Displaying 20 results from an estimated 1000 matches similar to: "[Newbie] SIP via fwd"
2003 Jun 22
3
asteisk, sip & NAT
hi
My stations are behinds a firewall, the system is windows 2000 & 98, i
use sjphone
asterisk is on the internet gateway where is the firewall Shorewall the
system is linux debian (sid) kernel 2.4.20
j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy)
to write my sip.conf but i can't call an external sip user. (an external
user can call me)
i try without asterisk with
2003 Aug 10
3
Registering SIP with FWD and ICONNECTHERE
Hi! I am new to Asterisk too, I got the similar problem and I would like to
know how to get * to work behind NAT.
When I have the SIP Debug turn on, I got the error 479 from FWD when * try
to register with FWD, it looks like * is using the local IP (192.168.x.x) in
the Contact field. I have put the nat=yes in the [FWD.Pulver.com] content,
but it does not seems to make Asterisk aware the
2003 Jun 18
1
ISDN BRI
hi
---------modem.conf :----------
msn=240862922
incomingmsn=240866365,6365
device => /dev/ttyI2
group=1
device => /dev/ttyI1 ; ttyI3, ttyI4
---------extensions.conf ;-------
[sip]
exten => _XXXXXXXXXX,1,Dial,Modem/g1:BYEXTENSION
(Sjphpone) Call to : 024076xxxx
result :
--Executing Dial(Sip/roseau-6163","Modem/g1:BYEXTENION") in new stack
-- Called g1:024076xxxx
--
2004 Jan 27
1
Cisco 7960 Problems
Has anyone ever seen these errors generated by a cisco 7960? none of our other brand phones seem to generate these erros:
Jan 27 21:54:07 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie
s exceeded on call 000ded24-d7000024-5d2ca17a-29c81cf4@65.204.176.54 for seqno 1
01 (Response)
Jan 27 21:54:08 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie
s exceeded on call
2004 Jan 08
5
Dialing the Phone from OS X Address Book with AppleScript, XML-RPC, PHP and Asterisk
I run an Apple OS X workstation and I've got a server on the same LAN
that's both a webserver and an Asterisk PBX.
I wanted to be able to originate calls in the OS X Address Book
application, and have Asterisk dial them and connect them to the phone
on my desk.
I've assembled a system that uses AppleScript to connect, via XML-RPC,
to a web application that, in turn, connects to
2003 Aug 19
1
Problem with * server and FWD
I have a small HUGE problem with *.
I have installed * but I have 2 problems.
1 - Making call to FWD.
2 - Receiving call from FWD
More info of the problem at the end.
Here is the sip.conf file.
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = sip ;default Default for incoming calls
register =>
2004 Dec 23
1
Can't Make Outgoing Call
Hi,
I can't get dial-out working. I'm trying to call 523936. Is there
something wrong with my setup here? Could someone please give me a few
pointers?
Regards,
Norman Zhang
[fwd-out]
exten => _8.,1,SetCallerID(${FWDUSERID})
exten => _8.,2,SetCIDName(${FWDUSERNAME})
exten => _8.,3,Dial(SIP/${EXTEN}@fwd,70)
exten => _8.,4,Macro(fastbusy)
exten => _8.,5,Hangup
*CLI>
2003 Dec 20
2
More beginner questions
Using DIAX softphone which seems to be working OK can get to VM/echotest etc
in the demo context
Am trying to setup FWD but get the following problems
Can hear it ringing when dialing FWD no 612 for time. Connects but no sound
from remote end.
Does anyone have any suggestions.
Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to
the internet port 5060 being forwarded to
2004 Jan 17
1
Registering multiple FWD accounts
Can multiple FWD accounts be registered?
