Displaying 20 results from an estimated 30000 matches similar to: "SIP Call transfer with RTP transfer as well?"
2007 May 11
1
Need a RTP/SIP Proxy to be used as SBC (Session Border Controller)
Hi all,
I have been using asterisk to do such kind of thing,
But I must admitt, this is not 100 % conveniant (Mainly because Asterisk
isn't a SIP Proxy).
I just wanted to know if you knew/used some kind of SBC or packages which
would deal both with SIP AND RTP !
SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ?
Any tip, info greatly welcome !
Thanks,
JM
2005 Mar 02
1
IVR setup problems
Hi guys still have the problem to setup the IVR correctly.
I am forwarding call from ser :
if (method == "INVITE") {
if (uri =~ "sip:1[0-9]{10}@*"){
log(1, "Forwarding to Asterisk\n");
rewritehostport("xxx.xxx.xxx.xxx:5061");
t_relay();
break;
}
}
inside sip.conf
2006 Jan 18
1
Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds
Hi all!
This is my VoIP network scheme
H323EndPoint ----- --- GW H323/SIP-IN
-- -- SIP Phone
| |
(Sipquest) | |
|
| | |
2005 Jan 13
3
SER vs Asterisk for SIP
Why is SER considered a better SIPserver than asterisk , why is it that SER
can handle more clients than asterisk can. And if this is just cause of say
poor SIP handling code in asterisk then is there anything being done to fix
it. Just wanted to know why SER claims to be better than asterisk as a SIP
server. ?
--
regards
Vikram (http://www.vicramresearch.com)
2005 Sep 09
1
ASTCC speaks and cut RTP channel
Hi list.
I have a fine running Ser+Asterisk environment and have just installed
ASTCC. It?s working fine either, including its caller-id authentication
feature (the one we pass the card-number as CALLERID variable and
number-to-dial as EXTEN variable).
The issue, a great one, is that when the credit is about one minute to
end, the ASTCC prompt gets into the call, says that "you have one
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
2005 Mar 01
1
Some asterisk ser problems
I have some simple questions and i need your help guys.
I have ser server which working fine, between users.
I am trying to add some more features to the ser. Most important is the IVR.
I installed Asterisk and i am trying to register user in asterisk with no success.
Part of ser.cfg file where i am trying to redirect the call to the asterisk.
2007 May 24
3
Urgent: DTMF does not work with rtpmap:101 telephone-event/8000
Hello asterisk-users list.
I have been scratching my head for almost a week. We are trying to set a
service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working.
In our scenario the SP is sending call to our ser server and ser is
forwarding the call to asterisk. In the asterisk debug I can see the
DTMF keys are coming but ivr does not recognice those keys at all. I can
see this in the
2005 Aug 08
1
Call forward & SER as SIP router
Hi,
I'm trying to transfer an incoming call from the PSTN to another PSTN number through a SER - Asterisk system. SER doing only routing..
pstn call-> SER -> asterisk (call forward) -> SER -> pstn
Logic for SER: If something comes from the pstn, send it to asterisk. If something comes from asterisk, send it to the pstn.
Every time I am getting a "Got SIP response 481
2006 Jun 01
1
connecting asterisk to pstn help
Hello Masters
Here i going explain what Iam doing and where i need help ..
Iam running Sip Express Router ,Asterisk, on same box (for
testing) my Sip express router is working fine and i can accept global
register requests with valid account and in front of Sip express router
(SER) Iam using Mediaproxy-1.4.2 which is handler to rtp/rtcp streams
between nated clients
2006 Dec 18
1
MWI, Realtime SIP, Voicemail and Extensions, UAs registered with SER
I have the following setup:
- UAs registered with SER/OpenSER
- SIP peers (non cached), extensions, voicemail setup (not message storage)
defined in Asterisk 1.2 using Realtime
When a message is left in the user's mailbox, no Notify message is sent to
SER.
1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then
the notfy is sent to SER.
2. If realtimecache=yes is set in
2014 Oct 07
1
Grandstream GXP2160 + SRTP
Hello,
I am trying to setup a Grandstream GXP2160 IP-phone with secure calling
(SRTP).
Secure signaling SSIP for registration is working great !
I follow this guide :
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
But when I try to make a call with SRTP, I get stuck. There is an
initial INVITE which is anwered with a 401. There should follow a new
INVITE with a nonce,
2007 Aug 23
1
[Serusers] why combine ser with asterisk
Asterisk is an excellent PBX system, and makes a very good endpoint in
the SIP chain for all sorts of things -- IVR systems, voicemail
applications, automated messages, etc.
It has an extremely well-written CDR engine, so many people mesh it with
billing applications to produce accurate accounting information. It also
is fully aware of the media stream, which means it's capable of cutting
2004 Sep 20
6
SER + Asterisk
Hi there,
I've seen people using SER with Asterisk. I took a look at SER
website, and I didn't see the point in using it, since Asterisk
already handles SIP very well (apparently, at least).
But, as I'm starting, and some of you (more experienced) use it, I
know that there's something there... So I would like to know why to
use SER. Is it because of scalability, performance,
2004 Aug 15
0
Sip to Sip Calls via Asterisk
Hi All,
I have a weird problem. I have asterisk setup using the G729 Codec to
receive Incoming calls both from a SIP Gateway (SER and Quintum) and via
ISDN using i4l and have rules setup in extensions.conf for sending calls out
either back via the SIP Gateway or ISDN. What I want to do is have PSTN
calls come in via the SIP Gateway, be answered by the auto-attendant and
then sent back out to
2004 Oct 07
2
Asterisk ---- SER ----- GAteway and Reinvite
Hi,
i'm using * with SER and a cisco 3725 as Gateway.
I noticed that the reinvite is not working if i use SER and if i don't use IT
(*---->Gateway) the reivite works so the * server is able to let the RTP
direct from gateway to SIP Clients.
Do you know in which way can i let it work with the SER too.
Becouse i need SER to manage other VOIP communities but if i'm not able to use
2003 Oct 15
1
SER vs STUND with Asterisk..
One for the gurus..
I have seen there has been a lot of discussion about using SER with
Asterisk.. This to me seemed like an over kill becasue it would
basically be doing most of what Asterisk is doing anyway unless you
create some weird and wonderful config in SER..
Anyway, I decided to go and have a quick read through the SER docs and
in the section about NAT they say that the best way to
2005 Feb 08
1
SER Interaction: Agents and Extensions
Hey gang,
I'm trying to work out all possible scenarios using SER & Asterisk in our
upcomming deployment. The example scenario is 50 different customers, all
with different numbers of SIP UAs. All UAs would register with SER; This
will help keep any inter-office conversations off our bandwidth since SER
doesn't handle the RTP stream.
Calls from PSTN to UA are easy to handle.
2007 Oct 29
1
SER / Asterisk and mediapath
Hi,
I'm trying to have a SER machine send calls to an Asterisk server
working as an IVR. I was able to do this part just fine. Also, when
the caller makes certain options in the IVR, the call is then
transferred to an extension via SER. This part is also just fine.
However, I'm trying to get Asterisk out of the media path once the
caller has made a selection in the IVR. Can
2003 Nov 06
2
Asterisk and SIP Proxy on same machine?
Hi
Is it possible (or recommended) to run both Asterisk and
say SER on the same physical machine? How about port conflicts?
Maybe the easiest way is to change the default SIP port on Asterisk?
But how will that work if I register some SIP accounts directly
from asterisk (like my SIP provider) but then wanna dial outbound
pure SIP calls via my SER... Has anyone got a functional system like
this