Displaying 20 results from an estimated 3000 matches similar to: "SIP trunk problem"
2006 Mar 24
14
IAX Incoming/Outgoing
I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a
2008 Apr 04
2
SJphone behind NAT/Firewall without sound
Hi.
I need connect some LAN stations with SJphone to an Asterisk Server
published on Internet.
My Lan Clients access to Internet using a small linux firewall/proxy
server. I use the next firewall script. That is a simple script with
default policy ACCEPT, and NAT to share Internet. I can connect to
the asterisk server, authtenticate the users in the server, and dial
to any extension, but
2004 Dec 04
2
SJPhone SIP Tab
Hi,
I'm following, http://www.voip-info.org/wiki-Asterisk+phone+sjphone.
However, I cannot find the SIP tab. Would someone please give me a few
pointers? The screen capture can be seen at URL below
http://www.dslreports.com/forum/remark,12022987~mode=flat
Regards,
Norman Zhang
2005 Mar 06
3
SJphone on PDA registering with Asterisk???
I try to setup SJphone on my PDA, but it does not register with Asterisk.
I have setup a sip account on asterisk, ...
Can anybody give me a hint?
bye
Ronald
2005 Mar 22
1
Nat and firewall port forwarding - is it really required?
I have a question which I'm sure has been asked before but my research has
yet to find it.
I have Asterisk running on a Linux server but in order to get it to connect
I needed to punch a hole in my firewall on port 5060 for it to receive the
registration responses from broadvoice.
If I run sjphone as a softclient on my home PC I do not need to punch that
same hole and it works just fine.
2006 Nov 04
1
Redirect problems using IAX2 and SIP
Asterisk 1.2.7
RedHat 9.0
I frequently have the need to redirect calls that come in on a DiD
provisioned by my ITSP, back to the ITSP so that they can terminate
the call on the PSTN. For example when an external call comes in, I
often have to send it to a cell phone. I believe that this is
referred to as "hairpinning" the call.
I do this by answering the incoming call and then I use
2006 Nov 04
1
Hairpinning problems using IAX2 and SIP
Asterisk 1.2.7
RedHat 9.0
I frequently have the need to redirect calls that come in on a DiD
provisioned by my ITSP, back to the ITSP so that they can terminate
the call on the PSTN. For example when an external call comes in, I
often have to send it to a cell phone. I believe that this is
referred to as "hairpinning" the call.
I do this by answering the incoming call and then I use
2005 Jun 08
5
Xlite not communicating with Asterisk
Dear All,
I have downloaded the xlite version 2.0 for windows and I made the
following conf in the xlite itself as the document suggested in order to
make it work with Asterisk but still it doesn't work as a matter of fact
when I tried to make a tcp dump I can see no packets going between the
windows client and the Asterisk server at all, here is the my conf on
the xlite itself:
in the
2004 Jun 17
3
SJphone regestration problem - Help!
I am having a problem with SJphone registration, having read the list
and wathced it for a while for similar problems. I just can't seem to
figure out the problem.
I tryed to follow a tutorial from
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone,
but in SJphone (SIP tab), I can't find the following setting.
Use local outbound proxy - checked.
Proxy IP Address:
2003 Sep 13
2
SJphone DTMF?
Hi. I have sjphone installed on windows and working
except for dtmf. I read the docs for sjphone and it
uses inband dtmf. I configired dtmfmode=inband but it
still does not recognize it. Someone on the lists
said that inband only works using alaw or ulaw but i
tried only allowing that too but still no go. Hmm..
any other ideas? I can't get any other client to work
on windows :-/
I
2005 Oct 08
1
need help-can't not register to asterisk from softphone
Dear all expert,
(i posted this question one time, but i couldn't reach the answer
-so allow me to post here)
1)I download asterisk realse version 1.2 beta1.
After that i compile it successfully and run it with:
asterisk -vvvc
2)I follow the instruction in
http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html
in sip.conf:
i add two account:
[ivan]
type=friend
username=ivan
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",
2016 Nov 21
3
Asterisk 13.12.2 : strange queue behaviour
On 21-11-16 15:17, Matthew Jordan wrote:
>
> On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
> Hello
>
> when using Asterisk version 13.12.2 I notice that it takes up to
> 30 seconds (sometimes even longer) for a call queue to call its
> members.
>
>
2004 Jun 21
1
Siemens Optipoint 400 SIP Problem
Hi there,
I tried to get a few "Optipoint 400 SIP" working with *, but it refused to work properly.
In my testing-network i have two Sjphones (they are working really fine) and
three optipoints.
I?m able to dial the number of a Sjphone on all of the optipoints.
The call is signalled at the Sjphone with the right number of the optipoint and an connection can be established.
But when I
2005 Jan 17
2
Offtopic: improving softphone latency on Linux?
Hi folks
last weekend, I tried Windows Messenger first time and was stunned by
the little latency it gives. Until now, I've been using softphones on
Linux exclusively, like iaxcomm, linphone and sjphone, and they all give
me about 1, at times even 2 secs delay. Whereas Messenger really seems
to be in the millisec range.
Of course, I'm now curious why there is that difference. Clearly,
2004 Dec 01
1
SIP expiry time
Hi,
I notice that SJPhone is registering to asterisk with an expires of 120
secs. However, when I invoke the command "sip show peer [sip id]". I notice
that the output indicates the expires 427 and the expiry is 900. Can someone
explain these numbers to me?
I also notice that just before SJPhone re-register, when I try to make a
call to the SJPhone, asterisk will complain that
2004 May 30
11
New Firefly version
As Promised, I've released a new version of Firefly (ver 1.8) with IAX &
SIP support back in.
Get it from Virbiage site or here's the direct link
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
If it crashes on startup, export your Firefly tree from the registry
(current user -> software -> firefly), then delete tree from your
registry. If that fixes it, send
2003 Jun 20
1
[HS] results testing asterisk with ISDN BRI & look for help to understand configuring SIP with asterisk
configuration
ISDN BRI
card : ISDN Olitec PCI 128 (hisax gazel)
internet connection by ISDN 64kb/s
dynamic IP
nom de domaine registered to : dyndns.org avec ddclient to register IP
par ddclient
asterisk (on internet gateway)
configuration pour ISDN BRI par virtual modems /dev/ttyI* (modem.conf)
logical telephone SIP "SJPHONE" on 2 local stations "windows"
(i don't succeed
2003 Jul 01
2
Today's Message from linphone; update on Khpone and SJPhone and X-Lite
Today's "frustrated programmer" award goes to Linphone, which has the
following debug output:
> (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer!
I get this message when I connect to linphone using a softphone, or when
I try to use linphone to connect to asterisk and listen to an
announcement. I suspect that
2003 Dec 16
2
DIAX-SJPHONE REGISTRATION PROBLEM
I am having a problem with softphone registration, having read the list and watched it for a while for similar problems I just cant seem to figure out the problem. Using SJPHONE or DIAX , I can make outgoing calls but I can't get them to register with asterisk, I have other sip devices registering OK-7940's. From the list and the digium web site this seems to be a straight forward set up