Displaying 20 results from an estimated 10000 matches similar to: "Re-routing of existing calls"
2003 Nov 25
4
Options for 3rd party call control
I am working on a project on 3rd party call control for a call center, for
which I think Asterisk may be useful. What I would like to do is:
- Have a call come in to Asterisk.
- Asterisk asks another machine, over a slow IP link, such as a modem, how it
should route the call. Asterisk passes the called and calling numbers.
- This other machine looks up the destination, based on called and
2006 Mar 06
4
Asterisk download file locations
This is a request to the website manager for asterisk.org.
The build scripts for our ITSP product include the URLs to download the
Asterisk files, such as:
wget "http://ftp.digium.com/pub/asterisk/asterisk-1.2.5.tar.gz"
However, if a new version is released, asterisk-1.2.5.tar.gz is moved to
the "old" directory. This breaks our scripts until we can update them
and send
2005 Feb 17
4
Call termination database
I've been considering doing a web based database system, where you can
post your termination offerings or wanted, then search by location,
price, minimum volumes, etc.
I'd probably make it free, supported by advertising my consulting
company, or Google Adwords, or something like that.
I've got the design written down, all ready to start coding. I could
probably have a prototype
2005 Jun 16
6
Case studies for Asterisk Voicemail
I'm planning an Asterisk Voicemail system of around 3000 users spread
across several sites, each site connected by a fast network to a central
site. We're considering 2 models:
- Central Voicemail with VoIP calls from remote sites (easier to
administer the system(s)).
- Voicemail server at each site with shared database and NFS server at
the central site (easier to connect to the
2005 May 31
3
Opinions of Sphinx?
I'm planning a system of 120 SIP or PRI channels using speech
recognition (fixed grammar of 500 words) menus.
I could use a Cisco router and VoiceXML, but would prefer not to on cost
grounds.
Has anyone tried Asterisk and Sphinx (bonus points if in a production
environment)? If so, what's your opinion on quality of recognition,
stability, resource usage, etc?
Anyone have any
2006 Apr 06
1
Integrics ITSP 1.6 released
Integrics is pleased to announce the release of ITSP version 1.6. This
version has the following new features:
- Comes in 2 editions:
* Carrier edition, for 250 to tens of thousands of users on hosted
systems. Integrics sells this edition directly and through partners.
* Office edition, for 10 to 250 users. This edition is sold only
through our partners, for them to sell as PBX systems at
2006 Jan 03
5
Asterisk on Dell blade servers
We've been asked to quote for a large cluster running Asterisk and our
ITSP in a box product. The system will be SIP throughout, with mixed
codecs.
We're considering using Dell blade servers, 1855 or similar, on the
grounds that we normally use Dell machines and they work well, but we
need higher rack density.
Has anyone used these? Any feedback on whether they're
2005 May 15
1
Scalability of chan_oh323
I have a customer who wants to do large volumes of H.323 to H.323
hairpinning. We haven't tested this scenario for large volumes before;
maybe someone on asterisk-users has.
If they buy a top of the line PC, how many concurrent calls are we
likely to get? Routing logic will be simple, the machine won't be doing
anything else, and let's assume no transcoding for now.
We're not
2006 May 09
1
Many music on hold files
A feature we're often asked for in our ITSP product is to allow
customers to upload their own music on hold, or to have it recorded for
them by a recording studio with the latest news, weather, etc,
punctuated by "Welcome to <customer>, please hold".
Since there may be thousands or tens of thousands of customers, and
perhaps 10% of customers may want this feature with a
2010 Jul 03
1
VoIP Users Conference Recordings
Hi,
Alistair Cunningham of Integrics was our guest yesterday. We talked
about Integrics new product Geons, a suite of software for building
large-scale distributed enterprise applications. The recorded session
is now available here:
http://www.voipusersconference.org/2010/geons/
The extremely rare John Todd was sighted (and heard) at this event.
