Displaying 20 results from an estimated 2000 matches similar to: "Re: delaying "answer" for a number of ringsor an amount of time"
2006 Feb 02
1
delaying "answer" for a number of rings or an amount of time
I want Asterisk to delay answering the POTS line via a Wildcard (a Zap
channel) by some period of time, either a number of rings or just a
number of seconds.
I have tried this:
[from-pots]
exten => s,1,Wait(30)
exten => s,n,Answer
...
exten => s,n,Dial(SIP/brian&SIP/joe,10,H)
exten => s,n,Voicemail(u2001)
exten => s,n,Hangup
exten => s,103,Voicemail(u2001)
exten =>
2006 Feb 03
1
Re: delaying "answer" for a number of rings or an amount
Well in my setup I have a few IP phones connected to Asterisk as well as POTS phones on my analog line. When a call for my daughter comes in on the analog line (determined from callerID) I send it to her own voicemail after 20 seconds of ringing. It all works quite well.
Here's a step-by-step of what happens below:
1 - a call comes in and Asterisk rings SIP/Brian and SIP/joe for 30 seconds.
2006 Feb 02
1
SV: delaying "answer" for a number of rings or anamount of time
http://lists.digium.com/pipermail/asterisk-users/2005-September/125146.html
-----Ursprungligt meddelande-----
Fr?n: asterisk-users-bounces@lists.digium.com genom Brian J. Murrell
Skickat: to 2006-02-02 20:14
Till: asterisk-users@lists.digium.com
?mne: [Asterisk-Users] delaying "answer" for a number of rings or anamount of time
I want Asterisk to delay answering the POTS line via a
2006 Feb 03
3
SV: SV: delaying "answer" for a number of ringsor anamount of time
>From what I understand it means that the *hardware* in your computer *acknowledges* the call as soon as it is recieved and then sends it to asterisk dialplan for processing.
You would essentially need to put the delay before the call ever reaches asterisk. So this problem isn't asterisk related... if I've understood your question and the answer I found correctly.
Regards,
Jan
2004 Aug 03
0
ZyXEL 2000w In Call Menu/Hold configs
Hi Everyone,
After a fair amount of faffing ive managed to get the 2000w working with
asterisk for IP -> PSTN calls (i.e. get the phone to make and receive calls
over our BT line). The final solution is to set up outgoing VoIP calls but
I now know that without a SIP aware router I can think again! (damn you
iptables!)
In the mean time I'm trying to figure out why I can't get the
2005 Feb 24
2
softphone has problem to call out via X100P card
Hi all,
I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC.
With the following configuration, I can use one softphone (2000) to call the
other one (2001) and/or the voicemail at 2999.
Here is my problem:
1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via
X100P card, I got busy tone. (i.e. I want to use the phone line which is
connected to the
2005 Aug 10
1
asterisk query mysql problem or bug?
Hi;
I have entries as below in DB,
mysql> select * from sip_buddies;
+----+------+----------+------------+---------+------------+--------+-------
-----+------------+----------+------+
| id | name | context | defaultip | host | mailbox | type |
regseconds | ipaddr | username | port |
+----+------+----------+------------+---------+------------+--------+-------
2005 Jul 05
4
Asterisk on Linksys WRT54G
Hi all,
Any one tried installing Asterisk on Linksys WRT54G? We have but facing
problems with SIP to SIP calls. The phones ring and calls are established
but we cannot hear any voice at all. I tried allow=all in the general
section but did not work. So I forced ulaw. Can any one please check it out
and let me know what is wrong?
Here are the conf files:
Asterisk Version: Asterisk
2003 Nov 15
2
Internal server error - cannot align media streams - help needed
Hi,
I configured asterisk on redhat linux 9 box. I installed two different
ip softphones (SJPHONE and X-PRO) and got them registered with asterisk.
The call from one phone to another does get routed via asterisk, but
there is one problem coming up. As soon as call is accepted by the end
user , it is automatically disconnected with the error "cannot align
media streams". If I enable SIP
2005 Feb 12
1
iax.conf config and iax based clients
Hi,
I am a newbie in asterisk. trying to configure firefly third party edition
to connect to aserisk 1.0.3 im able to authenticate but cannot dial
extensions. I have been reading the documentation cant seem to find the
correct configs. Attached the error message and configs. What am I
missing?
