Displaying 20 results from an estimated 8000 matches similar to: "estimation on phone network capacity"
2008 Mar 24
3
Unable to obtain dialed number through ZAP
Hi all,
This is not a repeated post as I am just adding more information for my
previous post.
Asterisk version 1.4.18
TDM card: Digium TDM411B
Zaptel version 1.4.9.2
Line: PSTN line
I tried to obtain the dialed number using $DNID and $CDR(DST) . All of
these variable returns 's'
I also tried exten => _3345335,n,Noop(this is ok) where 3345335 is my
number but it does not go there.
2008 Mar 23
3
Unable to capture CallerID through Zap
Hi all,
I am using Digium PCI board to receive PSTN call through regular phone
line. It is no problem for me to receive calls, but I am not able to obtain
the Caller ID if the calls are from the phone line.
exten => s,1,Answer()
exten => s, n, Verbose(1|incoming number is ${CHANNEL} calling to ${EXTEN}
routing to ${phonenum} )
exten => s,n, Verbose(1|callid is ${CALLID(num)})
exten
2008 May 21
1
T38 fax solution with Asterisk possible?
Hi,
I am looking for a very low cost way of receiving and sending T38 fax
reliably. Is there any possible solution using Asterisk as the PSTN SIP
gateay and Digium E1/T1 card? Is there other open source package that can
help to accomplish this purpose?
Regards,
Mark
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2008 Nov 19
4
question about connecting with Mobile Base Station
Hi,
Is it possible to connect Asterisk with a mobile base station to handle call
switching? What kind of protocol will I need to use to convert to sip?
Any pointer or info will be greatly appreciated.
Best Regards,
Mark
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2008 May 14
3
Question about SS7
Hi,
I have read about SS7 recently and learnt that it is a signalling protocol
used in PSTN for call management, setup, etc. The thing that I don't
understand is how SS7 plays a role in VOIP. When I make calls between
landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it
because the SS7 signalling is already done by Asterisk already? From the
prespective of
2011 Nov 03
15
DID from Direct from Telco
Hello Everyone,
Unlike going through DIDx, DIDLogic etc.., we have an option of
getting DIDs directly
from local telco Bell Canada. Currently our SIP Trunk provider
assigned a DID to us,
and as you know, they just redirect requests it to our PBX.
However, when dealing directly with a telco, what equipment will we
need? Basically
giving us the same capability as a DID provider. If someone can
2009 Jul 23
2
Analog FXO or IAX DIDS for new facility?
I am a Linux sysadmin who has been tasked with developing the phone
system for our nonprofit's new US headquarters building. We cannot
bring our legacy phone system with us, so I am building this completely
from scratch. I have already read "Asterisk: The Future of Telephony"
and done a fair amount of googling. I am completely sold on Asterisk,
and the new building's
2008 Mar 18
3
capacity
Hi,
I am planning to deploy an Asterisk system to supply 4-6,000 students with
voicemail capabilities. The system will be set up with non-DIDs, route
incoming calls to voicemail, then send an email notification. Anyone with
some ideas on how I should go about spec'ing the server this use?
- Eve Ellen
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2008 May 30
1
SPA 3102 unable to detect hangup
Hi,
I have a Linksys SPA 3102 using as ATA. The call routing is : Phone -> PSTN
-> SPA 3102 -> SIP Proxy -> Asterisk
The problem I am having is that when the phone hangs up, SPA 3102 can't
detect it and relay the CANCEL message.
Is this problem with my SPA 3102 config or it just works like that by
default?
Thanks in advance for your help.
Regards,
Mark
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2010 Apr 14
3
Converting GSM calls to SIP
I have asked a GSM operator in my country if he can route a number or a
short code to my asterisk server via SIP (since they dont give DIDs in my
country) the operator said they do not support SIP, they have no way of
converting GSM calls to SIP to then send them to me. I would like to know
what is needed from the operator side to do this, what kind of material is
needed, or what can be done from
2015 Nov 18
4
Linux ate my RAM...
