Displaying 20 results from an estimated 7000 matches similar to: "Nat & Sip & Pain"
2005 Sep 13
1
FW: Nat & Sip & Pain
Hi Ray,
I was wondering if the "qualify" option is used [in sip.conf] to keep a
connection (from the SIP phone inside the firewall to the Asterisk
server outside the firewall) open then would the firewall not allow two
way communication without incoming port mapping/NAT (providing that the
SIP phone started "talking" first)?
I'm not sure about that - I'm being
2005 Sep 14
2
STUN vs NAT Helper
I'm wondering if anyone can recommend one over the other. I'm mostly
interested in running open source solutions, so I would prefer if
your recommendations are within the open source arena.
Basically, I contemplated the idea of using SER as a NAT Helper and
possibly as a SIP server for a portion of our user base. We prefer to
have Asterisk in the mix because of the additional
2004 Oct 05
2
Dialing a # in phone number?
Hi,
I have not been successful in working out how to dial a # within a phone
number. EG:
exten => _12345,1,Dial(Zap/1/0868563823#,5,t)
or
exten => _08XXXXXXXX,1,Dial(Zap/1/${EXTEN}#)
I'm trying to append a # character so that I can use a cellsocket
(mobile phone to pots adapter) connected to an x100p. I think that
asterisk is simply ignoring the # character. The docs on
2005 Sep 15
1
USB ISDN (OT question)
Derek,
could you give me some details regarding the solar power supply you're using for your installation?
Thanks!
J?rg
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
> Derek Conniffe
> Sent: Thursday, September 15, 2005 12:28 PM
> To: Asterisk Users Mailing List -
2004 Oct 04
2
Off Topic: Dead GS BudgeTone-100
Hi everyone,
This is off topic and is for GS technical support really but it seems
that there are a lot of Budge Tone 100/101/102 users out there.
I've got a Budge Tone-100 (101 - without the extra 10base ethernet
connetion?) here. I changed the configuration through its web based
interface and I clicked the reboot link. But then something went wrong
and ever since then it doesn't
2004 Dec 10
5
Granstream phones message button
To all:
(newbie)
I have setup a BT 100 phone and mostly everthing is working pretty good
except for the message button. I have place value in the appropiate
field in the web configuration but nothing seems to work. When I press
the button the speakerphone led goes on but the phone does nothing else
(no dialtone, no sip request to *). Does anyone have this buttton
working? I would like to
2005 May 23
1
ZyXEL Prestige 2000W - cant make a call?
Hi All,
Today I got a couple of ZyXEL Prestige 2000W WiFi phones. I'm having a
problem making SIP calls although I can receive calls just fine. When I
try to make a call the phone makes some sound (like "bup bup bup bup bup
bup beep beep") and then I just hear hissing background noise (not too
loud - like comfort noise).
I upgraded to the latest firmware on the phone - Wj.00.10
2005 Sep 10
4
Fritz, mISDN, Help
A plea to all!
Has anyone had any success with two or more avm fritz pci cards with either
misdn, chan_misdn, or chan_capi, and any version of linux 2.6.x?
I have managed to get misdn to load under 2.6.13 and detect two cards using
misdn-capi and chan-capi (using capiinfo and capi info under asterisk) - but
the second card/controller doesn't answer or dial calls.
But if I try misdn
2005 Feb 04
2
How to Create customized audio file to use with ASTCC??
Hello all,
Can anyone help me out with this issue ?? I got ASTCC running, but the
audios doesn't match my needs (currency, etc.). is there any way to
create my own audios and replace the current one??
Thanks.
Daniel.
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2004 Dec 07
1
How to play messeage when user picks up the phone
Is it possible to play a message, when user pickups a phone.
For example:
press 1 to use this provider,
press 2 to use this ...
etc..