*CLI> Urgent handler
Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected
connect
2005 Jul 01
1
no voice
Hi All
We are unable to hear any voice where as in tcpdum it shows that RTP is flowing both ways
ERROR CONDITION
---------------
-- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack
-- Called 2000
-- SIP/2000-0ead is ringing
-- SIP/2000-0ead answered SIP/2001-f6c4
-- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead
Have searched web and
2005 Jul 15
2
[Aserisk-Users]no audio inside the net
Hi list, i've problems with my * server and the 4 phones which are
linked to it. i've 2 grandstream bt100 with the firmware upgraded to
101, a wi-fi phone (i don't know its brand) and another ip phone i
don't know its brand. with this sip.conf :
[general]
port = 5060
bindaddr = 192.168.100.229
context = default ;x changed from default to sip
localnet = 192.168.100.0/24
2005 Jan 19
1
My dialplan just stopped working one day
Hrm,
All of a sudden for some reason Wait() and Playback() are returning
non-zero and its causing calls on my inbound SIP leg not to complete.
I'm not sure why
-- Executing Answer("SIP/2181-4518", "") in new stack
-- Executing Playback("SIP/2181-4518", "silence/1") in new stack
-- Playing 'silence/1' (language 'en')
== Spawn
2004 Dec 17
1
Troubleshooting Asterisk
Guys,
Ok - nowhere near as complex as most of the discussions on here ( ex telco
engr for 18 years here).. But thought I'd ask for some assistance.
Have just set up my first * Pbx - having a play with it and a couple of
Cisco 7960 (configured as SIP) phones.
The phones are tftp'ing into the server ok, and picking up the configs all
ok.
Everything _seems_ to be working, but I
2004 Jan 02
1
Asterisk Gotoif / last called
Hi guys
Ive been trying to get this to work for ages now, basicaly im trying to do if ${woteva} = "" (nothing), or its none existenant then do label 1, else label 2. for my last called function, so it will play a different message if theres no last call in the system or it was anonymous.
ive tried
exten => 1000,1,GotoIf($[${last-call${CALLERIDNUM}} = ""]?4:5)
and heaps of
2005 Jan 17
1
Attempting native bridge
ERROR CONDITION
---------------
-- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack
-- Called 2000
-- SIP/2000-0ead is ringing
-- SIP/2000-0ead answered SIP/2001-f6c4
-- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead
Have searched web and archive w/o good results.
Thks in advance for any help,
Dave
sip.conf
--------
[general]
port =
2004 Dec 13
1
incoming call from pstn to fxo not working with Asterisk
When somebody call me on my pstn # cable connected to my fxo card it does
not work when I check my computer the following error shows
Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on
asterisk1 (pid = 2160)
Verbosity is atleast 3
-- Remote UNIX connection
-- Starting simple switch on 'Zap/1-1'
== Starting Zap/1-1 at incoming,s,1 failed so falling
2003 Aug 26
0
TDM10M && Siemens Euroset 2015
Hi all,
--------
I have installed a TDM400 with one active FXS port (TDM10B) an connected
it to a Siemens Euroset 2015 analogue phone.
I have installed some smom IP phones to the network as well and
configured them as usual (sip.conf). For configuring the TDM10B I have
used FXO signalling in /etc/zaptel.conf and in
/etc/asterisk/zapata.conf. I definded the TDM channel and the Snom
phones to the
2004 Aug 13
1
OH.323 Dialout Problem
Hi,
I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular
phone. Asterisk configuration is listed below. When I attempt to place a
H.323 call, I receive the following errors:
- Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20")
in new stack
Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path
exists
2006 Feb 01
1
SV: Re: CallerID Problem
This is what i found on Cisco's site:
"Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a "488" message to an Invite message.
Conditions: This symptom is observed on a Cisco platform when the terminating gateway is configured with the G279B (annex B) codec and when the Session Description Protocol (SDP) for the incoming Invite message