I have the following output in my sip.conf file:
register=74928:xxx@fwd.pulver.com/74928
register=75160:xxx@fwd.pulver.com/75160
register=74573:xxx@fwd.pulver.com/74573
[fwd-74928]
type=friend
secret=xxx
username=74928
host=fwd.pulver.com
[fwd-75160]
type=friend
secret=xxx
username=75160
host=fwd.pulver.com
[fwd-74573]
type=friend
secret=xxx
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
2004 Dec 18
4
Free World Dialup and Asterisk
Hi forum,
I have been fighting days and days configuring FWD and asterisk with NO success
I have the following scenario.
My sister in Spain with FWD dialup client
My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone.
Spain LAN
FWD
2003 Aug 12
1
Using Asterisk with FWD through NAT
Hi All,
Is there any way to connect (register, initiate and receive calls) with
Asterisk to FWD through NAT? Since I own my router port forwarding is not a
problem.
I tried with
Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwd.pulver.com
but since Asterisk still use internal IP in some SIP fields I got "479 We
don't accept private IP contacts. Please set your external
2003 May 24
4
Free World Dialup behind NAT
Hi,
after reading about it on the list I decided to set up a Free World
Dialup account. For those of you who don't know, that is a sip proxy
where you and your friends can singn up free and then you can just
connect to it with any sip client and call anybody that is registered
for free. Pretty much like iaxtel (I belive that was the name of it) for
the iax protocol. It even supports clients
2005 May 13
4
1-800 with FWD
Sirs,
Thank you for your quick response.
But when i try to make a call to FWD the following error appears:
For example, when i call to 612 (a service number of FWD)
-- Executing Dial("SIP/Phone4-e85b",
"SIP/612@fwd.pulver.com|90|Ttr") in new stack
-- Called 612@fwd.pulver.com
-- Got SIP response 500 "I'm terribly sorry, server error occured
(1/SL)"
2004 Apr 28
9
chan_sip.c max number of retries?
Still getting the same error.
Apr 29 11:57:49 WARNING[1125329600]: chan_sip.c:503 retrans_pkt: Maximum retries exceeded on call 6b8b4567327b23c6643c986966334873@211.28.255.135 for seqno 102 (Critical Request)
please advise anyone!!!!!someone!!!
jai
2005 Jan 18
3
Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore)
I bought three plus two Grandstream BudgeTone 101 phones.
The shipping cost more than the phone itself from Pulver store.
The first shipping had one phone defect. Nothing on the display. (Can
happen!)
The second shipment had one phone with a defect display, but it still
worked.
The second phone's handset was defect too (microphone did not work).
Changing the handset from this one to the
2003 Dec 24
2
FWD problems
I've been having issues getting FWD to work. I posted this same Q to the
FWD forum (no responses yet), but I was hoping someone here had some insight:
http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=news;action=display;num=1072263468;start=0#0
I just signed up for an FWD account (I know I had one before, but I lost
it.. :)
I've got it running through Asterisk - all working fine
2003 Jun 25
4
Asterisk and FWD
I can't get my Asterisk to register/place calls with FWD. Here's what I have
in my SIP.CONF:
register => 11111@fwd.pulver.com/11111
[fwd]
type=friend
secret=somesecret
host=fwd.pulver.com
username=11111
fromuser=11111
fromdomain=fwd.pulver.com
I'm using CVS version of Asterisk, checked it out last week. I get
authenticate error when registering with fwd, and all my calls to
2004 May 12
1
Musical interruptions
Whilst on a call, I'm getting the following...
-- Started music on hold, class 'default', on SIP/phone3-a7d5
-- Playing 'pbx-transfer' (language 'en')
-- Unable to find extension '#' in context 'default'
-- Playing 'pbx-invalid' (language 'en')
ie - without anyone pushing keys - I hear the music on Hold - as does
the
2003 Aug 12
1
Working with FWD, IPTel, SIPPhone?
I'll admit it. I'm a asterisk newbie (but no stranger to telephony).
The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone
specials) on a private segment calling to a Linux box acting as the
segment's firewall with a leg on our public network. The phones are
setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks
to the Asterisk HOWTO).
Getting IAX