Displaying 20 results from an estimated 2000 matches similar to: "FWD/SIP audio suddenly stopped working"
2003 Nov 28
0
Can't seem to connect/call fwd network Help!
I have tried everything and still can't place / receive calls from the fwd network. At one point today I was able to call my test machine on the fwd network, I'd answer the call on the test machine (which stated Call Connected), but then the computer I was calling from, through the Asterisk server would give me a 403 Error. I am using sjphone software. I am able to call various
2003 Dec 26
0
fwd problem with *
Hello
I am trying to register for fwd from * but having problem and unable to solve it.
I keep getting this message
*CLI> NOTICE[1125329600]: File chan_sip.c, Line 4800 (handle_response): Failed to authenticate on REGISTER to '<sip:89699@fwd.pulver.com>;tag=as62a7f29b'
NOTICE[1125329600]: File chan_sip.c, Line 4800 (handle_response): Failed to authenticate on REGISTER to
2003 Sep 28
1
Forwarding SIP over IAX problem: No One Available
I'm hoping someone can help me out with this. I am basically just
trying to separate my outbound and inbound calls into separate contexts
instead of having everything in a single context. Any help would be
appreciated. Perhaps I've missed something really obvious....
Here is the network layout:
<remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>
2005 Mar 19
1
Asterisk Quicknet FWD Problem - no path to translate from Phone/phone0 to SIP
I cant seem to be able to figure this out. As much as I can tell it is
a codec problem.
I can dial out to 612@fwd.pulver.com and the "Call Me" test there rings
my phone. However when the callee endpoint answers, there is a failure
to translate:
Outgoing Call for 612
612 is not a local user
-- Called 612@fwdpulvercom
No path to translate from SIP/fwdpulvercom-dd5a(2) to
2004 Dec 22
2
Can't Receive/Send Calls
Hi,
I can't receive/send calls with Asterisk. Could someone please give me a
few pointers on my configuration?
Regards,
Norman Zhang
; sip.conf
[general]
disallow=all
allow=ulaw
port=5060
bindaddr=0.0.0.0
externip=x.x.x.x
localnet=192.168.22.0
mask=255.255.255.0
context=inbound-sip
maxexpirey=180
defaultexpirey=160
tos=reliability
srvlookup=yes
register =>
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
2004 May 19
1
Strange Sip (FWD, SipGate and such) problem
Hi all
I use sipgate and FWD but seem not to get it going. I do not have NAT on
the asterisk box (static ip).
The asterisk box has 2 network interfaces. One internal and one external.
Now when I make an call to a FWD or SipGate number all I get is
-- Executing NoOp("SIP/113-6d2e", "") in new stack
-- Executing Goto("SIP/113-6d2e",
2004 Jun 21
0
dialplan help!-RESOLVED
All,
I was a bit too focused on where I thought the problem was - turns out
I wasn't crazy and the dialplan does work as expected. The problem was
with dtmf detection - setting relaxdtmf=yes did the trick. Sorry for
the premature post for help.
Begin forwarded message:
> From: Ben Witso <benw@bgwcomp.com>
> Date: Mon Jun 21, 2004 7:28:42 PM US/Central
> To: Asterisk-Users
2005 Jan 16
2
FWD<->NAT<->*
I found this configuration file on Wiki for FWD behind firewall
; SIP Configuration for Asterisk
;
[general]
disallow=all
allow=ulaw
port=5060 ; Port to bind to
bindaddr=0.0.0.0 ; Address to bind SIP channel to
externip=xxx.xxx.xxx.xxx
localnet=172.16.1.0
localmask=255.255.255.0
context=inbound-sip ; Default context for incoming calls
maxexpirey=180
defaultexpirey=160
tos=reliability
2004 Oct 07
1
Confused about NAT and Authentication with FWD
I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT.
The problem:
I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem.
