Has anybody succeeded in making outbound/inbound SIP connections to
voip.net (or broadvox, which voip.net is just a reseller of)?
I can make calls fine through their ATA, but my control panel password
doesn't seem to be my SIP credential.
- a
--
PGP/GPG: 5C9F F366 C9CF 2145 E770 B1B8 EFB1
Try as I might, I can not get incoming calls from ViaTalk to match
against my user entry. I have both peer and user entries, and incoming
and outgoing calls work, but incoming calls do not move to my in-viatalk
context (they stay in the default context.) Has anyone else managed to
get this to
Hi,
I have been trying to configure my Asterisk to use a Sip provider for
out and incoming calls.
I only have one user and password for connect to my sip provider.
My sip.conf is:
[general]
;disallow=gsm
;allow=ulaw
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0
Does the registration show up?
try sip show registry at the CLI
also try sip debug peer sip_proxy and post the result.
Might be able to see what's going on there...
mark
On 7/1/05, David [EMAIL PROTECTED] wrote:
Hi,I have been trying to configure my Asterisk to use a Sip provider forout and
Try two different entries:
sip.conf:
[general]
;disallow=gsm
;allow=ulaw
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
callerid=No CallID
register =
Thanks to every body for the solution.
It works fine!! :D
El Viernes, 1 de Abril de 2005 06:02, MF Hulber escribió:
The way it works with my provider is that although both numbers enter
the same context, each number will match its own extension. If I have
two numbers: 11 and
Hello.
I have two hired pstn numbers with the same voip provider.
I want to distingish in the sip.conf file, what of two phone numbers was
dialed, but i don't know how to do the match, because the sip client and the
sip host are the same for both numbers.
How can i match in sip.conf by the
Pepe Aracil wrote:
How can i match in sip.conf by the (TO: ) header in sip negotiation?
When you register with your provider, add /extension suffixes to the two
register = lines, which will direct the incoming calls to different
extensions in your incoming context.
The way it works with my provider is that although both numbers enter
the same context, each number will match its own extension. If I have
two numbers: 11 and 22 it works as follows:
[sip-in]
exten = 11,1,Noop(First number dialed)
exten = 22,1,Noop(Second
FWIW My ITSP sends all calls to *any* of my numbers to the extention of
the first registered one.
So even though I have:
register = xx:[EMAIL PROTECTED]/exten1
register = yy:[EMAIL PROTECTED]/exten2
register = zz:[EMAIL PROTECTED]/exten3
calls to any of the numbers go to
Hello. I'm new in the list and sorry for my poor english :)
I have this two entrys in the sip.conf file, one for incoming calls (vtele_in)
an the other for the outgoing calls (vtele_out)
-- piece of sip.conf ---
; entry for incoming calls
[vtele_in]
type=user
context=sip-in
host=voztele.com
Try merging both and use type=friend
Julian.
On Sun, 13 Mar 2005 21:07:06 +0100, Pepe Aracil [EMAIL PROTECTED] wrote:
I only can get outgoing or incoming calls work well, but not both.
How can i solve this problem?
___
Asterisk-Users mailing list
Hello all,
I'm a newbie in * and i want to start by making
internall calls between ip phones (Grandstream BT100,
and HT286),
if someone can help me with an ewample of sip.conf
file specially with the register field in [general]
defintion.
Thanks
Découvrez
www.voip-info.org
On Fri, 21 Jan 2005 11:30:51 +0100 (CET), ihsane moutaib
[EMAIL PROTECTED] wrote:
Hello all,
I'm a newbie in * and i want to start by making
internall calls between ip phones (Grandstream BT100,
and HT286),
if someone can help me with an ewample of sip.conf
file
i have tried to connect my asterisk server to vonage like this:
Sip.conf:
register = 1yournumber:secret@atlas-east.vonage.net:5060
[vonage]
type=friend
username=1yournumber
secret=secret
host=atlas-east.vonage.net
port=5060
allow=all
maxexpirey=15
dtmfmode=inband
fromuser=1yournumber
Hi,
Is there some parameter that I should pay attention
to when using externip parameter on sip.conf?
