Displaying 20 results from an estimated 10000 matches similar to: "[1.2.5] DTMF not being set correctly (RESEND)"
2009 Apr 09
2
DTMF
[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted:
Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but
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2005 Sep 05
2
USING TWO ACCOUNTS WITH BROADVOICE
Hi,
I have two accounts with broadvoice.
Now, I want to be able to distinguish between them.
I though that this would be simple by adding "/EXTEN" at the end of the
register statement. For example:
register => num1:pass@sip.broadvoice.com/1000
Unfortunately, this is not working.
When I call into my box I hear busy tone.
My config looks like this:
[root@voip asterisk]# cat sip.conf
2004 May 18
1
Configure asterisk for outgoing.. need authuser parameter?
Hi,
I have access to two providers. On one of them the authuser is the same as
the username, so outgoing works. On the other one I can only get
incoming -
what ever combination I try for outgoing I get an error. The register
command
has the ability to specify both usernames (which is why incoming works) but
outgoing doesn't seem to, and without that I'm stuck.
They are defined as:
2005 Aug 16
1
Issue with DTMF Tones - Codec Issues
Topology:
PSTN<-T1 PRI->NEAX2400<-T1 PRI->Cisco 3825<-Ethernet-> Asterisk VoIP server
When I make a call to a VoIP user from the PSTN, the call gets routed
through the PBX, and Cisco. Because of that the DTMF tones are passed
inband, which I can hear on the VoIP end of the call. However, I have
one extension on asterisk set up so that I can check voice mail when
away from my
2008 Dec 29
1
DTMF does not work
I got no resonses to this and some funny bounces so I'm trying again.
First of all Merry Christmas.
Second, my first problem with my provider not staying registered with
our server was my fault. We moved our server room and I restarted the
test system and the production system causing them to ping-pong back and
forth registering with our provider causing random problems, they are
both
2005 Jan 27
1
Stumped by BroadVoice SIP
Hello guys.
I am a fairly new user to Asterisk, and I'm just having a tough time.
My goal is to set up a VOIP PBX. I have signed up with a BroadVoice
number, and I have three systems with SIP phones.
The PBX and the SIP phones are all behind a Cisco PIX running NAT.
I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with
little luck.
The SIP phones are two X-Lites on
2004 Sep 01
5
dtmf problem
Hello!
I have asterisk updated from CVS on 31/8/2004 with
sample configuration. I have just changed the
sip.conf to register asterisk with sip proxy in out
intranet.
Then I can successfully make call to asterisk and go
to demo IVR, but no response to dtmfs.
I try to make call from several sip phones: Cisco7960,
Ata186, Snom200. All of them send telephone-event in
INVITE, but asterisk answers
2005 Mar 08
1
All Circuits are Busy Now
I have downloaded and installed Asterisk@home and I have installed X-Lite on my Windows machine and I am able to connect it to the Asterisk server. I went ahead an created an account on Broadvoice today and followed the directions on http://voip-info.org/wiki-Asterisk+settings+Broadvoice and http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but when ever I try and make a call from
2008 Jan 07
1
Strange migration problems from asterisk 1.2.13 to 1.4.10, dtmf related?
Hello,
I have the following problem. I am migrating my asterisk infrastructure
to a new server and I encounter a strange problem. The configuration is
as followin: IAX clients connect to asterisk which forward calls to a
sip box connected to a phone line. On the old server everything works ok
but on the new server, even if the logs are identical it seems like the
dtmf number does not get passed
2003 Dec 12
1
simple question on sip.conf
Hi folks,
I want to fix hole in my asterisk set up.
I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN,
Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go
'other' places. This senario works fine.
Now the issue is someone else running a vocal or another SIP proxy can
redirect his calls to my * as well. Those calls two will come through
general
2005 Jan 05
4
Broadvoice / * re-register issues
HELP!
