Displaying 20 results from an estimated 500 matches similar to: "SIP Options"
2004 Sep 20
6
Newbie has a few basic questions please.
I think I am missing the whole purposes of *. i see that it can do
mainy things, but in laymans temrs I am not sure what it does.
I am very proficient in Linux and would like to use * for the
following:
1) I would like to get rid of my landline(verizon) and use voip as my
main means to communicate on the telephone. I would like to be able
to plug in my plain old phone into my linux box and
2005 Jan 18
1
No compatible codecs
Original Post
----------------
I have an Asterisk related problem with mutualphone.
I can connect to any number with any softphone that I am using (iaxcomm,
SJPhone, and a few others).
Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to
mutualphone destinations. Other destinations go fine.
A working phone call (e.g. from iaxcomm) gives the following on the
console:
--
2011 Oct 06
3
Digium FFA + Gafachi T38 outgoing issues
Hi, folks.
I'm having a heck of a time trying to get outgoing T38 faxing (I don't
need inbound right now) working with FFA and Gafachi. G711 faxing works
(as well as can be expected over the internet), but I want the higher
reliability of T38.
I'm running Asterisk 10-beta1.
When I drop my callfile in to make the call, I get this:
-- Attempting call on SIP/18884732963 at
2004 Aug 11
2
StanaPhone and Asterisks
I am trying to get Asterisks to connect to our StanaPhone so that I can use it to route my outgoing PSTN calls to. We have a free account and if I can get this working are willing to pay for an actual minutes with them.
Here is what I have in my sip.conf:
[stanaphone]
type=friend
secret=pAsSwOrD ; skewed for this message.
username=3475341914
host=sip.stanaphone.com
2005 Feb 14
5
Sipura g729 call quality to PSTN
If this has been covered before - I appologize.
We use some Sipura SPA-2000's with the g711 codec and all seems fine
(except for the occasional failure to register errors in my asterisk
logs - but I will save that for another post).
g711 call quality is on par with our Cisco 7960's. However, when
using the g729 codec, the call quality on the Sipura device goes
downhill on the PSTN side
2007 Sep 18
1
stanaphone issues. can someone verify my config?
Sorry if this comes thru twice, I had the wrong account selected to send the
first time...
Callers to the number get ringing, I get stuff in my asterisk console, and
it calls my softphone and ata, but answering either gets silence, and the
caller gets the ringing stop, if they wait ages they get the stanaphone
voicemail.
I have had the account for ages, and it never has worked, other sip
2007 Dec 17
3
VoIP service providers/PSTN termination points
Hello ppl,
Am looking at some PSTN termination providers in US. If this question
has been repeated, please point me to the correct link, as I've tried
searching the archives but have been unsuccesful so far.
I have come across quite a few companies which provide the same, such as :
Iconnecthere <http://www.iconnecthere.com>
Vonage <http://www.vonage.com>
Teliax
2007 May 01
3
Stanaphone business ok?
I see that stanaphone is not accepting new customers. Does anyone
know if they are doing ok? I have a number with them and would like
to start redirection people before it gets canceled on me if they are
having trouble....
thanks
Todd
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
2004 Aug 29
5
Broadvoice BYOD Plans - 3-way and Call Waiting
if you have anyone questions about your service you can contact us at the
support 978-418-7300
James Jones
Broadvoice Technical Support
________________________________
From: asterisk-users-bounces@lists.digium.com on behalf of Ben Wern
Sent: Sat 8/28/2004 4:34 PM
To: Asterisk Users
Subject: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting
Can anyone who is using Asterisk
2005 Feb 10
4
asterisk as sip client behind nat
Hi, I am pretty new to all of this but was able to
set up an asterisk server and have been able to
succesfully connect to asterisk with x-lite as sip client.
I have also connected asterisk to FWD (using iax2) and
to voipjet (also using iax2).
Now I am trying to connect asterisk to Stanaphone.
It has to register as a SIP client but I am not being
succesful at all.
My asterisk server sits behind a
2007 Apr 25
2
No Audio with SIP to only one provider when switching servers
I have been running Asterisk for years on a machine with a public
IP. Most recently, I have been running 1.2.17, from the day it
came out, with no (noticeable) problems.
Yesterday, I switched over to a new server that is on the same
public subnet, one higher than the original server.
I built 1.2.17 from source on that machine (as I did on the old
server). My firewall on the new machine is
2017 Feb 03
2
Spotty internet connection
On 2/3/2017 3:35 AM, Steve Clark wrote:
>
> What kind of cable modem/gateway do you have? Just wondering because
> my 12 year old Toshiba finally
> crapped out and Spectrum gave me a new one. Its and ARRIS TG1682G and
> it only gives me a private IP not
> like the old one which gave me the public IP so I can't ssh to home
> from work anymore, so I am wondering
> how
2007 Dec 10
1
T.38 fax solution, opinions?
Hi,
I'm putting together a fax solution for my company that utilizes T.38. I
wanted to throw out my plan and get some feedback if anyone is doing
something similar or sees a blatant problem with it.
We're currently rolling out SPA-942 phones for the standard desk phone with
vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls
for satellite offices are handled by
2008 Oct 27
1
CDR Records are not working
Hello Asterisk-Users,
For some reason my CDR records for disposition and billsec are not working
correctly.
I always receive a 0 for billsec and the disposition is always at "NO
ANSWER', even when I grab the calls.
I experience this with Asterisk 1.6.0.1 and Asterisk 1.4.22.
Here is information on how I do the call:
-----------------------------------------------------------------
2004 Jul 25
1
X100P Inbound Issue
Hello,
After much searching of voip-info.org and google, I'm finally giving in and asking the list.
The setup I have is this:-
Single X100P card in a Debian system
Inbound/Outbound POTS line connects to the X100P
Sipura 2000 and Budgetone 100 on the LAN
1 Cordless and one conventional phone connected to the sipura
Account on Stanaphone.com for eitherbound SIP calls.
(I have other SIP
2005 Jun 23
2
Asterisk 'losing' upstream provider registration state during small network outages.
Now that I have most everything actually working I've noticed that about
every 3-4 days on average..... and at worse... Once a day my asterisk box
seems to lose it's registered state with our sip provider and no longer will
take any incoming calls.
The caller simply hears a fast busy (reorder)
If I do a reload at the command prompt all is well for another few
days.....
What I'm
2006 Jun 12
2
How can I use my regular phones with Asterisk running on my Linksys WRT54G router?
Ok, I've done some more research and I don't think I want an FXO box...
What I'd like to do is use BroadVoice (with their BYOD plan) and then
run Asterisk on my WRT54G router. I'd also like to use my regular home
phones without having to use a special "SIP" phone... (eg. I like my
Vtech normal cordless phones)
What do I need to buy to get this working? It sounds like I
2005 Sep 29
2
Unable to send fax using BroadVoice
Has anyone had success sending faxes via a broadvoice byod account?
Everything 'looks' to go as expected, but then my fax hangs up and I get a
printout with Error 351. I am wondering if it is a codec issue or something.
Any help will be great.
Neri
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2005 Feb 26
2
Interface * with ATA from ATA FXS port?
Me again... I have service with a company that does not allow for a BYOD
plan. They will not give out credential or server info either. Is it
possible to run the FXS port of the ATA to an FXO port in *?
The service I have is throug Broadvox Direct using the Mediatrix 2102. I
have tried this using loop start and kewl start. The * box sees the
incoming ring, picks up and starts my dial plan. But