I ask this because after using sipsak I've noticed
maybe the reason that i don't get voice in one direction could be because at the
SDP/SIP messages are references to the 192.168.1.50
- Original Message -
From: Kristian Kielhofner [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, November 29, 2004 11:46 PM
Subject: Re: [Asterisk-Users] SIP.Conf help? (srvlookup)
Todd Duffin wrote:
I don't get it...I
I am trying to get FWD to work with SIP (Since IAX has
become very flakey)
sip.conf
[general]srvlookup=yes
register =
XX:password@fwd.pulver.com/XX
[FreeWorldDialup]context=sip-intype=userhostname=fwd.pulver.cominsecure=very
Todd Duffin wrote:
I am trying to get FWD to work with SIP (Since IAX has become very flakey)
sip.conf
[general]
srvlookup=yes
register = XX:password@fwd.pulver.com/XX
[FreeWorldDialup]
context=sip-in
type=user
hostname=fwd.pulver.com
insecure=very
[fwd-out]
- Original Message -
From: Kristian Kielhofner [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, November 29, 2004 11:17 PM
Subject: Re: [Asterisk-Users] SIP.Conf help? (srvlookup)
Todd Duffin wrote:
I am trying to get FWD
Todd Duffin wrote:
I don't get it...I literally cut a pasted that in (and fixed the
UN/PW's)...and same issues. I can ping or dig fwd.pulver.com just fine,
so I know it isn't an issue with my machine trying to resolve the host?
Does anyone have any idea's?
Todd
Todd,
What does sip show
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] sip.conf not paying attention to
allow/disallow
In my sip.conf, under general I have:
disallow=all
allow=g729
allow=alaw
allow=ulaw
Then I have a specific sip:
[RNK]
clip
disallow=all
allow=alaw
allow=ulaw
allow=gsm
If I do
:[EMAIL PROTECTED] On Behalf Of
Matthew Boehm
Sent: Tuesday, 23 November 2004 5:46 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] sip.conf not paying attention to
allow/disallow
In my sip.conf, under general I have:
disallow=all
allow=g729
allow=alaw
allow=ulaw
Then I
In my sip.conf, under general I have:
disallow=all
allow=g729
allow=alaw
allow=ulaw
Then I have a specific sip:
[RNK]
clip
disallow=all
allow=alaw
allow=ulaw
allow=gsm
If I do this:
exten = _9.,1,Dial([EMAIL PROTECTED],60)
The call still goes out as G729 even though I've told the RNK to
Boehm
Sent: Monday, November 22, 2004 4:46 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] sip.conf not paying attention to
allow/disallow
In my sip.conf, under general I have:
disallow=all
allow=g729
allow=alaw
allow=ulaw
Then I have a specific sip:
[RNK]
clip
disallow=all
allow=alaw
allow=ulaw
]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Boehm
Sent: Monday, November 22, 2004 4:46 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] sip.conf not paying attention to
allow/disallow
In my sip.conf, under general I have:
disallow=all
allow=g729
allow=alaw
allow=ulaw
Then I
: Re: [Asterisk-Users] sip.conf not paying attention to
allow/disallow
It's not a bug. You are setting global parameters. When you do that, it
overrides the peer settings. Try setting for each individual peer. If
you
have many peers (100 or more), use a database solution such as RealTime
Title: Re: [Asterisk-Users] sip.conf extensions.conf
Hi, my sip.conf and my extensions.conf :)
I hope it's useful
**SIP.CONF**
[general]
port = 5060 ; port to bind for sip connections
bindaddr = 0.0.0.0 ; ip to bind for sip connections
context = default ; default context for incoming sip
I have an asterisk server(x100p wildcard) that function as a gateway.
