plans start of with an Answer which I now know
starts the billing time. Some of the dialplans then get forwarded out to
POTS via a carrier and so the actual amount of time that should be
billed is being distorted.
I've done a few tests this morning and found that if I don't start with
an answer then the billsec of my forwarded call is actually the length
of
2010 Apr 13
2
SNOM M9 base station A to base station B
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<small><font face="Helvetica, Arial, sans-serif">Hello,<br>
<br>
I have a question concerning SNOM M9 base station.<br>
<br>
If my customer places a SNOM M9 base
2009 Apr 13
10
Asterisk-beginner : cannot make phonecalls using Asterisk
Hi there,
this is the first time that I'm building an Asterisk-server.
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.
Thought it would go easier...
I have 2 Grandstream IP-phones : BT-201 and GXP-1200.
These are my settings :
sip.conf :
[root at asterisk asterisk]# cat
2009 Mar 30
2
no ringtone - just silence/bridging of external calls
Hi Community!
If this issue was already topic, please excuse or delete my request...
Topic 1 "no ringtone":
I configured a SIP registration with my SIP provider (SIPCall).
Everything works fine except the ring tone for the caller. The caller
hears silence until the called party takes up the phone.
I used the DIAL command with the r and R option but no luck... :(
Has anybody the same
2009 Sep 07
2
features.conf : feature map ==> getting feature to work
Hi there,
I need some help with a 'custom' feature.
I have following feature defined in features.conf :
[applicationmap]
opnemencallee =>
#3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m
In my dialplan :
[from-HostAst]
exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee)
exten => s,n,Dial(SIP/grandstream,30)
I want the callee to be able to press #3 to be able
2010 Mar 01
1
rtcachefriends & qualify
[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify
is incompatible with dynamic uncached realtime. Please either turn
rtcachefriends on or turn qualify off on peer 'gerrie'
Am I correct that when I turn on rtcachefriends in sip.conf,
database-changes in my MySQL-DB will not be reflected untill a reload ??
Am I correct that when I turn off qualify in my realtime
2009 Oct 21
3
Searching on how to keep local calls... local
Hi list.
Does anyone know how to keep calls between 2 local SIP-phones on the
local private network when the 2 local IP-phones are registered to an
online public Asterisk-server ??
What network-element / router do I need to install to prevent the
RTP-traffic from flowing via the internet ?
Config :
Asterisk --internet-- > router/firewall --> connected local IP-phones
Internal call :
2009 May 12
2
Hangup()-command does not hang up the line
When I call my Asterisk-server from my cell phone on one of the
PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card,
and in the dialplan the end of a context is reached and Asterisk needs
to execute the Hangup()-command, I notice the following :
- Asterisk tells me that the conversation was hung up (the log files
tell me the command was executed)
- On my cell phone I hear
2004 Aug 29
1
not getting ringing/busy/answer feedback on my PRI
I posted a problem earlier thinking it was due to a lack of sound card.
Several members stated that you do not need a sound card to play audio to a
PRI channel. I did some further testing and discovered that there is a
problem with call progress tones or signaling on my PRI. I think that the
reason I am not hearing audio from the MeetMe() or Playback() apps. is
because the the calling side of
2003 Nov 13
3
iax configuration
Hi,
I have configured 3 users in my iax.conf, i am using iaxcomm phones. Iaxcomm has excellent voice quality although there is no ringing tones(either ring back or ringing tone),but i can live without right now.
I find that for each user i want registered i have to add his name and his ip address.I have been using "host = dynamic".Isnt there any way that i can define a dialmap such as
2004 Jun 10
3
Iax2 ringtone problem
Hi,
i have a problem with iax2 and ringtone.
Here is the call path
pstn -> asterisk -> iax -> firefly or any iax phone.
My problem is when i receive a call on my iax phone, the ring sound is very distort and bad.
If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal.
Otherwise, it is like a machine gun with iax
Help would be really
2009 May 19
5
OT: SIP hardphone with multi-color BLF
Hi,
Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors
(or more) ?
This could be very useful to support extended presence, for instance.
Regards
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2009 May 08
2
Not receiving voicemail message in mailbox
It should be as simple as editing voicemail.conf :
; Voicemail Configuration
;
[general]
; Formats for writing Voicemail. Note that when using IMAP storage for
; voicemail, only the first format specified will be used.
format=wav49|wav|gsm
; Who the e-mail notification should appear to come from
serveremail=asterisk-voicemail
; Should the email contain the voicemail as an attachment
attach=yes
;
2003 Oct 03
9
No Ringback on Iconnect
When I place a call using Iconnecthere as my sip provider, I hear no
ringback when making a call. Does anyone else have this problem or
offer any suggestions? Thanks, Kevin
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2008 Jun 06
2
Bad ringback tone on zap channel
Hi,
I've noticed that sometimes instead of getting a regular ring tone
when calling out on a Zap channel, I get this obnoxious loud noise
which forces me to hang up.
Is this a problem in the Zaptel driver? I seem to recall that ringback
tones are generated by zaptel when dialing out from a SIP phone over a
Zap trunk.
Thanks.
2018 Dec 12
3
Outbound call: caller gets no ringback on session progress
Hello!
An extension registered at asterisk 13.23 initiates an external call (pjsip). After the Invite, the
callee (-> ISP) sends a
100 Trying
183 Session Progress (*without* SDP)
Asterisk now sends to the extension:
183 Session Progress (*with* SDP)
183 Session Progress (*with* SDP) (really two times)
The callee meanwhile sends
180 Ringing (*without* SDP)
which is
2005 Mar 11
8
No ringback over IAX - LiveVoip
Hello All,
I saw some coverage of this in the list archive but no one seems to have
posted a resolution.
I am using Asterisk@Home 0.06 and when I get a call from LiveVoip over
IAX I dump it into my IVR.
>From there the call is routed to groups based upon input.
However, there is no ringback indicated to the IAX caller.
Does anyone know how to resolve this problem?
Thanks,
Wiley
2008 Apr 04
5
Ring back when free?
Has anyone here implemented "Ring back when free" in Asterisk?
The way it works in the UK is as follows:
1. A calls B. B is engaged (busy).
2. A hears "The number you called is busy. To use ringback, press 5"
3. A presses 5, and hears "Your ringback request has been accepted".
4. A hangs up.
5. Later, B hangs up. The system then calls A (if A is now busy, it
2007 Mar 01
4
Cannot hear ringback music from telco
Hello,
We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to
the telco, users mainly use snom 320/300 SIP phones.
When dialing to an external phone number with custom ringback music, users
reported that they could not hear the music but can only hear the standard
ring tone generated by the system.
Is there any kind of settings need to allow the ringback music pass to the
2015 Aug 25
4
Ringback issue
My last problem was nicely solved through this mailing list so
hopefully this new problem will have the same happy outcome.
My situation is that I have many extensions. Here is a sample:
[client-phone](!)
type=friend
host=dynamic
secret=XXXXXXXXXX
dtmfmode=auto
disallow=all
allow=ulaw
allow=gsm
allow=g723
allow=ilbc
subscribemwi=no
[4165555555](client-phone)
secret=xxxxxxxxxxxxxxxxxxxxxxxxxx