Displaying 20 results from an estimated 900 matches similar to: "Troubleshooting Asterisk"
2005 Jan 24
3
OT: Libnewt sourcecode?
Hi,
I'm trying to compile zttool from the Zaptel lib, but I just can't find the sorcecode for Libnewt.
Anyone got a link?
Since i'm using LFS, I can't use precompiled packages.
--
Med venlig hilsen / Best regards
Michael L?jtnant - Systems Engineer
ZyXEL Communications A/S
Columbusvej 5 - 2860 S?borg
Tel (+45) 3955 0700 - Fax (+45) 3955 0707
2004 Jun 22
1
Unable to create channel - CVS Broken?
Hi,
Just started to get this error after updating to the latest CVS. Asterisk dies if it can't create a channel - not so good.
-- Executing SetCallerID("SIP/750-2550", "39660426") in new stack
-- Executing Dial("SIP/750-2550", "CAPI/39660426:22179808") in new stack
Jun 22 13:52:05 NOTICE[262161]: chan_capi.c:1172 capi_request: didn't find
2014 Mar 14
1
syslinux.efi [PXELINUX EFI 64 boot] not properly TFTP'ing ldlinux.e64
H Peter,
I notice the Intel Boot Agent (in the BOOT ROM) takes a different
approach. When TFTP loading the initial syslinux.efi.
It receives the OACK with both options set -- tsize (of 145744)
and blksize (of 1408).
Apparently it has challenges parsing this "two option" OACK packet.
But it remembers the tsize of 145744. So it issues another read
request, this time requesting only a
2004 Jan 26
1
SIP behind NAT - use of "externip" option
I am having difficulty configuring SIP behind NAT (using latest CVS).
Using sip.conf:
[general]
port=5060 ; Port to bind to
externip=ww.xx.yy.zz
bindaddr=0.0.0.0
nat=yes
register=>[userid]:[password]@voiptalk.org/2000
[voiptalk.org]
nat=yes
externip=ww.xx.yy.zz
type=friend
secret=[password]
nat=yes
reinvite=no
canreinvite=no
I fail to register. SIP Debug gives:
SIP
2004 Dec 23
1
Problems with incoming IAX calls...
Trying now to set up the final part of my * switch. I must admit I've had
great fun over the last week or so playing with it, and would like to thank
you guys for all the assistance (past and present, since I've been trawling
a lot of old posts!!!).
Scenario - using voiptalk.org to supply the incoming gateway, tied to an
0845 number for convenience in testing. Internal 7960 -> 7960
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.
-----????-----
???: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]??
asterisk-users-request@lists.digium.com
????: 2004?12?23? 11:47
???: asterisk-users@lists.digium.com
??: Asterisk-Users Digest, Vol 5, Issue 336
Send Asterisk-Users mailing list
2004 Dec 20
7
'I'nvalid extension handling problems, even with workaround
Hello folks,
I'm having trouble configuring Asterisk to play an "invalid extension" message to
anyone dialing an undefined extension.
First I tried using the 'i' pseudo-extension, but it didn't work at all;
searching the wiki I found that page:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension
where it basically says that the 'i'
2004 May 22
1
Sip proxy registration help
Hi All,
I have just installed Asterisk and am trying to connect it to a SIP
account that I currently have with www.voiptalk.org but without any
success. Although I know that voiptalk do provide asterisk accounts I
don't want to convert the SIP account until am happy that it's gonna
work for me. The asterisk box is currently behind a firewall and the
following ports are being forwarded
2006 Mar 28
0
DTMF recognition inconsistent in Asterisk
Hello,
I am experiencing a strange problem and I am wondering if anyone may have
some pointers as to how to overcome it.
I have an account with VoipTalk here in the UK which I have connected to
my Asterisk server. VoipTalk supports IAX2 and SIP and I have connected
to my Asterisk box using both methods. The problem is when I dial into my
Asterisk box via my VoipTalk incoming PSTN phone
2004 Aug 30
1
IAX.conf problem (NEWBIE ALERT!)
I have several of incoming numbers on IAX from voiptalk and magrathea
but have a problem with IAX.conf. If I follow the example from voiptalk
[VoIPTalk Incoming Number]
type=friend
username=VoIPTalk Incoming Number
context=[XXXXXXXX]
and make incoming entries in IAX.conf for the numbers like below with a
different entry for each number pointing to a different context,
incoming numbers always
2004 Dec 22
2
Why use 'Answer'?
Why is it that newcomers always feel like inserting 'Answer' is a
necessary step in their extension.conf entries?
>[voiptalk.org]
>;forwards any calls starting with an "8" thru voiptalk.org
>exten => _8.,1,Answer
>exten => _8.,3,SetCIDNum(55555555)
>exten => _8.,4,SetCIDName(My Name And Surname)
>exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g)
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 Feb 21
1
NAT-helping outbound proxy
Hi,
We're deploying a small VoIP solution for a group of teleworkers.
Naturally, this exposes us to all sorts of fun, most of which we seem to
have working properly. However, some NAT issues are still bugging us and
we have noticed that often these situations didn't exist when users were
connected directly to our VoIP provider, voiptalk.org.
They have something which they call a
2010 Jan 09
1
UK dialing tone
Hi,
I use VoIPTalk as my provider and unsure of a minor issue. When people call me they get a US ring tone instead of UK. Is this a Asterisk configuration issue or one for VoIPTalk ? I am running 1.6.2.0 and IAX2 trunks.
Thanks - Phil
2005 Jul 11
2
DTMF not sending properly via IAX
I'm not sure if this is a -users or a -dev question, since the answer
probably comes down to something in the code.
I'm running the latest CVS-STABLE, and am subscribed to PSTN service
using IAX2 via Voiptalk in the UK.
I've just been alerted by a customer that the sending of DTMF from my
asterisk box to a remote PSTN user doesn't work, although it used to.
To test it, I have
2006 Dec 14
1
VoipTalk unable to accept calls at present?
I am trying to get asterisks to work with http://www.voiptalk.org 's IAX
service. I have configured asterisks as per their instructions and am
using the x-lite soft phone. When I get an incoming call the softphone
rings but the caller (from pstn) gets a recorded message saying the
number is unable to accept calls at present. Does anybody know what
might be causing this?
Thanks
2004 Jun 16
1
VOIPTalk silver service
There was some discussion on this list recently about the voiptalk silver
service. I've just had an e-mail from them saying that the price has been
reduced to 2.99 per month. However, they still only provide an 0870 number
whereas pipecall provide a local call rate 0845 number in the fee.
Chris
2004 Jun 16
1
Remote rebooting a Cisco 7940
Hi,
I have seen a couple of scripts that should be able to remotely reboot the 79xx phones, but I haven't been able to make it work for my 7940.
Anyone able to guide me in the right direction?
I am running the SIP 7.1 firmware.
--
Med venlig hilsen / Best regards
Michael L?jtnant - Systems Engineer
ZyXEL Communications A/S
Columbusvej 5 - 2860 S?borg
Tel (+45) 3955 0700 - Fax (+45) 3955
2004 May 19
1
Old sound in new call.
Hi,
I have a problem that I just can't figure out how to solve.
I start *, dial it using a ISDN phone over PSTM, to a Hisax card installed in *
I get the demo-greeting, listen for a few seconds and hang up.
I dial it again, but this time the first second is sound from where the previous call ended, then the greeting starts as it should.
Right now I have removed all codecs but codec_gsm.so
2005 May 18
2
Call forwarding...
Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...
Hi,
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based -