I have some local soft phone (for example 3) and I want:
1- call from one internal softphone the other internal softphone
2- call out on the PSTN from internal softphone
3- call out on the sipphone.com
4- receive call from
Am too in same situation...but inmy case even 1,2,3 are not working. Can
you send you .conf files ?
thanks,
anand
On Fri, 5 Nov 2004, Mauro Locatelli wrote:
I have an asterisk server(x100p wildcard) that function as a gateway.
I have some local soft phone (for example 3) and I want:
1-
Hi Mauro, can u send me yours sip.conf and extension.conf ?
Ciao Gianluca
- Messaggio originale -
Da: Mauro Locatelli[EMAIL PROTECTED]
Inviato: 05/11/04 16.47.23
A: Asterisk Users Mailing List - Non-Commercial Discussion[EMAIL PROTECTED]
Oggetto: [Asterisk-Users] sip.conf
Now I can't because the *.conf file are at work..
Monday I send all:)
Mauro
-Messaggio originale-
Da: [EMAIL PROTECTED] per conto di Anand S. Katti
Inviato: ven 05/11/2004 18.40
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [Asterisk-Users] sip.conf
Ye monday morning I send all:)
Ciao Mauro
P.S.i have all at work..
-Messaggio originale-
Da: [EMAIL PROTECTED] per conto di Fares Gianluca
Inviato: sab 06/11/2004 0.07
A: [EMAIL PROTECTED]
Oggetto: RE: [Asterisk-Users] sip.conf extensions.conf
Hi Mauro, can u send me yours sip.conf
Sorry to be a pain reposting this question, but a mail server problem
resulted in my last message missing the boat slightly:
How do I associate a SIP entity with a registered account on a PSTN
gateway?
I have 2 register lines and 2 entities in sip.conf.
When I dial into asterisk from the PSTN
Hello
I have this in my
sip.conf ... it's a cisco which I authenticate on it's IP adress
this works..
asterisk authenticates sip calls from this IP as user 1234567 and uses the right
context,
only, the CALLERID
is the ip adress e.g. 19216801 instead of the callerid which i try to set
try is with just
"callerid=1234567"
- Original Message -
From:
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 25, 2004 8:33
AM
Subject: [Asterisk-Users] sip.conf user
with defaultip= works butcallerid not settable (= ip)
He
PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] sip.conf user with defaultip=
worksbutcallerid not settable (= ip)
try is with just callerid=1234567
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL
I trust you are stopping and restarting asterisk between changes?
Try commenting out the defaultip line next.
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 25, 2004 9:06 AM
Subject: RE: [Asterisk-Users] sip.conf user with defaultip
, October 25, 2004 3:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] sip.conf user with
defaultip=worksbutcallerid not settable (= ip)
I trust you are stopping and restarting asterisk between changes?
Try commenting out the defaultip line next
]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Monday, October 25, 2004 3:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] sip.conf user with
defaultip=worksbutcallerid not settable (= ip)
I trust you are stopping and restarting asterisk
= line
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Monday, October 25, 2004 3:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] sip.conf user with
defaultip=worksbutcallerid
.. But the callerid still doesn't get overruled
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Monday, October 25, 2004 3:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] sip.conf user
withdefaultip
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 25, 2004 9:43 AM
Subject: RE: [Asterisk-Users] sip.conf user
withdefaultip=worksbutcallerid not settable (= ip)
PS..
If I send NO callerid at all from my cisco, asterisk translates
Subject: Re: [Asterisk-Users] sip.conf user
withdefaultip=worksbutcalleridnot settable (= ip)
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 25, 2004 9:43 AM
Subject: RE: [Asterisk-Users] sip.conf user
withdefaultip=worksbutcallerid
Hi,
After evaluate the different options,
Ihave decidedmake an attempt changing chan_sip.c to retrieve full
sip.conf from mysql database. Since Matthew may have made advances during last
weekend,It would be good have aquick report of your research.
