Displaying 20 results from an estimated 20000 matches similar to: "broadvoice and gsm codec"
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]:
2004 Dec 04
2
Broadvoice outbound 404 error
Is anyone else experiencing 404 errors on outbound dial with Broadvoice? I've
followed the instructions posted by Broadvoice to configure sip.conf, and
inbound calling works fine. Every time I try to dial out, I get a 404 "Not
Found" error.
Here are the relevant sections from my configs.
sip.conf:
context=broadvoice-in
register =>
2005 Jul 24
1
Help with Asterisk@home and Broadvoice incoming calls..
Hello everyone,
Well here is my initial posting to the list, and I will admit Asterisk is new
to me. I just got everything running here a couple days ago, so still learning
the ropes for sure.
OK, here is my problem. Currently I have it setup talking to a couple Cisco
IP phones, and some Xten softphones, this works great. I also got an account
with FreeWorld Dialup using IAX2 and that
2004 Oct 23
7
Asterisk and Broadvoice, no incoming voice
I just signed up for the BroadVoice service a few hours ago, but for
the life of me I can't get any incoming voice. The incoming
connection is fine as it rings my extension from outside, but I can't
hear anyone talking. Outgoing voice is working fine though.
I've been looking through the archives, but I haven't found a solution
to the problem yet. I even tried another router
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
2008 Mar 16
1
Problem with incoming calls on Broadvoice after upgrade to 1.4.18
Hi all,
I just upgraded to Asterisk 1.4.18 a few days ago and I don't use
Broadvoice TOO often, however I have a Vermont number with them and so
my mother in law calls it to talk to my wife once in a while, so
that's why it took me so long to notice it wasn't working. Anyway,
when she calls she gets a busy signal (as I've tested when calling it
from my cell).
When I enable
2004 Dec 04
1
Codec translator problem (g723.1,ilbc => alaw)
Hi, I cannot get SIP channel working with folowing codec configuration:
[sip]
disallow=all
allow=g723.1 ;I need this codec between sip phones (BT100)
allow=ilbc ;Use this codec to others
Calling between BT100 SIP phones is OK - asterisk makes native bridge
(with g723.1) between them.
When I'm calling from SIP to other channel (iax,zap,...), asterisk is
not able to chose right codec
2005 Sep 18
2
Asterisk Won't Process Call
We have a basic application that runs a SIP channel to pick up a call
and process it. We are using Broadvoice and it's been working great.
We recently rebooted the machine and now when a call comes in Asterisk
picks up the call but does not process it. Asterisk seems to send the
call back to Broadvoice. Nothing at all has been changed in the
configuration to warrant this. Below is the
2005 Jul 28
2
SIP Debug
Using AMP, the configuration I have used to work fine with Broadvoice.
Now it gets a busy signal every time. I've checked "sip show
registry" and it says it's registered just fine. I've tried "sip
debug" and it shows calls coming in, but they always get a busy signal
& I can't tell why.
Here's a SIP Debug output:
Sip read:
INVITE
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
2005 Jan 26
4
A working BroadVoice config example
I finally got my incoming and outgoing to work on Broadvoice with a
configuration file that is no where close to the one given by them.
Here it Is (sip.conf). For others who have a working config could u please
share so that I could compare. Thank You
[broadvoice]
type=friend
username=[number]
fromuser=[number]
secret=[password]
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
2006 Oct 24
10
Meetme... No channel type registered for 'zap'
When I call meetme:
exten => 1000,1,Answer
exten => 1000,n,Meetme(|||d)
Asterisk is complaing with:
-- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack
-- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack
-- Playing 'conf-getconfno' (language 'en')
Warning, flexible rate not
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
2004 Nov 26
2
Help with broadvoice outbound plz... ;)
*sigh*
Ok, I have fought and fought with this. I have read all the FAQ's, I have
scanned the list archives. I can receive calls on * from my Broadvoice
acct, but I cannot place calls...
I have the 'World Unlimited' plan, and like 5 area codes that are local to
me in Dallas.
Can anyone help me?
here are my config files...
- sip.conf -
[general]
context=default ;
2005 Aug 19
2
Sudenly unable to get incoming from Broadvoice
So it was all working well and then suddenly I'm unable to get incoming
calls from BV. Outgoing is fine. I'm using AAH.
I have the following settings;
register=9738281625@sip.broadvoice.com:PASSWORD-GOES-HERE:9738281625@sip.broadvoice.com/2208
[broadvoice]
username=9738281625
user=phone
type=peer
secret=PASSWORD-GOES-HERE
qualify=1000
port=5060
nat=yes
insecure=very
2004 Jun 01
2
BroadVoice usage?
Hi all,
I've been trying to use BroadVoice as a SIP service provider. They don't
officially
support * but are helpful when it comes to answering questions for setup
parameters. They claim they have no firewalls or access lists that need to be
set up so I can get access to their servers.
However, something's still not quite right when I use the parameters.
It looks like our Asterisk
2005 Jul 19
4
Asterisk Quit Registering with Broadvoice
Hello -
I've been using Broadvoice with Asterisk for a couple of months with no issues. Today, it has stopped registering. The Sip Debug from CLI is below. It tries to register five times and then gives up. Any suggestions? As you might suspect, I have not been able to get Broadvoice on the phone and usually get cut off after being on hold about 5 minutes.
Strategic portions of IP
2004 Oct 05
2
broadvoice connection problem
All,
I signed up for a broadvoice BYOD plan over the weekend (very
excited about their offering) and after about an hour I had asterisk
registered and was making in and out bound calls. However, the next day
(without changing anything) I couldn't call in or out and haven't been able
to get it going again. I can connect using a softphone (X-Lite) and make
calls in and out
2005 Feb 26
1
Dial out through Broadvoice
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get the following:
Executing Dial("SIP/147.135.0.129-0815bc60",
"SIP/16037862111@proxy.bos.broadvoice.com|30") in new stack
-- Called 16037862111@proxy.bos.broadvoice.com
-- Got SIP response 480 "Temporarily Not Available" back from
2009 Feb 23
3
GSM codec is a good choice ???
Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
with GSM sound files.
The problem is I have IP phones Utopix HyperPhone 202 which support
only G.729a/u and G.723.1 high/low, but not GSM.
If I choose G.729A the "pass-throu" calls among users are OK, but
Asterisk can't transcode GSM to G.729A in voicemail calls.
My company doesn'y want to pay for a G.729