Displaying 20 results from an estimated 20000 matches similar to: "Why is host= being ignored in sip.conf ?"
2006 Apr 12
1
Where is the difference sip.conf - Real-time ?
I have two phones (111 and 112) on a LAN, and I have on a users site a
phone 333.
phone 111 uses sip.conf, while 112 uses real-time set-up.
111 can call 333 AND the audio is working
112 can call 333 but audio is just white noise.
333 can call 111 or 112 and audio is working.
The phones are identically set-up (just user name = phone number and
password are different)
sip.conf (for 111 - all
2005 Feb 18
1
VoIP Service Provider
Hi everyone in the asterisk community. Am new to asterisk, while doing
the installation I notice that sip.conf examples were not clear for
beginners like me so I would like to share my current working
configuration with everyone.
Swifttel.net is a new VoIP service Provider out of Georgia. Their web
site is www.swifttel.net. Currently we have service with them and it has
been a pleasant experience.
2005 Jan 30
0
One way call when the * server and phone in a local network
Hi everyone,
I started playing with Asterisk server a few days ago. So far, I only have
made it 50% work.
Here is my situation, IP phone A and Asterisk server are in a same local
network behind an ADSL router (public ip = a.b.c.d), and the * server is set
as the DMZ host of the router.
IP phone B is in another network.
IP phone A
(10916) ---> ADSL
2006 Mar 14
1
Codec Issue
Hi,
I have an issue which is kind of a catch 22 situation. I had outgoing calls
to my new PSTN provider working perfectly. Then I started focussing on
incoming calls. It seems that I can solve an error which gets my incoming
calls working but that in turns means my outgoing calls don't work. -
Strange
Anyhow I was getting an error:
Process_sdp: No compatible codecs!
And from the SIP
2007 Aug 18
1
incoming calls in SIP
Hi, when I try to call in it tells me: NOTICE[11664]: chan_sip.c:10637
handle_request_invite: Failed to authenticate user "585415198"
<sip:585415198 at 82.208.46.240> <sip:585415198 at 82.208.46.240>;tag=as18abefe8
Can someone help me out of this? I have Asterisk 1.2 on the Ubuntu 7.04.
Outcoming and internal calls functions well. Thanks
sip.conf:
[general]
callevents=yes
2005 Mar 02
1
IVR setup problems
Hi guys still have the problem to setup the IVR correctly.
I am forwarding call from ser :
if (method == "INVITE") {
if (uri =~ "sip:1[0-9]{10}@*"){
log(1, "Forwarding to Asterisk\n");
rewritehostport("xxx.xxx.xxx.xxx:5061");
t_relay();
break;
}
}
inside sip.conf
2015 Aug 05
2
Asterisk uses "Anonymous", but why?
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage" target="_blank"
2006 Apr 19
0
sip.conf codecs: ulaw, alaw and g729
Hi,
When ever I put g729 in allow for trunk the other two codecs (ulaw and alaw)
stop working and I get the frame type error for them, but g729 works fine.
I've cleared general part of sip.conf of codec info to be on safe side. If
ulaw and alaw are the only ones allowed they work fine. Asterisk shouldn't be
doing any encoding or decoding, all codecs should be passing through. Any
2006 Mar 30
0
BUG: FOP reports incorrect (duplicate) IP address until restarted
Hi,
This problem may be related to a configuration problem but I believe it
is a bug in the FOP since restarting the FOP server clears the problem.
