Displaying 20 results from an estimated 40000 matches similar to: "HELP... SER + Asterisk as feature server"
2005 Jan 18
0
DTMF is being MUTED by asterisk to/from SIP channels from SIP or ZAP
I am having a problem trying to do inband DTMF passthru via asterisk.
My setup:
PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2)
Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly)
12/10/04 and 01/17/05 (no difference)
CAC ABII-T100P to/from analog lines to/from asterisk
BTW, I have used a ABI and it works just like the ABII with asterisk.
What I am seeing is:
I make a call from a
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
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 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 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 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
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
2003 Sep 06
3
Ser vs Asterisk?
Could someone give me a 10,000 foot view of what the differences are
between Ser and Asterisk?
I'd like to implement one or the other handle a small number of local
ip phones, tie a couple of asterisk (or ser) machines together across
the Internet, implement a couple of FX gateways (to handle incoming
pstn calls, and for outgoing pstn calls), and use features mostly
common to pbx's. No
2005 Aug 14
4
Multiple Asterisk Installations + SER
I'm trying to implement a shared asterisk server for multiple
(different) companies. Here's what I've done so far:
- I've installed multiple asterisk instances on one server (via
vserver). Each * is for one customer, and has it's own extensions (like
100, 101, 102, etc.) Note that the same extension can exist on other *
instances
- The SIP Clients register themselves with *
-
2006 Nov 23
1
Call Transfers in SER + Asterisk architecture
Hi,
I'm deploying a SER + Asterisk architecture, where SER is used as SIP
registrar, and Asterisk is used for voicemail and PSTN gateway.
This system is already able to make Call Transfers (Blind and Attended)
internally between phones registered in SER, although, I can't make
Call Transfers in some scenarios involving PSTN numbers (which need to
pass through Asterisk).
The problem
2007 Apr 24
1
SER/OpenSER, I Finally Get It.............General Observation
Sorry if this hit the list twice, sent out yesterday, but didn't see it show up.
Hi All,
Can Asterisk be used as a SIP proxy, blah, blah, blah???
I've glanced over questions like this through the years, with a good idea on
what a SIP proxy is and what Asterisk is and IS NOT. I never really took
the time to lab-up SER and test drive it to see what advantages might be
gained from using
2005 Feb 25
0
SER vs. Asterisk - call in progress to PSTN
We're having a problem with Asterisk when we try to pass a call off to a
Lucent PSTN using SIP. This behavior does not exist with SER:
With Asterisk
An ISDN call is started, at the T1 level we receive ?call proceeding?
and immediately we receive a ?Call in Progress? just like the far end
party has answered.
With SER
An ISDN call is started, at the T1 level we receive ?call proceeding?
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
2005 Aug 09
1
voip solution with SER, ASTERSIK and CCM
We are planning to install a voip system based on asterisk for 2000-3000 retail locations and up to 6000-8000 sip accounts/users.
Instead of setting up a new, centralized PSTN gateway, we are intend to use a CISCO gateway/router of an existing CISCO voip solution in the headquarter and we must able to call all CISCO based voip phones in the headquarter running together with a CCM.
SIP-Phones
2004 Jun 07
0
FW: Problem with Asterisk PRI forwarding to SER
_____
From: Habiyakare Aimable [mailto:aimable@terracom.rw]
Sent: Monday, June 07, 2004 11:49 AM
To: 'asterisk-users@list.digium.com'; 'gt'; 'support@digium.com'
Subject: Problem with Asterisk PRI forwarding to SER
Hi all,
I have a problem. We have a phone system setup like this:
SIP phones------------>SER--------------->Asterisk---------------->PSTN(PRI
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
2004 May 25
1
Using Ser and Asterisk together
Hi all,
I would like to know if it is possible to use asterisk
and ser together in a single computer system using ser
as a sip proxy and forwarding any voice call request
to asterisk for calling into the pstn gateway. (or any
other alternative that is possible is also welcomed
for suggestions). If it is possible can someone kindly
show me the necessary configuration files or refer me
to any page
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
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?