Displaying 20 results from an estimated 30000 matches similar to: "Asterisk & SER"
2004 Apr 14
0
Asterisk and SER - choppy sound with G.729
Hi,
We are using Asterisk running on FreeBSD, as IVR / Voicemail for SER. We have redirected certain calls from SER to *. On * there is some 'testing' extension. It's simply playing some demo now ;-)
As long as I use plain G.711 the sound is nice. When I switch to G.729 the sound is choppy, not recognizable. What is going on? Debug shows everything is normal..
I understand that all
2006 Jan 20
5
When/whether to use SER?
I have seen a lot of references to SER.
Currently, I have:
1 PRI to Telco
1 PRI to old PBX
Several SIP phones with the intention of having approx. 200.
I do have people that travel with softphones (currently X-Lite, but will be testing EyeBeam for better codec and echo cancel
capabilities)
Currently the traveling users have to VPN in first which I am sure is adding extra overhead to the calls.
I
2003 Aug 02
1
Asterisk + SER
I was just looking through the SER archives and someone mentioned using
SER with *, his comments about * where very complementary, is there
anyone using SER on this list. I could do with a bit of a HOWTO.
--
Dave Cotton <dcotton@linuxautrement.com>
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 24
0
Question of SER to Asterisk to PSTN
Dear ALL:
My scenario lists below:
Assume: UA1 with sip id "1011"
And dial number to PSTN is "0939749xxx"
There is no modification rule at my CISCO.
(It will not change any dialed number)
UA1 ==> SER ==> UA2
(SIP to SIP)
UA1 ==> SER ==> Asterisk ==> CISCO 5300 ==>
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.
2005 Aug 17
0
Asterisk (multiple) + Ser
I have several Asterisk servers installed and one SER server which will
act as a gateway to PSTN, en redirect server.
I was thinking to implement it the following way:
- Register all the * servers at SER (is this neccessary?) -> this works
via register=>asterisk:password@serbox in sip.conf
- Setup aliases in SER for the telephonenumbers to the appropiate *
server: serctl alias add
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 Jan 30
0
re: help with redirect from SER
hello all,
i have a problem, and i'm tearing my hair out...any assistance is
appreciated. I am trying to redirect from SER to Asterisk, both on the same
machine. In 1.09 I didnt need to set up a peer for SER, just
autocreatepeer=yes, and rewritehostport from SER as below, and asterisk
accepted the requests without a problem. When I updated to 1.23 requests
from SER to asterisk die quietly, no
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 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
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 Mar 16
1
Re: [Serusers] ser+asterisk - security
Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN).
You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified.
----- Original
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
2005 Aug 28
0
SER and Asterisk authentication
Heya,
I'm trying to get SER up and running as a front-end for a couple of Asterisk
boxes for SIP clients. I'd like clients to register with the SER platform.
However, I'd like clients to authenticate with Asterisk when they try to
make outgoing calls via Asterisk. Otherwise it seems that users could bypass
my SER box and register directly with the Asterisk boxes and bypass
2015 Mar 06
0
cant get incoming calls in asterisk
*friends help me *
*cant get incoming calls in asterisk*
*(when i connect **80081 in softphone ---every thing is ok**)*
*<--- SIP read from UDP:200.152.125.221:5060 <http://200.152.125.221:5060>
--->*
*INVITE sip:80081 at 10.47.10.10:5060 <http://sip:80081 at 10.47.10.10:5060>
SIP/2.0*
*Record-Route: <sip:200.152.125.221;lr;ftag=as6872d065>*
*Via: SIP/2.0/UDP
2005 Aug 19
1
Nat + Asterisk + Ser (Far end Nat Traversal)
Hello,
I have several * servers behind a SER server (in a local ip range). The
SER server is also publicy reachable. On the other site, I have SIP
clients that are behind another NAT or in the same NAT range as the *
server. Can someone give me some directions/hints etc. on how to make
this work. I think I should be using MediaProxy with SER. But do the SIP
clients need to register at the SER
2004 Jun 25
1
SER and NAT
I have a really simple question about a fairly complex problem:
I have a Cisco 7960 behind a NAT. I have an Asterisk server behind
a different NAT. I have a SER server (with rtpproxy installed) on a
public IP adress. I've opened ports with static NAT to * and the
Cisco. Without using SER, I can register the phone to *, I can complete
calls, I just can't move audio. Reading the
2004 May 31
1
Asterisk and SER Setup Questions.
Dear All,
I have the following setup.
Quad T1's<->Asterisk (PBX)<->(LAN<->DMZ)<->SER<->(Firewall)<->(Internet)
|
Local US Help Desk (Snom 200')
This setup works well. I can pass calls from over the internet to the
Asterisk PBX via SER using X-Ten Lit.
I have a couple of questions;
1. How do I
2005 Feb 11
0
Asterisk as a UAC forwarded by SER
Hi everybody,
I have a SER Server (Sip Proxy / REGISTRAR) and a Asterisk Server (PSTN and other services). I've got some clients that make calls to each other through my SER Server, that's to say, non external or international calls. I would like my clients to make external and international calls through my server but for that they must authenticate at another server to have a valid