Hello everyone,
Excuse the title. I'm trying to do something very specific that goes
against some common assumptions.
I am aware of how Linux uses available memory to cache. This, in
almost all cases, is desirable. I've spent years explaining to users
how to properly read the free output.
I'm now trying to increase VM density on host systems (by host, I mean
the physical system, not
2008 Nov 18
1
sound quality between two back-to-back asterisk
Hi,
I have two asterisks that are connected to each other via a back-to-back E1
link using a pair of sangoma cards.
With the following scenario: SIP-PHONE <-> Asterisk <-> E1 <-> Asterisk <->
SIP-PHONE, the sound quality degrades significantly. I can't understand
why as the amound of packet lost should be very minimum.
Does anyone know why? Does it have anything
2008 Mar 24
3
How to capture destination number when receive call through ZAP
Hi all,
I am using Digium PCI board to receive PSTN call through regular phone
line. It is no problem for me to receive calls, but I am not able to
capture the destination number through the ZAP channel
exten => s, n, Verbose(1|destination to ${EXTEN} )
${EXTEN} returns 's' instead of the actual destination number. Since I have
multiple phone numbers, I want to be able to route
2008 Feb 27
3
About faxes recived through a PRI and passed to a fax machine connected to a FXS port
Hi, all
I want to configure a few FXS ports in an Antribank-16 to be able to
receive faxes sent throught a PRI:
E1 ==>Zap * ==>FXS * ==>Fax machine
My asterisk box has a Digium TE120P (for the PRI).
Versions are *=> 1.4.17 | Zaptel=>1.4.8 | libpri=>1.4.5
The Astribank is not configured yet, because I am a little bit
confused about how to do it.
Let's say I configure
2009 Jan 13
4
What are the various models of DID providers
Hi,
Inspired by a recent rant about one particular provider, I am getting
very curious about something I've never mastered. I'd like someone to
explain this here or at least post a link or two that can educate me
and probably countless others who have no knowledge in this area. I'm
sure there are several of you reading this that know all about the
subject.
What are the various
2004 Sep 14
4
Memory oversubscription
Hi. First of all, thanks for Xen. It''s terrific!
I''m interested in doing memory oversubscription and am wondering if Xen
can do this (now or in the future).
For example, on a machine with 100MB available physical memory, can I:
1. Create a domain with a 90MB allocation
2. Inflate a balloon in that domain and return say 40MB back to Xen
I know there are mechanisms for doing
2008 Mar 26
2
customizing faxrcvd in PHP
Dear all,
I am working on customizing hylafax's faxrcvd script into PHP. Does anyone
has any sample or guideline that can share with me to give me a quick start?
Two questions I have are: 1. How to simulate the receival of fax without
actually sending one? 2. Where can I find the log that is "echo" from
faxrcvd? 3. How to I config Hylafax so that it uses my PHP script instead
2013 Jun 22
2
SIP Trunking Mantra (Origination)
Hello Everyone,
We are currently having talks with various service providers, and
trying to determine what the best way is to interconnect in order to
have access to the PSTN network. As you know there are two ways of
doing this:
Traditional PRI: Have trunks grouped into a transport layer such as
OC3/12. With DIDs attached to the group. As you many know, this
approach would also require a POP
2006 Apr 26
1
Intergrate Asterisk IP PBX with Legacy PBX, continuing existing funtionality of legacy pbx
Hi All,
I would like to explain the layout that i am trying to achive. I am so
helpless on this regard.
So here is the story ........
" This is with regard to the setup which you can find at the
"Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am
attaching the picture for your information.
Now I am taking a challenging step to of integrate IP PBX with our
2007 Oct 12
5
Affordable SIP Trunk for Home PBX ?
So I have my asterisk box up and working internally at home and all is
good so far. The next thing I wanted to do was make and recieve calls
to regular land lines now.
I don't have a POTS line and was looking for probably a SIP trunk.
I have seen mentions of Skype integration with Asterisk, but does that
include say Skype IN and Skype OUT ? Or is that integration component
really just for