Displaying 20 results from an estimated 2000 matches similar to: "Can't connect to voip provider over NAT"
2007 Jul 19
2
open up firewall ports for Asterisk - safe?
Right now I've been working on setting up an Trixbox server on our
internal network. Its behind the firewall, but I'd like to open up the
firewall to it because we sometimes have developers working off site and
I'd like them to be able to connect.
Is this safe to do? I've got the "Allow Anonymous Inbound SIP Calls"
box unchecked in freePBX. Is there anything else
2010 Feb 11
2
SIP RTP ports not released when channel is hung up
Hello,
using Asterisk 1.4.28, I encountered a problem with SIP
RTP port allocation.
I found some entries in mailinglist and bugtracker regarding
this issue, but only old ones.
My rtp.conf has
[general]
rtpstart=30000
rtpend=30100
so 100 ports available. I know that up to 4 ports per channel can be used
and so up to 25 channels are possible.
But even earlier I often get the error about
2008 Oct 10
2
Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
After getting some ERRORS like this:
[Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup
media stream for this call.
[Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup
media stream for this call.
[Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup
media stream for this call.
[Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports
2019 Feb 23
2
configure SRTP port range?
On 2/23/19 4:19 PM, Joshua C. Colp wrote:
> On Sat, Feb 23, 2019, at 11:04 AM, hw wrote:
>
> <snip>
>
>>
>> directmedia is not explicitly enabled; I guess it's the default.
>>
>> Joshua basically says there is no way to control which ports are being
>> used for SRTP because that it is "up the endpoint". Such endpoints, in
>>
2007 Jun 12
2
Softphone behind NAT issues
We are trying to use a softphone from a location that is behind a
firewall. We are using x-lite as the softphone.
So far, we've been able to get the phone to register with the asterisk
server, and it can receive voice from the asterisk server (IE,
voicemail, etc).
However, asterisk can't hear anything from the softphone. We have used 2
different machines to test this on. We are watching
2007 Oct 01
1
SIP trought Firewall
Hi to everyone!
I have succerfully instaled my new Asterisk 1.4 on my debian etch.
I have my users in sip.conf like this:
[200]
type=peer
host=dynamic
context=home
secret=200
callerid= 200
dtmfmode=rfc2833
nat=yes
mailbox=200 at home
disallow=all
allow=ulaw
I can make calls in my LAN but i can?t ear comunications with another client
trought Internet.
My Asterisk is in my LAN and i not have a
2010 Jun 15
2
a2billing for residential voip usage
Hello List.
I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for "?VoIP residential services"? if yes, how? if no,
2009 Nov 16
1
can't call through voip provider
Hello.
Sorry to repost this message but, I don't have the original message in my inbox nor in my sent box.
Well, last week I posted a problem I am having trying to use an asterisk server use a voip provider and a pstn. Pstn works fine but, I cant even connect to my provider's server. I don't know what I'm doing wrong.
I tried using a soft phone and I'm able to register and
2005 May 16
1
ShoreTel 210 MGCP phone drops calls with MGCP RSIP
I've got a ShoreTel 210 MGCP phone drops calls. My packet
capture indicates that the phone may be trying to renew its registration
with *, but reports Restart Method of Disconnected (frame 2), then *
seems to take that as a sign that it has lost the connection and closes
things down. The phone, meanwhile, seems to think it can continue the
conversation until a few ICMP "port
2013 Mar 18
6
Diagnosing call problem
Asterisk 11.1.0
Various soft-phone SIP clients
call center with 10-12 agents online at once using asterisk queue
Occasionally an agent will get a call (or more often a series of calls
in a row) where neither party can hear the other, or can only hear each
other sporadically. A MixMonitor recording of the call plays only the
caller - none of the agent's audio is heard in the recording.
2015 Aug 13
2
One way audio - doesn't seem to be NAT issue
Hi D'arcy
Have you checked your RTP port ranges (I'm sure you have), and also that the
server IP for RTP as specified in the initial SIP is correct?
Not sure how this will relate to your setup, but we had something similar
here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP
service provider in the middle.
We had slightly different parameters, e. g. that we would
2010 Jan 02
4
Help getting info from caller
Hello. Happy New Year to everyone.
I have a small WISP and would like to have customers to call our number to check their balance. I am planning on writing an AGI with php so it can get the customer info from the customer database. I don't know how to interact with the caller while in the agi script so this is what I have in mind:
[test-agi]
exten => 33,1,Answer()
exten =>
2009 Nov 26
1
Unable to open sound file error
Hello.
I have a question regarind sound files in asterisk 1.6. I have a sound package in ulaw format and I would like to know if I have a sip extension with allow=alaw would asterisk convert that file to the codec the user is allowed to?
I am having a problem playing a file that exist in /var/lib/asterisk/sounds/es/good.ulaw
but asterisk is telling me it doesn't. Here's what I get when
2009 Dec 12
1
how to randomly use provider?
Hello List.
I would like to know how I can use two or more service providers with asterisk to be used randomly for ei, if an user tries to make a call I would like to randomly use a provider. It doesn't matter where the call is destined to.
Thanks.
2010 Dec 10
1
Audio ports
I see in sip debug it says Audio is at port 10342
Express Talk suggests Audio at UDP 8000-8020
I've tried changing Express Talk to 10000 and forwarded 10000-10400.
Is there a possibility Express Talk won't work in the 10000 range?
Is it possible to limit Asterisk to 8000-8020?
Thank you,
Gary
2003 Sep 03
2
IAX2 ports usage
hi all !
we've got IAX2 protocol working between several Asterisk servers. Now we are concerned with doing bandwidth management to maintain an acceptable voice quality. We thought of prioritizing the udp traffic. ( Giving a high priority to those IAX2 udp ports.)
I know that IAX2 uses udp/4569. Is there any other traffic/ports that we need to consider for bandwidth shaping w.r.t IAX2.
2004 Jan 14
3
grandstream asterisk configuration
hi,
I have the following configuration:
Grandstream --> NAT (Netgear RP614)-->Internet-->Asterisk(public IP)
i can register fine and call ringing is working as good. The problem is =
i cant hear audio both ways and i get this error:
WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error:
Resource temporarily unavailable
my sip.conf file is as follows:
2010 Aug 03
1
chinaroby fxo card - never heard of them
Hello.
I'm looking to buy a FXO card to do some testing with two phone lines I have at home and was looking in ebay some and found some cheap ones but, the I've never heard of the brand or manufacturer: chinaroby. They run for about $99 plus shipping. Have any one used these? or please recommend one... Money IS an issue.
Thanks.
2015 Aug 10
2
webrtc no audio
hello,
i'm facing strange problem
asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230
person1 to person3 are behind different NATs
audio devices double checked
call from person1(chrome) to person2(chrome) works
call from person1(chrome) to person 3(chrome) - no audio on both side
(RTP flowing only in one direction)
call from person2(chrome) to person 3(chrome) - no audio on both side
2009 Dec 13
1
Unable to open file...
Hi List.
Don't know if I already posted about this problem but, if I have I apologize for the double post.
I am trying to test a time of day extension dialing 80, all I'm trying to test is if is morning I would like asterisk to say "Good Morning" but, when I run the test I get the following error message saying that the file doesn't exist and it does:
Night..............