Now if in the extension.conf file I have,
exten =>
2004 Aug 07
2
Asterisk : No Sound No Dial
Thanks for taking a look greg and hank. This seems to be getting bettre
everyday..help please
My sjphone is running on the same box as asterisk...i believe then the red
hat firewall should not be a problem.
Whenever i dial from CLI i get
#########
Executing Goto("OSS/dsp", "default|s|1") in new stack
-- Goto (default,s,1)
-- Executing Wait("OSS/dsp",
2006 May 02
0
Insights on SIP channel usage in * 1.2.7.1 are welcome!
I've had a heck of a time getting a SIP channel to work in Asterisk
1.2.7.1 (Redhat 9.0). I've done it successfully a number of times on
pre 1.2 versions of Asterisk so perhaps it's version related. Any
insights are welcome!
At first I wasn't able to dial out on the SIP channel _whenever_ I
started Asterisk (i.e. not just when the box was booted). I always
had to do a reload
2004 Jun 28
2
Incoming IAXTel/IAX2 issue
Hi all,
I spent most of the last weekend testing and trying to diagnose some
mostly incoming call issues. I'll start with the easy one in the hopes
it might have a positive impact on the others. First- I have an account
with IAXTel. I can place calls to other IAXTel subscribers and also
through IAXTel to landline toll free numbers and all works great. iax2
show registry shows I am
2004 Oct 01
1
Solution to my Grandstream lockups
Like many others on this list, I had been experiencing periodic
lockups with my Grandstream products (Handytone 286 ATA & BudgeTone
101). The lockups consisted of seemingly dead devices, no dialtone or
response, until I power cycled via software or hardware. The
workaround had been to reboot the device every 30 minutes with a cron
job. I contacted Grandstream and although they didn't
2006 Apr 08
0
Re: [asterisk-dev] bug or bad chan_sip.c
Tzafrir,
How did you set sip:tzafrir@local.xorcom.com
I use ser----asterisk
look at my sip.conf and extensions.conf
Regards
Harry
////////////////////////////////////////////////////
[general]
context=sip
realm=nxs.yi.org
bindport=5050
bindaddr=nxs.yi.org
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=1000
allow=all
musicclass=default
language=fr
insecure=very
allowguest=yes
2006 Apr 08
0
Re: [asterisk-dev] bug or bad chan_sip.c
Tzafrir,
How did you set sip:tzafrir@local.xorcom.com
I use ser----asterisk
look at my sip.conf and extensions.conf
Regards
Harry
////////////////////////////////////////////////////
[general]
context=sip
realm=nxs.yi.org
bindport=5050
bindaddr=nxs.yi.org
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=1000
allow=all
musicclass=default
language=fr
insecure=very
allowguest=yes
2004 Dec 23
1
Qestion about TDM over enthernet
who can tell me how to do TDM over enthernet ?
pc a connect pc b only use TDM card?
thank you
John.
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????: 2004?12?23? 11:47
???: asterisk-users@lists.digium.com
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2007 Feb 15
2
7912 phones loosing registration
I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to
be exact).
I get the X on the display sometimes for loosing registration.
I have the config file for the 7912's
SipRegInterval: 60
and asterisk is the default.
; maxexpirey=3600
;defaultexpirey=120
I've not changed them.
How can I keep these phones online and stop loosing registration?
Thanks,
Jerry
2007 Jul 17
0
help with sip configuration for sipgate.de on asterisk 1.4
hi there,
i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main
computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de,
there i can receive call and make them, i can hear the other end but they can not hear me, this is
only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help
2004 Sep 28
1
Codecs and negotiations
For some reason I now seem unable to control which codec is chosen. The
problem happens with outgoing calls to Stanaphone. Even if I chose
disallow=all and allow=ulaw as the only codecs it connects with GSM.
Has anyone else got problems with these settings? Any suggestions? As I
recalled it, such a setup would not establish a call if the ulaw-codec
was not offered by the provider. Stanaphone has