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw
register =>
##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234
[sip.broadvoice.com]
type=peer
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
2005 Mar 17
1
Last guy to get BV working outbound?
I have tried everything to get BV working outbound. All worked fine
until the BV change last week. I called BV and they changed me to sip
gen with a new password. I stripped my Asterisk server to one phone on
Zap/1 until I get this working. The same BV account works fine with a
SPA-3000 so I don't suspect a firewall problem.
Symptoms: Asterisk registers with BV Ok
Incoming calls work
2005 Jan 25
4
BroadVoice Help
Is the Broadvoice service up? I just signed up with them and started
receiving calls in no time but could not make calls. And after a few minutes
I cannot even place calls.
register => [number]:[password]@sip.broadvoice.com
[broadvoice]
type=peer
fromuser=[number]
host=proxy.lax.broadvoice.com
fromdomain=sip.broadvoice.com
context=from-broadvoice
dtmfmode=inband
any help would be
2006 Mar 04
1
Asterisk to a Huawei softX3000
Greetings,
I'm having a job getting asterisk to register with a Huawei softX3000
softswitch via SIP. I keep getting 401 Unauthorized. Funny thing is I can
successfully register SJPhone, a PA1688 IP Phone as well as a WiFi Phone
against the switch without *any* problems. I think it's got to be something
as simple as perhaps the register string which is currently
2005 Mar 08
13
Broadvoice latest changes and still not working
I have added the three lines to the sip.conf file based on the latest
changes
from broadvoice. I can receive incoming calls but cannot place any
outgoing calls.
The error I get is:
*CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569
-- Attempting call on SIP/Broadvoice/5068012 for application
Playback(demo-congrats) (Retry 1)
Mar 8 08:35:21 NOTICE[29290]:
2005 Aug 13
1
receiving calls from FWD
I have successfully configured asterisk to make outgoing calls over FWD,
but cannot receive incoming calls. The console shows no messages,
even though an XTEN client on the same network has no problems receiving
incoming calls.
This is the relevant part of sip.conf
[general]
.....
register => 688426:xxxxxxxx@fwd.pulver.com/6000
[fwd.pulver.com]
type=friend
username=688426
fromuser=688426
2003 Nov 19
2
g723 to g723 SIP call - warning message
Hi,
I am calling from a grandstream phone with g723 codec through * to iconnect.
Incoming context as well as outgoing context set to g723.1 codec in *.
Call get connected and I can talk. However I get the following warning,
which scrolls on my screen until I hang-up.
[root@asterisk sath]# cat g723.1
- Executing SetCallerID("SIP/-08122ae0", "1001") in new stack
--
2004 Sep 14
1
Setting up Asterisk with fwd
Hey all,
I'm trying to get my Asterisk server up and running on
fwd.pulver.com just to get the hang of it until I get
my FXO card in a couple of days. It seems to connect
but that's about it. If I try to dial into it from
another fwd # it says user is not online.
In sip.conf I have the following added:
register => xxxxxx:xxxxxx@fwd.pulver.com/489125
[fwd.pulver.com]
type=friend
2007 Apr 18
2
incoming SIP call
Hello all,
I'm having a quite simple configuration like:
SIP provider <=> asterisk SIP <=> lan
Everythings works fine but sometime I can't get incoming call.
here are some of the logs from set debug 25 set verbosity 25 sip show
debug and sip.conf and a part of extension.conf
thanks in advance
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net
2006 Apr 07
1
wellgate registration 3802
I have a new wellgate 3802 unit. I have not gotten it to
register with asterisk 1.2.6.
My proxy setting is the correct IP in the 3802.
My security config is 1001/1001 and 1002/1002 on the wellgate (simple at
this time).
My sip.conf has:
[wellgate3802L1]
type=friend
dtmfmode=inband
username=1001
secret=1001
host=dynamic
canreinvite=yes
nat=no
context=wellgate
[wellgate3802L2]
type=friend