Displaying 7 results from an estimated 7 matches for "mahomedy".
2010 Oct 21
2
1 way audio asterisk 1.6
Hi
?
I ?wonder if?anyone could give some light on SIP NAT.
I've having a friken headache with SIP NAT 1 way audio.
Client - NAT? - NAT - Server
Client can hear users from server side
but server cant hear client.
?
Ive tried every possible settings
externip set
localip set
NAT= yes / route
directmedia yes/ no
?
Ive check the sip headers in the debug mode and its using the external address
in
2010 Oct 20
1
SIP 401
Hi
?
I am trying to get 2 accounts from voipblaster to talk to each other.
Calls withing voipblaster network is free. If I configure two sip clients?with
the two accounts it works fine
however with Asterisk I am getting SIP 401
?
In my Sip.conf file I?under general
?
register = user:password at sip.voipblaster.com
?
then I have a sip peer
?
?
[FreeCall](default)
type= friend
context= incoming
2015 Mar 14
0
Billing
Hi
I have the following topology:
Sever A <----> IAX2 <-----> SERVER B <----> SIP <----> ITSP
Server A : Branch Server ( with billing cdr ) ?: ?Billing module for auditing calls at branch level
Server B : Billing Server? ?( with billing cdr ) ?: Official statements get generating from this server
I am trying to match my billsecs on server A TO that of server B
The
2010 Oct 04
0
session border controller
Hi
?
I am playing around the idea of setting up an asterisk box on the public domain.
This box will then be connected to various other sip providers for LCR for low
cost calls.
It was recommened that I use a SBC.? I never used SBC. From what I understand
it can help with different aspect when it comes to SIP communication such as
security
and billing.
?
Is SBC really necessary? If so what would
2010 Nov 06
0
gigasets A580IP Recall Button
Hi
I am trying to get the recall button working for the gigasets
What settings do i need to set in the advance settings?
?
Zakir
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2010 Nov 12
0
Asterisk Sip trunking routing problem
?
Hi
?
This problem is driving me crazy.
I have two severs which are trunked by the sip
?
Asterisk box A is natted thus in my sip.conf file I use the following
externip=DDNS address
?
Asterisk box B is NOT natted and has a static IP.
Asterisk box A and B are both registered with each other.
?
B holds the address of the DDNS public IP of A
for the sip trunk.
Everything works fine.
?
The problem
2010 Dec 06
0
Fw: Sip Hangup after critical packet SIP DEBUG attached
?
HI
?
I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn)
?
Out going calls from asterisk to the ata works fine
Incoming calls from the ata to asterisk cuts off with the error msg
?
Maximum retries exceeded on transmission 70854efe-4157e3a8 at 10.168.7.103 for
seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Dec? 6 13:52:43] WARNING[3921]: chan_sip.c:3858