Iwill also try to contact Ehud Gavron,
Helloall!
I am trying to load sip.conf from mysql database. I have followed the
instructions at http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers. Seems that the authentication (user psw) works fine but I would
like to get more information from mysql and I don't know how to retrieve
]
To: [EMAIL PROTECTED]
Sent: Friday, September 10, 2004 6:50 AM
Subject: [Asterisk-Users] sip.conf from mysql
Hello all!
I am trying to load sip.conf from mysql database. I have followed the
instructions at http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers.
Seems that the authentication (user
Hello,
On Fri, 10 Sep 2004 09:49:43 -0500, Matthew Boehm [EMAIL PROTECTED] wrote:
What more information? Are you talking about mailbox, nat, etc..all those
other options for SIP phones? I want to do SIP from database as well but
most of our phones are NAT and need that option stored in the
Victor Alvarez wrote:
I am trying to load sip.conf from mysql database. I have followed the
instructions at
_http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers_. Seems that
the authentication (user psw) works fine but I would like to get more
information from mysql and I don't know how to
Hi,
First of all thank you Matthew, Nicolas and
Ryan for your response.
I would like to get information like context,
mailbox, callgroup, pickupgroup, codecs... also nat! If I make the substitution
of the text file i wouldn't like to miss information in the
process.
]
To: [EMAIL PROTECTED]
Sent: Friday, September 10, 2004 12:44 PM
Subject: Re: [Asterisk-Users] sip.conf from mysql
Hi,
First of all thank you Matthew, Nicolas and Ryan for your response.
I would like to get information like context, mailbox, callgroup,
pickupgroup, codecs... also nat! If I make
Victor,
spend sometime in reading this
http://www.voip-info.org/tiki-index.php?page=Asterisk%20GUI%20phpMyEdit
it might help some.
arsal
- Original Message -
From: Victor Alvarez [EMAIL PROTECTED]
Date: Fri, 10 Sep 2004 18:44:29 +0100
Subject: Re: [Asterisk-Users] sip.conf from
I am having a problem with asterisk as a sip registrar. here is my
sip.conf file
general]
port=5060
[global]
register = 1000:[EMAIL PROTECTED]/1000
register = 1001:[EMAIL PROTECTED]/1001
register = [EMAIL PROTECTED]/2000
disallow=all;
What is the relationship between the peer definitions and the register
command? In reviewing the sample sip.conf it seems that you can place the
sip_proxY peer as the hostname. Is this correct? This
question adds the the Broadvoice thread and where to place the dtmfmode
variable.
Hello,
has anybody configured Asterisk with sipproxd (as
SIP-proxy and RTP-Proxy)?
is there an configuration-sample anywhere?
I have an asterisk server in my lan and an
linux-router (www.fli4l.de) with installed
siproxd.
If I understand right the use of the proxy is
username:
I have been doing some testing and have found issue
with certain devices and negotiating codecs in doing this Ihave noticed
something that seems peculiar to me. It seems that including allow=all
yields different results than having no disallow or allows in the sip.conf.
Could someone please
HI,
can somebody tell me how and where must I put the SIP register line? I
think is in [general] section of the sip.conf and that I have to put:
register = user:[EMAIL PROTECTED]:port/localextension
but, user and password of the SIP gateway? Because I'm trying this and
doesn't work...
thanks
I have noticed when I add dtmfmode=inband under the [general] section
in sip.conf I get flooded with warnings on the console after asterisk
answers a sip call...
WARNING[16401]: File dsp.c, Line 1106 (ast_dsp_process): Unable to
detect process 2 frames
WARNING[16401]: File dsp.c, Line 1106
I have noticed when I add dtmfmode=inband under the [general] section
in sip.conf I get flooded with warnings on the console after asterisk
answers a sip call...
WARNING[16401]: File dsp.c, Line 1106 (ast_dsp_process): Unable to
detect process 2 frames
WARNING[16401]: File dsp.c, Line 1106
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