Displaying 20 results from an estimated 3000 matches similar to: "re: help with redirect from SER"
2006 Feb 05
2
re: questions about sip requests to asterisk 1.2
hi all,
I keep asking the question and getting no replies, so i'll keep asking :-)
In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request
from SER, specifically
rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk
picks up the request and matches it to the dialplan, i.e. if in ser i was
sending to 151@myServer, it will make it
2005 May 09
1
Asterisk + SER and NAT
Hi,
We are testing a SIP solution * + ser solution for a large implementation.
All the clients are nated.
When a client is dialing outside the domain (to a FWD sip account for
example) all is perfect ! ;-)
But ,when a call is done to a sip account, the client is ringing, then the
caller can hear the nated client very well, but the nated client does'nt
hear anything. RTP issue no ?
I've
2004 Sep 08
0
re: asterisk, SER and autocreatepeer
Hi all,
quick question...i am using autocreatepeer to get asterisk to work with SER
without having to specify each UA in sip.conf and in ser separately.
2 questions:
1. obviously this is not very secure because anyone can bypass the SER
and register themselves as a peer with the asterisk. assuming i block
incoming requests on the port asterisk is running SIP on (excluding
requests from the SER, of
2004 Aug 21
0
autocreatepeer and sip peer options
Hi all,
quick question...i am using autocreatepeer to get asterisk to work with SER
without having to specify each UA in sip.conf and in ser separately.
2 questions:
1. obviously this is not very secure. assuming i block incoming requests on
the port asterisk is running SIP on (excluding requests from the SER, of
course) does this adequately protect the server from unauthorized users or
is there
2006 Apr 14
1
asterisk or ser
Hello:
I noticed in few references that asterisk and ser and complementary.
Meaning asterisk handles connections to PSTN and voicemail but SER is better
for routing SIP traffic.
Is anyone using just asterisk for production purpose. Meaning serving a high
number of callers.
Is it mandatory to use SER behind asterisk?
your feedback would appreciated.
-Gaid
-------------- next part
2005 Mar 06
1
SER -> Asterisk voicemail on busy/unavailable. Anyone did it? (googling says NO)
Hello all! I googled lists.digium.com and ser mailing list, but did
not find any working configuration of asterisk used as voicemail for
SER. This is my config
if (uri==myself) {
if (method=="REGISTER") {
save("location");
log (1, "Registered\n");
break;
};
2005 May 19
1
ser+asterisk problem
hello
I am using ser with asterisk
asterisk on 5070 (on back end)
ser on 5060 (on front end)
i am getting all requests at asterisk.
i tried by changing asterisk port
bindport=5090
but still getting all requests from sjphone at
asterisk.
can any one tell what is the reason
regrads
Kamran
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2004 Jan 29
1
re: help with voicepulse connect IAX2
hello,
after playing with an asterisk configuration for voip for a few weeks i'm
trying to get outbound dialing with voicepulse going - i've cut down the
asterisk to a very minimal install (1 SIP client) to try to localize the
problem. The SIP client works fine (SIP and * on the same NAT) and could
access the demo from samples before i removed it, and can call itself - so
i am
2004 Jan 15
1
SER & Asterisk
Hi,
I'm trying to bundle the powers of Asterisk and SER.
Asterisk for pabx functionalities and termination to landline/PSTN, and
SER as SIP Gateway/Proxy.
With my current configuration the SIP user just adds 0 as a prefix to a
number, and the call will go out to PSTN over Asterisk.
For this to work I added the rewritehostport() function in SER to
point to the Asterisk IP (different from the
2006 Jan 04
0
confusion about contexts - SER
Guess I got where your confusiion lies.... you DON'T set up the ALL contexts the users will have acces to in sip.conf, in this file you will only set 1 single context. All other contexts you want the users to have acces to, you will have to add them in the extensions.conf
So what you have to do is the following:
-user 2092, set it the createmenu context in sip .conf
- in extensions.conf
2007 Oct 22
1
app_swift issues
Hi all,
i'm trying to integrate cepstral and asterisk, and i have a problem i'd
appreciate any help with (i know it's a bit tangential, but i figure this is
the place with the most knowledge of app_swift and asterisk).
I've installed swift from cepstral.com with alison's voice, and it works
fine, from the command line i can do swift "hello there" -o test.wav and
then
2005 Feb 14
2
FW: SER Asterisk Voicemail
Any more ideas on my below mail? If a user is registered with SER and
leaves a voicemail message with asterisk (by using rewritehostport
etc in ser.cfg), then how is the user supposed to listen to the
message afterwards? Is there any other way other than the MWI method??
Thnaksm
Aisling.
---- Original Message ----
From: ashling.odriscoll@cit.ie
To: asterisk-users@lists.digium.com
Subject: FW:
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 Oct 12
1
question about PSTN pickup
hi all,
you'll have to excuse the ignorance (i'm a software guy, not a telcom
guy..)
Is there any way to know if a channel has been answered by an automatic
system (like voicemail) rather than a human being?
Specifically, I want to use a .call to make a call on a channel and only do
something if a person answers, not a machine of any kind. Is this even
possible, or is an answered
2005 Jul 02
3
call forwarding, most basic case
hello all,
i need some help and after trying the wiki i'm even more confused than i was.
i'm trying to set up call forwarding and running into problems...
i want the most basic call forwarding imaginable.
1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is
2005 Jan 28
0
asterisk call flow diagrams for ser voicemail combo
Hi everybody,
I am trying to make up call flow diagrams for for a setup which
include ser as a sip proxy/registrar and asteriks as a voicemail
server.
Is my sequence correct?:
UA 1 send an invite to SER. SER forwards this invite to UA2. UA2
sends back a sends back a 100 trying and 180 ringing message. SER
forwards these. However UA2 doesnt answer the phone,so what happens
then?...is there a
2006 Mar 10
0
Forward from SER to asterisk can't hang up
Hi All,
I have CentOS 4.2 with ser 0.9.6 and asterisk 1.2.4. Ser is listening on
5060 and asterisk on 5065.
The setup is that people use serweb to create an account and register a
phone. Their calls are routed from ser to asterisk and then inbound on
IAX2.
The server has a public and an internal interface. The real FQDN of the
server is nmibwksip3.nexusmgmt.com and it has cnames of pbx and
2004 Dec 09
0
Ser + Asterisk & DMZ
Hi all
I am in this strange situation: we had ser configured to relay calls to
numbers to asterisk extensions and all used to work nicely, with both ser and
asterisk running on the same machine with public ip (ser on port 5060 and *
on 5061). We had to move temporarily our server to another provider which put
our server on a dmz, so that now we have our server with private ip but
reachable from
2007 Mar 21
1
Metaswitch help needed
I'm attempting to connect to a Metaswitch, inbound only (at this time).
The Metaswitch is the only "connection" (at this time).
All I'm getting so far is a bunch of "OPTION" messages which my Asterisk
box replies to but I don't get inbound calls.
Here's my sip.conf. As you can see I've been trying a bunch of different
options without success :(
2004 Aug 16
0
re: asterisk as VM for SER
(sorry, posted without subject)
hello,
if anyone is using asterisk as a voicemail system for SER I would be
grateful if i could see a working ser.cfg and extensions.conf of such a
setup. I am having some issues with rollover to voicemail when busy, and in
setting up a VM extension for users to retrieve their mail without having
to enter their own extension.
When i get this working i'll write