Hello,
Running FreePBX Distro (v2.11) and getting[2014-02-16 12:47:23] WARNING[2425] chan_sip.c: Unknown insecure mode 'very' on line 428 warning in logs:
http://wiki.freepbx.org/display/ST/Setting+up+SIPStation+manually+in+FreePBX states to set insecure=very but from the warning it does not seem to like this.
Please advise if insecure should actually be =port,invite in FreePBX (or regardless of warning leave at =very).
Thanks
hi my dear all numbers. i am new in here. so if i do any mistake plz forgive me. i want help from all.
i install freepbx in my Ubuntu 12.04 lts desktop version. everything is ok.but the problem is i can setup vpn with free pbx. as because i want to terminate calls for voip. i have a soft-switch. i have no real ip address in my freebpx. so i need vpn for ip address for add ip with my switch. but i cant understand how i do it. can anyone plz help me.
Hi Team,
I have Elastix Installed and have 2 outbound trunks configured , i am able to dial with out any issues from 1st trunk with out any prefix , the second trunk i am using with prefix 9 and while dialing out from this trunk the call was taking more than 10 seconds to connect ...
I need this trunk to connect immediately , can any one help me on this.
Thanks in Advance.
Hari
Hi,
I have two freepbx servers and i was able to connect the two servers.
Scenario: Every calls to my 1st server was redirected to my 2nd server.
Problem: My problem is if i call the queues on my first server from 2nd server the calls are redirected again to second server.
How can i put condition on first server that if the calls come from second server no need to redirected it back to second server?
Here's my peers detail on first server:
username=532
type=friend
secret=secrect
qualify=yes
nat=yes
insecure=very
host=192.168.10.115
dtmfmode=rfc2833
disallow=all
allow=g729&ulaw&alaw&gsm
context=from-trunk
Hi Team,
I have configured SIP trunk and able to make outbound calls , while i am dialing some nos the calls are getting blank in my phone after 15min and i can see the trunk was live , but i am unable to here any voice , i am using grandstream phones and i am facing issues with all my phones.
Below is my outgoing trunk configuration and i need deadly help on this , this issue was not happening for all nos .
type=friend
nat=yes
host=x.x.x.x
fromdomain=x.x.x.x
context=maincontext
dtmfmode=rfc2833
disallow=all
allow=alaw
insecure=invite
Thanks
Hari
Hi Everyone,
I have configured asterisk server with a2billing in Debian server,when I try to add sip trunk from extension.conf that's working fine.But In the case of a2billing its showing some issues,I have added the same sip trunk via a2billing and I make a call,at that time the sip trunk call showing like :
=======================================
Using SIP RTP CoS mark 5
-- Called
SIP/x.x.x.x/91234567890|60|HRrL(10140000:61000:30000)91234567890
=======================================
Problem is "|60|HRrL(10140000:61000:30000)91234567890" this entries appending to every call, so sip trunk providers rejecting the call, how can I remove this entries from the call?
Please help me
We have multiple accounts with Bandwidth.com, and, up until recently, they have all been retail accounts. To transfer one of them, we need to send and receive traffic from their wholesale trunks, but all I can get from their test numbers is a "I'm sorry, this number is not in service." I've done the following so far: Duplicated the Bandwidth.com retail trunk that was already working. Changed host and outbound proxy to the next IP. Security is right, because we are giving each other access. The phone call is being received, there trunk is enabled for outbound and inbound, but somehow we're still getting this error. Does anyone know what else to check to see where this error is coming from and how to fix it? Here is a call trace we had setup:
(INFORMATION REDACTED BY MODERATOR PLEASE REPOST WITHOUT YOUR PUBLIC IP ADDRESSES AND OTHER INFORMATION YOU DON'T WANT THE WORLD TO SEE)
Dear,
I installed AsteriskNow then bought a Skype number and I configured successfully. Now I have 3 channels, in theory, my asterisk can receive 3 simultaneous calls at a given time. But the problem is: If I have 3 phones and I click call buttons on 3 phones at the same time (or almost that), one of them will receive 'line busy' message and 2 others get connected to my asterisk. But if I click call buttons on 2 phones at the same time, after 2 these phones get connected to my asterisk server, I click call button on the third phone, it also gets connected. This means that, at that time my asterisk is receiving 3 simultaneous calls. So, what happened?
Thanks!!!
Dear all
I installed elastix and configured. Everything working fine. But I found that one extension (or user)can login simultaneously from different devices. For example. user1 can connect to server using Laptop and same time from mobile device or any other devices.
I want to limit number of connection to the client. Client should connect only one device.
How to configure it ? Any suggestion ?
Thank you in advance.
Hello.
Is there a way to setup an inbound route depending on the TRUNK they called in on instead of DID Number?
Thank you!
Hello,
We have one pri line and grandstream gxp1400 instrument on which pri is configure and working but now we want to configure another line on same instrument for VOIP with different provider.
i had taken tata telecom MVOIP service but i dont understand how to configure this setting in freepbx as well as on grandstream 1400 instrument .
tata mvoip seeting to be configure on freepbx server.
TCL Signaling IP: 202.54.112.194
TCL Media IP: 202.54.112.197
Protocol: SIP
Dialing Pattern: 00
Supported Codec’s: G.729
DTMF Type: RFC 2833
Registration: No. Authentication is IP based.
