Displaying 20 results from an estimated 10000 matches similar to: "SER Interaction: Agents and Extensions"
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
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
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
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2008 Feb 14
1
Ser, asterisk and ip2ipgw
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<body bgcolor="#ffffff" text="#000000">
<font size="-1">Hi,<br>
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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
2005 Jan 25
1
SER Prob
Hi all,
Hope somebody can help-I really am stumped as to why this won't work.
I realise that this isnt an Asterisk problem (Please dont bash me on
the list) and I have emailed the SER list but I havent received a
reply and maybe someone on this list can help...Once this problem is
solved I am going to use Asterisk for voicemail etc with SER (I have
it set up)
I currently have SER set up and
2012 Jan 03
4
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Hi,
Please help me understand the following applications and what are its
advantages if we compare between each of them.
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Regards,
Kaushal
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2005 Aug 28
1
SER + ASTERISK voicemail
Hello,
I try set Ua---SER----Asterisk (voicemail/ARA)
|
Ua
ser stable
asterisk cvs head
I read
http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
to forward unavailable or busy sip agents to asterisk
voicemail in failure route.
How may I configure extensions.conf and ser.cfg ?
I have been trying without success!
Regards
Harry
2007 Nov 19
5
Registration problem: UA -> SER -> Asterisk
Hi,
we a have a SER (OpenSER) in front of 2 real-time Asterisk.
SER simply forward SIP messages to 1 of the Asterisks:
UA --> SER --> Asterisk
We have a problem with REGISTERs:
Asterisk answers with 200 OK, but changes the Contact header, inserting
the IP of SER instead of the original IP (the IP of the UA).
It seems that performs a sort of NAT-traversal, but all the elements are
on
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
2005 Aug 29
1
SER NAT any additional requirement
Hello
i am trying to use this exmple with SER-0.9.3
but still NATED Clients are not working any other
requirement
http://www.voip-info.org/tiki-index.php?page=SER+example+NAThelper
-----------------------------------------------------------
# $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters
2005 Aug 10
0
Asterisk and SER and Asterisks Queues
Hi all,
Can someone help with with Asterisk, SER, and Asterisks Queues?
I have three servers:
Server A: Asterisk with TE410 connected to PSTN
Server B: Asterisk connected to Server A via IAX2 trunk
Server C: SER where SIP agents register/connect to
What I wanted to do is configure Server A so that it would route
certain DIDs to specific UA that are registered in Server C. I don't
think
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 Jan 09
1
* as sip b2bua?
Hi everyone,
any chance * could be used as a b2bua without forcing the media stream
through the same box? I would love to do some computing on incoming
calls, do things like setting another callerid and the forward the call
to another sip UA - all without any audio traversing the * box. Any
ideas?
Thanks,
Thilo
2005 Aug 24
0
Re: [Serusers] SER IP PBX for multiple clients
lqbal,
I do plan on having alot of users. Two markets I'm trying to get some
volume users from are: residential consumers and business users.
Residential consumers should get basic line services such as their
own DID, voicemail, caller-id, call-waiting, three-way calling, and
basically, all the standard features you get from companies like
Vonage, etc. This particular market base
2005 Aug 24
0
Re: [Serusers] SER IP PBX for multiple clients
Waldo,
How do you let your customers manage 'their' PBX. I too have a setup
like you. However, I installed a * server for each customer, via
vserver. I'd like to now what kind of software/webbased package you use
for this.
I also have SER installed as a front-end server for the * servers. But,
as I'm still not very into SER, don't know exactly how this fits in.
Should I 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
2004 Dec 28
0
Packet flow in relaying from SER to Asterisk
Hi,
I know the following is mostly the issue of SER and I already posted the
same content to SER User list. Just for more input, I posted it to this
list. Sorry for the cross post for some people.
I've set up SER for UA to UA call.
I'm thinking of setting up SER to relay to Asterisk PBX to use conference
call and voicemail of Asterisk.
I will employ this system for client connection
2005 Feb 22
0
Do ser + asterisk_b2bua work ?
Dear ALL:
I find a program named "asterisk_b2bua" on
http://developer.berlios.de/projects/b2bua/
And I also download them(two components) and try to test it.
But I have not enough knowledge about asterisk. It seems a Software PBX.
Does asterisk_b2bua work? Does anybody ever try it?
I have questions about my scenario.
|======================> UA2
2004 Jun 23
0
Asterisk as a SIP UA and voicemail with SER not working anymore
Hi,
I downloaded the stable branch of asterisk a couple of month ago, and I'm using it as a SIP UA voicemail server with SER, and my setup works fine.
I do have a list of phones defined in voicemail.conf, in the sip.conf file I only have the setup of asterisk as a peer registering to ser. The extensions.conf file contain the extensions that link to the voicemail application. This setup is
2004 Aug 25
1
Voicemail forwarding from SER & extensions.conf
I have SER running with Asterisk, both on seperate servers. If I call
another SIP number from my SIP phone SER looks up the phone number to see if
it's online. If it's not online it forwards the call to Asterisk. How do I
configure the extensions.conf file so that calls being forwarded to Asterisk
destined for VoiceMail do not conflict with normal outbound calls destined
for the PSTN?