If you are developing a product or service
2006 Apr 28
1
Integrics release Enswitch 2.0
Integrics is pleased to announce version 2.0 of Enswitch, the most
integrated platform available for offering commercial telephony services
such as ITSP, hosted PBX, calling cards, call shops, number translation
services, and much more.
Enswitch was formerly known as ITSP in a box, and Enswitch 2.0 is
effectively the same product as ITSP 1.7. The product has been rebranded
as, although it
2006 Jan 04
2
Using *RT for HA purposes was: RealtimeMultipleAsterisk boxes, iaxusers
I think I have 4 options.
1, Modify chan_sip.c to update a new field in sipusers realtime table
with the status of the sip peer/user. Then use agi to dial sip calls.
Check the status field if OK then dial the fullcontact from the sip
table. If not goto voicemail or where ever else I want the call to go..
The UA would only register to one server, so only one server *should* be
writing to the
2005 Feb 19
3
Terminating IAX calls in E1 PRI interface - what do I need to be able to send arbitrary caller id to called party ?
Hi,
I'd like to terminate IAX call on PRI interface. What conditions should be
met to be able to send arbitrary caller numbers to called party, so he can
call back to supplied ISDN number (that is different for every IAX user) -
not through PRI interface, but plain ISDN call !!
Thanks in advance,
regards,
Rob.
2005 Mar 09
4
Which box?
I'm sure this is a stupid question, but I'm not finding an answer
anywhere. Do I need a dedicated box to run asterisk, or can I put in my
server (running Fedora) and leverage some of the free cpu cycles and
disk space? Thanks,
Dunc
2020 Oct 30
3
Multiple IP addresses and using same IP for outbound calls as inbound
Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass
it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio
On Thu, Oct 29, 2020 at 20:44 David Cunningham <dcunningham at voisonics.com>
wrote:
> Hello,
>
> Does anyone know a way with chan_sip to tell Asterisk to use a specific IP
> address for its end of the communication for a specific
2020 Oct 23
2
Multiple IP addresses and using same IP for outbound calls as inbound
OK, thank you George.
On Sat, 24 Oct 2020 at 03:16, George Joseph <gjoseph at digium.com> wrote:
>
>
> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <
> dcunningham at voisonics.com> wrote:
>
>> Hi George,
>>
>> Thank you for the response. I'm a little unclear on what you mean by a
>> transport. We're using chan_sip, not pjsip.
2005 Feb 23
5
Difference between E1 and PRI
Hi all,
I have seen the term E1 and PRI used interchangably when referring to
a voice service with 30B channels and 1 D channel. Are they just
different terms for the same thing or is there some technical
difference. Even Newton's telco dictonary seemed a bit fuzzy on this
topic. I have seen it said the PRi is a protocol that runs on top of
E1. Is this true?
2005 Feb 11
3
Newbie: ISDN E1 the same in all countries?
Hi.
I'm looking at ordering a 30-channel ISDN connection from telia (a swedish
operator) and then using a Wildcard TE110P card with that and asterisk to do
IVR.
Can I be certain that the TE110P card will work with that ISDN connection? A
30 channel ISDN certainly sounds like an E1 connection, but I couldn't get
any clear answers from the operator if
it is.
Has anyone used the TE110P
2020 Oct 22
2
Multiple IP addresses and using same IP for outbound calls as inbound
Hi George,
Thank you for the response. I'm a little unclear on what you mean by a
transport. We're using chan_sip, not pjsip.
Do you mean a device in sip.conf, using bindaddr to set the address to bind
for that device? We've only used bindaddr in the [general] section before,
but if it will work in a device that could be the answer.
On Fri, 23 Oct 2020 at 00:13, George Joseph
2014 Sep 06
0
bootable dvd
----- Original Message -----
From: "Bill Cunningham" <billcun at suddenlink.net>
To: "Gene Cumm" <gene.cumm at gmail.com>
Cc: "For discussion of Syslinux and tftp-hpa" <syslinux at zytor.com>
Sent: Saturday, September 06, 2014 9:31 AM
Subject: Re: [syslinux] bootable dvd
>
> ----- Original Message -----
> From: "Gene Cumm"