Here is the scenario: Using AgentCallBackLogin and have four agents
logged in a call is made to one of the agents directly from an internal
phone. Okay so far. Call is hung up and the same extension is used to
call another agent okay again, no
2005 Aug 16
1
Issue with DTMF Tones - Codec Issues
Topology:
PSTN<-T1 PRI->NEAX2400<-T1 PRI->Cisco 3825<-Ethernet-> Asterisk VoIP server
When I make a call to a VoIP user from the PSTN, the call gets routed
through the PBX, and Cisco. Because of that the DTMF tones are passed
inband, which I can hear on the VoIP end of the call. However, I have
one extension on asterisk set up so that I can check voice mail when
away from my
2006 Jan 18
2
CALLERIDNAME/CALLERIDNUM Deprecation
Previously, when I wanted to forward to incoming callerid when I
forwarded a call to another number I had to set the callerid on the
outgoing call to be that of the incoming number. So today I do this:
exten => s,n,Set(CALLERID(name)=${CALLERIDNAME})
because I want the outgoing callerid that I forward to not be the normal
callerid of the local extension but I want to forward the incoming
2013 Sep 18
2
sipgate outgoing calls
Hello
i am trying to setup sipgate gateway
i can get incoming calls fine, but when i dial in and then try to dial
out i get this in asterisk command line
-- Called 01179248615 at sipgate
[Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
handle_response_invite: Failed to authenticate on INVITE to
'"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1'
--
2015 Jul 29
3
Windows Asterisk Help
Hi All,
Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7.
Here is my sip.conf
[general]context = demo ; Default context for incoming callsbindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)srvlookup = yes ; Enable DNS SRV
2007 Oct 02
0
Supervised call transfer problem
Hi all,
I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered to an external SIP carrier (sip.uni.it)
If a call reachs Asterisk through the SIP carrier, then it is forwarded to the external SIP proxy extension (530 at weboffice.dyndns.org), when the extension 530 that has answered the call tries to transfer the call to another extension (513 at
2010 Nov 13
0
problem registering to ekiga.net
Hi!
I want my PBX to be reachable at my ekiga.net account. It seems I am
registered:
vajna2*CLI> sip show registry
Host Username Refresh
State Reg.Time
ekiga.net:5060 magwas 585
Registered Sat, 13 Nov 2010 13:48:22
However when others try to call magwas at ekiga.net, they find me unavailable.
My asterisk
2003 Dec 12
1
simple question on sip.conf
Hi folks,
I want to fix hole in my asterisk set up.
I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN,
Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go
'other' places. This senario works fine.
Now the issue is someone else running a vocal or another SIP proxy can
redirect his calls to my * as well. Those calls two will come through
general
2005 Feb 20
0
SIP to SIP calls have no audio until put on hold and taken back off
A previous poster mentioned the same thing, with no response:
http://lists.digium.com/pipermail/asterisk-users/2004-
December/080161.html
Fresh asterisk 1.0.5 install on FC3, started with "make samples",
nothing fancy. It's so bland, I'm surprised the list isn't full of
people having the same trouble.
I have several Uniden UIP200 phones and a single Grandstream BudgetTone
2010 Feb 08
0
Help with iax.conf {tesco|freshtel} 1.6
I have something going on that I don't fully understand after a weekend
of looking for answers.
I have an iax account with Tesco that works flawlessly with the Zoiper
client - but is giving me trouble with inbound calls in Asterisk 1.6.
After some playing I have ended up with an iax.conf file that looks like
this:
[general]
calltokenoptional = 77.75.0.0/255.255.248.0
maxcallnumbers = 16382
2004 Jun 25
2
Can one send CLID NAME over PRI?
Is it possible to send CLID NAME on a PRI?
The numbers we send out are being received by telco and propagated,
but the names we send out are not showing up.
Is this a feature in PRI? Do we need to set PRI_NET instead of PRI_CPE?
Is this just not possible? Is this a telco config issue?
Thanks for your help... I've read voip-info, and various other sources, and
search engines, and google...
2007 Mar 29
2
sip: failed the authenticate on INVITE
I've got a problem with a SIP Account I am trying to dial in with. The
correct extension rings but when I pick up the call is not made and I
get a busy signal. Dialing out works just fine - just calling this
number doesn't seem to work.
Any pointers?
Thx
Michael
excerpt from sip.conf:
[general]
context=default
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0