Thanks
2005 Jan 06
3
DTMF problems on phonecell
hi all.
was having problems with my phonecell connected to
wildcard fxo port. i get problems with detecting DTMF.
i have tried relaxDTMF but to no avail. i have asked
this before but would like possible causes. is it to
do with echo? problems with the GSM network? haven't
updated my asterisk for a long time. could this be a
problem that has been sorted out. please would
appreciate ur input
2005 Mar 01
1
Cisco 7940, Voicemail & DTMF
Would anyone know why Voicemail in * doesn't get the DTML keypresses
from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do
with "dtmf_avt_payload: 101" setting in SIPDefault.cnf in the tftp server?
Thanks for any help!
Derek
--
Derek Conniffe
Rivertower Ltd
DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146
Mobile: (Local Ireland) 086 856 3823
2005 Feb 02
1
Cisco 7940 [SIP], DTMF and Voicemail
Hi everyone,
I'd say this question has come up and been answered before but I haven't
been able to find it.
I have a Cisco 7940 that I've upgraded to SIP firmware (currently
P0S-3-06-3-00 - for some reason there was a failure when trying to
upgrade to V7 so I left it at V6).
The problem I'm having is that when I connect to voicemail the DTMF key
presses dont seem to work
2005 Feb 07
1
How to Create customized audio file to use withASTCC??
Hi Derek,
I'm not sure your recording will match with my needs. I wanna be able to do this myself with our currency here. Can you just tell me what to use and how to use it ??
Thanks.
Daniel.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Derek Conniffe
Sent: lundi 7 f?vrier 2005 11:59
To: Asterisk
2005 Mar 02
3
Multiple lines
Hi,
Question...
Is there a way to receive two phone calls on the same phone, or, for
example to receive a phone call, put the call in stand-by and then make
another call and finally, why not put them all together in conference...
Thanks
David Masure
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2005 Jan 15
2
IAX2 Channels & Bandwidth
Hi all,
I'm using VOIPJET to make international calls with an IAX2 connection
between my local asterisk server and their server(s).
At times I seem to have a problem if 5 or more international calls are made
at once - I'm on a 1024kbps download and 256kbps upload DSL line (only the
asterisk server uses this DSL line). Today I switched the codec from ulaw
to ilbc in an attempt to lower
2005 Sep 30
0
[Fwd: TDM40B - "Unable to play dialtone on channel X" ?]
Hi everyone,
Sorry for forwarding and top-posting this email again but its as if my
TDM40b has keeled over yesterday. After a few hours last night and
swapping the card to another asterisk server (with exactly the same
result) I needed to have the FXS ports working ASAP this morning so I
have repaced the functionality of the TDM40b with some Grandstream
handytones which I already had in
2004 Oct 01
0
S100U / wcusb Zaptel driver / Crash / Kernel problem maybe?
Hi Everyone,
I've been using Asterisk now for a few months for my small office (which
is mostly just me while other guys are always on the road so we rely
heavily on telephones) - I'm very excited with Asterisk as it can do
everything I've ever wanted to do with a PBX.
I'm having a problem with an S100U USB --> Telephone interface. I
haven't actually made it work yet
2004 Dec 10
0
SS7 to E1 & CPC
Has anyone worked out a way to transfer the Calling Party's Category
codes to Asterisk through E1 / T1 connections? I know this is normally
available on SS7 interconnects but is it also available to asterisk on
the ISDN signalling channels? (I kind of doubt that it is......)
Thanks,
Derek
--
Derek Conniffe
Rivertower Ltd
DDI: (Local Ireland) 01 201 0146 (International) +353 1 201
2005 Feb 10
0
7940 VM DTMF not detecting
Hi all,
I have a 7940 running the latest SIP firmware (V7 - thanks Doug Lytle
for the tip on the V7 firmware upgrade!).
Its almost working perfectly - I can make calls either though my local
PSTN or over VOIP but for some reason if I dial my voicemail (which is
mapped fine to the VM button on the telephone) it doesn't detect my DTML
keypresses so when I press 1 for new messages it just