Calling Number (ANI/CLI): It should be 10 or 11 Digit (E.164 format) valid ANI number.
please help me .
regards
wasi
Hi everybody,
I have a serius problem about trunk connections.
My sip provider blocking my server ip for connection attack.
I Checked full log file but i cant find reason why my asterisk doing connection attack.
My version is:
Asterisk 11.7.0 built by palosanto
[2014-04-07 04:05:04] NOTICE[2707] chan_sip.c: Peer '999' is now UNREACHABLE! Last qualify: 844
[2014-04-07 04:05:24] NOTICE[2707] chan_sip.c: Peer '101' is now UNREACHABLE! Last qualify: 395
[2014-04-07 04:05:29] NOTICE[2707] chan_sip.c: Peer '998' is now UNREACHABLE! Last qualify: 3084
[2014-04-07 04:06:11] ERROR[2707] netsock2.c: getaddrinfo("sip.verimor.com.tr", "(null)", ...): Name or service not known
[2014-04-07 04:06:11] WARNING[2707] acl.c: Unable to lookup 'sip.verimor.com.tr'
[2014-04-07 04:06:11] NOTICE[2707] chan_sip.c: -- Registration for '000000000@sip.verimor.com.tr' timed out, trying again (Attempt #2)
[2014-04-07 04:06:13] VERBOSE[2707] chan_sip.c: -- Registered SIP '998' at 178.XXX.XXX.XXX:54242
[2014-04-07 04:06:17] NOTICE[2707] chan_sip.c: Peer '998' is now UNREACHABLE! Last qualify: 0
[2014-04-07 04:06:27] NOTICE[2707] chan_sip.c: Peer '101' is now UNREACHABLE! Last qualify: 2002
[2014-04-07 04:07:11] ERROR[2707] netsock2.c: getaddrinfo("sip.verimor.com.tr", "(null)", ...): Name or service not known
[2014-04-07 04:07:11] WARNING[2707] acl.c: Unable to lookup 'sip.verimor.com.tr'
[2014-04-07 04:07:11] NOTICE[2707] chan_sip.c: -- Registration for '000000000@sip.verimor.com.tr' timed out, trying again (Attempt #3)
[2014-04-07 04:08:11] ERROR[2707] netsock2.c: getaddrinfo("sip.verimor.com.tr", "(null)", ...): Name or service not knownHi Guys
i´ve a problem with my system, i hope you can help me.
I try to connect 2 Asterisk by a sip trunk. If i look at sip show peers and sip show registry everthing is ok.
This is my config:
Server2:
[xxx]
username=xxx
type=friend
secret=kjf72356kfh5
host="ip"
disallow=all
allow=alaw
allow=ulaw
dtmfmode=rfc2833
context=default
register=xxx:kjf72356kfh5@ip/xxx
server2:
[xxx]
username=xxx
type=friend
secret=kjf72356kfh5
host=dynamic
disallow=all
allow=alaw
allow=ulaw
dtmfmode=rfc2833
context=from-test-outbound
Calls from server1 to server2 work, but calls from sip1 to com1 doesn´t work.
A console dial from sserver2 tells me this:
-- Executing [1100@from-pstn:8] Dial("Console/default", "SIP/123456789/1100") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [1100@from-pstn:9] NoOp("Console/default", "es geht weiter") in new stack
-- Executing [1100@from-pstn:10] Hangup("Console/default", "") in new stack
Over a tcpdump i can see that server2 doesn´t send anything to com1.
If i change the host=dynamic from server2 to the host="ip" it works.
But the host need to be dynamic
Hope you can help me.
Thank you for your help.
MotherTrucker
Hi,
I am sorry if this question has already been asked before. I searched a lot over the google but could not find the solution. Pro'lly I didnt search with right search terms.
I am newbee to Asterisk, I have run into a problem, where I have 2 SIP trunk providers.
The problem is I cannot connect them both simultaneously at the same time. I have to disable either one of them for the other to be connected.
sipgate.co.uk:5060 N 2125892 120 Unregistered
gw1.sip.us:5060 N 5248454447 105 Registered Mon, 02 Jun 2014 22:18:51
I am behind NAT and have following port binding
nat=yes
externip=mylivestaticip
localnet=192.168.1.0/255.255.255.0
SIP.US Trunk
Peer Details
username=myusername
type=peer
trustrpid=yes
session-timers=refuse
secret=mysecret
rfc2833compensate=yes
qualifyfreq=120
qualify=yes
nat=yes
insecure=port,invite
host=gw1.sip.us
fromdomain=gw1.sip.us
dtmfmode=rfc2833
disallow=all
context=from-trunk
canrevinvite=no
allow=ulaw
Register String: username:pass@g1.sip.us/username
SIPGATE.CO.UK TRUNK
Peer Details
username=username
type=friend
secret=secret
qualify=yes
nat=yes
insecure=invite
host=sipgate.co.uk
fromuser=username
fromdomain=sipgate.co.uk
dtmfmode=rfc2833
disallow=all
context=from-sipgate
canreinvite=no
allow=ulaw&alaw
Register String: username:pass@sipgate.co.uk/username
Can anyone please tell me what seems to be wrong at my end. Would appreciate all the help.