Displaying 20 results from an estimated 3000 matches similar to: "How to ask questions the smart way"
2008 Oct 18
2
SER + Asterisk
I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk;
but I'm not sure if this is the correct forum.
I have as DID, sip account with one VoIP provider; currently I"m using just stand alone SIP phone and register with the VoIP provider via:
stun.fwdnet.net
Is it possible to use SER to register with the provider and forward the call
2008 Dec 13
3
SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
One of the above is frequently used to front-end Asterisk.
I used OpenSER to front-end a farm of Asterisk servers and was very happy
with it. The ability to take a box out of service or to route a specific
DNIS to a box for testing rocks.
Since OpenSER has died (I don't care about the
politics/personalities/trademarks), Kamailio and OpenSIPS have risen from
the ashes. What are you using?
2008 Nov 05
1
SER/Asterisk interworking mailing list.
Greetings,
As a developer and consultant who spends considerable time on projects
involving the fusion of Asterisk and products derived from the SER
ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have
found that there is a great volume of interest in this topic on the
mailing lists associated with all communities involved, but a
comparative lack of focus that results in
2009 Feb 11
2
OPTIONS packets
Hi all,
I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is
OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy
is not replying back...The issue is the UNKNOWN username that reside in the
OPTIONS packet as you can see in the captured packets as you can see below:
1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060
2. OPTIONS sip:OPENSIPS_IP
2008 Sep 09
2
SIP to IAX?
Hi all!
I am looking for some software that would work as a proxy between a SIP
device (SIP phones and ATA boxes) and the Asterisk system running IAX. The
reason is that I can only get IAX trow the firewalls, and can't change the
settings.
One solution I am using are getting several Asterisk system communicate trow
IAX bout in this case would I rater have every persons computer act as a
proxy
2011 Mar 04
3
OT: OpenSIPS vs Kamailio -- which do you use and why?
I'm starting a new project similar to a previous project where I used
OpenSER to front a bunch of Asterisk servers.
Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely
candidates.
I'm leaning towards OpenSIPS because it's in EPEL so I can install it with
yum. Also, because I think the name sounds more 'professional' when
discussing architecture with clients :)
2009 Jan 19
3
[somewhat OT] seeking ideas/input for my thesis
Hello VoIP guys
Sorry for being somewhat off-topic. At the moment I am studying
informatics in the seventh semester and I need to start thinking about
my thesis. As I am very interested in VoIP technologies I thought about
picking this as my main topic. So far I have only little experience in
this area. I have been fiddling around with siproxd and pfSense and have
red the one or the other packet
2008 Sep 19
2
Specific SIP answers on incoming calls?
Hi,
when I still had ISDN, I was using Hangup(causecode) to send e.g. "Wrong
number" to unwelcome callers.
Meanwhile, I am only using SIP providers (no PSTN lines any more) and I
would like to do similar, i.e. send specific SIP headers. Besides "wrong
number", I would especially like to send 302 temp moved with a specified
address to deflect certain calls.
Is there any way to
2009 Jan 30
3
looking for a link or pdf ot something about opensip/openser and load balancing
hi
i need a link or something about asterisk load balancing i cant find any, i
only found a paragraf in an email
anything wiil be wolcome
thanks!
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
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2009 May 15
1
Spiral SIP Request problem
Hello,
I am using OpenSIPS to register all the users and planning to use asterisk
for Auto Attendant, Queues, Voicemail and Conference Bridge.
I have a scenario where the signaling does not happen properly:
1) A user from Opensips dials an extension 7000 which is an
auto-attendant extension. The call is routed to asterisk to play the auto
attendant messages like Welcome and Dial the
2008 Nov 20
1
Load balancing Asterisk.
Hello!
We're looking for a solution to reliably load balance our
Asterisk boxes. So far we've been using a hodge-podge of
directing different services to different boxes/IPs, but
eventually I'd like to consolidate things so we can present
a single IP address to the outside world.
My question is - how do we go about doing that? I've read
a lot of things like load-balancing via
2007 Jul 05
1
Simple CDRs w/Asterisk/OpenSER.
Suggestions on how to use Asterisk to collect CDRs from a OpenSER-based
proxy / call routing setup? I need to get simple CDRs; not for detailed
settlement/rating, but just for reconciliation with an ultimate TDM
carrier just to make sure we only get billed for what we're actually
using.
I'd use the often-heralded approach of dumping a call from OpenSER into
Asterisk and having it
2008 Jul 09
2
Asterisk dimensioning
Hello all,
I need to install asterisk for 900 sip users with 2 PRI ports.
It is posible to handle this number of calls/extensions with only one
asterisk machine?
Which is the best way to install that? two asterisk with openser. One
asterisk with openser .....
Is it necesary run a SER server on this enviroment?
Any clue will be welcomed.
Thanks in advance.
VoipCrazy
2009 Mar 06
1
Asterisk and sip router integration
Hi,
Does anyone have some good examples of a Kamalio or OpenSips
configuration that integrates with Asterisk?
Essentially I want to use the SIP router as the UA, but still run all
the calls through Asterisk (for dialplan, etc..)
I've looked for examples on the project web sites, but I haven't found
anything decent yet.
Thanks.
-- James
2009 Oct 15
3
DS3 capacity calls using asterisk
Hi All,
We are trying to implement a DS3 capacity calls (672 concurrent calls) using
asterisk server. I wanted to ask are there any compatible DS3 cards with
asterisk? I tried searching a lot but could find DS3000P from digium but
unable to get this product. Does anybody have any idea of having any DS3
card in asterisk box so as to handle around 600 calls?
Thanks
Sandesh
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2007 Apr 10
1
Maximum retries exceeded on transmission
Hello
My asterisk is receiving calls from OpenSER but all calls hangup in 20
seconds.
This only happens because Im using Asterisk2Billing's AGI (without
A2Billing it doesnt hang up).
does someone knows whats the problem??
Here is my Asterisk debug:
(xxx.xxx.xxx.xxx -> the phone's IP)
Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread:
Unable to spawn mp3player
Apr
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 Aug 12
3
Asterisk + CDRTool
Hello
Anyone who have already use/configure Asterisk with CDRTool ?
Or maybe can suggest another CDR GUI ?
regards.
Harry
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2008 Feb 12
3
LCR in Asterisk
Hi all,
I am planning to implement LCR routing on my already running asterisk
server. Uptill now i have found out that asterisk has no support for lcr, i
have to do something about it myself, for example using the AGI. Im looking
for ideas here. Whats the best way to start implementing lcr in asterisk.
Should i use agi and start implementing my own lcr script or is there any
plugin available which
2007 May 11
1
Fwd: SER as a Session Border Controller
I am curious if it is advisable to use implement Asterisk as a Session
Border Controller for a VoIP reseller environment. Users will terminate
calls SIP to my server, which will authenticate them via RADIUS, perform a
LCR lookup, select an appropriate trunk (based on LCR), and terminate the
call (update RADIUS accounting at end of call). All while acting as a B2BUA
to prevent the users from seeing