Displaying 20 results from an estimated 5000 matches similar to: "best practices for running asterisk as SIP B2BUA"
2008 Feb 14
1
Ser, asterisk and ip2ipgw
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<font size="-1">Hi,<br>
<br>
i use a ser, as proxy sip server(authentication), then a cisco router
as sip2h323 gw(authorization and accounting). i want to start asterisk
as sip statefull
2009 May 18
7
callcenter / dialer / predictive dialer / vicidial program is now open
This is a global message to all to announce our callcenter / dialer /
predictive dialer / vicidial program is now open.
Codecs: G711, GSM, G729, G723
Protocols: SIP
Duration Rate : 30/6 (6/6 with monthly minutes over 100,000)
Channels : 100 to start with , more on demand.
We are predictive dialer friendly , your account will not be shut off.
Contact us to do a test run.
Mike
2009 Mar 14
3
TRANSFER EVENT ON QUEUE_LOG
Hi,
Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6.
I make an attended transfer (asterisk feature), and I cant see the event.
Any idea? Should I submit a bug report?
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2008 Oct 29
4
Dimensioning a telephony system based on openser!
Hi,
I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk
1.4 + CDRTool with freeradius telephony system.
Asterisk is used only for voice mail and redirectioning calls.
Every calls should pass through mediaproxy so that i can account them.
The goal was to create a simple prototype of what could be a VoIP
provider.
Now i need to dimensioning this system to work
2009 Apr 13
2
retransmision error con asterisk 1.4.24.1
se?ores alguien le ha presentado este problema al acceder al voicemail
o al hacer una llamada a la pstn
1940> Playing 'vm-received' (language 'es')
-- <SIP/111-08d91940> Playing 'digits/yesterday' (language 'es')
-- <SIP/111-08d91940> Playing 'digits/at' (language 'es')
-- <SIP/111-08d91940> Playing
2008 Oct 27
11
Fring: Open VPN client to be installed on the mobile, which mobile?
Hi All;
I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile:
Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server?
Regards
Bilal
2007 Jul 30
5
Silly MeetMe() question.
I've got the ztdummy kernel module loaded and seem to have all the desired
prerequisites in place, but Asterisk never seems to compile with MeetMe()
application support enabled, nor does there appear to be a module I am
failing to load that would contain this application.
Is there something really obvious I am missing?
Thanks,
--
Alex Balashov
Evariste Systems
Web :
2008 Nov 10
3
directrtpsetup without reinvite
Hi,
I want to be able to bridge two sip channels using direct RTP
between my endpoints (Audio IP : not local) but without
using reinvites. So I set up my asterisk sip endpoints as follows:
[test1]
type=friend
host=dynamic
username=test1
dtmfmode=info
context=test_rtp
allow=all
canreinvite=no
directrtpsetup=yes
[test2]
type=friend
host=dynamic
username=test2
dtmfmode=info
context=test_rtp
2009 May 27
2
problem with T.38 media headers
Hi Guys,
Something I have noticed while dealing with T.38 and re-invites in Asterisk 1.4.22.
I have a provider who re-invites with the following sdp (message flow
PROVIDER_EQPMT -> ASTERISK):
"""
.
v=0.
o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER.
s=-.
c=IN IP4 CONN_IP_PROVIDER.
t=0 0.
m=audio 0 RTP/AVP 0.
m=image 26858 udptl t38.
a=T38FaxMaxBuffer:288.
2007 Aug 03
4
PRI - DS3 Calls Dropped
I have a customer installation with an Adtran DS3 mux. The DS1's go into my Asterisk servers that run IVR/Call recorders. The DS3 provider is Qwest, and they tell me that they routinely drop the DS3 service to redundant back-up's and that this is a common practice that happens thousands of times to DS3 lines daily across the US without any service interruptions. They say that the
2009 Nov 02
5
Forward DID to another server
hello all,
i have 2 asterisk boxes on that 1 have public IP Address and another is only
have local IP address
now on public IP there are some 7 DID forwarded , now i want to forward 3
DID out of 7 DID to
local machine we called server B , I know there are DIal , and Switch
statement in asterisk ,
but is there any other convenient way to do this. because if call ratio is
high then my call legs
2009 Nov 23
1
Is Answer really needed
Hi
All my incoming dial plans start of with an Answer which I now know
starts the billing time. Some of the dialplans then get forwarded out to
POTS via a carrier and so the actual amount of time that should be
billed is being distorted.
I've done a few tests this morning and found that if I don't start with
an answer then the billsec of my forwarded call is actually the length
of
2007 Jul 08
2
asterisk is not sip proxy
Hello Asteriskers,
I'm confused about why Asterisk is not a SIP proxy and why exactly
this can affect the performance of a large Asterisk system.
I know that Asterisk acts as a useragent endpoint, but my doubt is why
exactly Asterisk could overload the call flow if the RTP voice stream
goes from the caller to the called party.
Does someone know how many calls or pencentaje that could handle
2008 Dec 01
3
OT: What do you guys think of this?
http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/
FUD? Interesting? Boring? New news? Old news?
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
2008 Jul 19
2
OT Astricon/Digium Beach Ball Mailing
Just an FYI for Digium. I received a mailing today from you guys
which was nice. The price of mailing was ~$1.60 and inside was an
inflatable beach ball.
Cool, but I tried to blow up the beach ball and the the seam where the
part opens to inflate the ball was not connected to the ball
whatsoever, so it went right in the trash.
I wonder if the sick heat had anything to do with it, was mine just
2007 Jun 17
2
CNAM.
So, is there anyone out there that provides rather generic but
comprehensive CNAM-style directory services via SIP, to end-users? So
I can put names to my calling numbers?
Thanks!
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
Direct : +1-678-954-0671
2008 Jan 04
3
b2bua
Is there a way to disable the b2bua feature in asterisk.
I would like asterisk to work as a sip server and not be involved in the RTP path between phones.
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2009 May 26
8
Bandwidth management and ADSL router
Hi All;
I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX.
Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting?
Regards
2008 Aug 21
1
DSS1 vs SS7
Hi,
I am requesting for a E1 connection from my telco. They are asking if I
want DSS1 or SS7, and I am stuck here. Could someone tell me the difference
between the two? How should I decide which one to use?
Thanks in advance for your help.
Mark
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2009 Jan 14
8
evaluate SIP response codes in dialplan
Hi!
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
I have an Asterisk server which forwards requests to various PSTN
gateways with SIP. If the Dial() attempt is not successful I want to
differ at least these 3 options:
- called destination is busy (486): e.g. activate auto-redial
- called destination does not exist, unassigned number (404)
- gateway is broken,