What kind of specs do I need for a asterisk box that will have a pri for pstn and about 65 sip phones I was thinking a Xeon 3.05
simprix wrote:>What kind of specs do I need for a asterisk box that will have a pri for >pstn and about 65 sip phones > >I was thinking a Xeon 3.05 > > >What length is a piece of string when you cut it? I was thinking 2.374 m Sorry about the sarcastic answer but if you look through the mailing list archives you will see that this question gets asked all the time with and the person asking never gives enough information.. Things like are you using IP phones or analog phones? if IP phones what codecs do you plan to use? how many concurrent calls to you expect? etc.. Its difficult to give someone an estimation of load with no information.. Maybe this will help you.. http://www.voip-info.org/wiki-Asterisk+hardware+recommendations Later..
Dear List, I've compiled asterisk (both 0.9.0 and the CVS-04/19/04 source trees). I'm using the oh323 channel driver version 0.5.10, OpenH323 v1.12.2, PWlib v1.5.2 When run on a RedHat 9 system, I am constantly getting seg faults. This happens even when I tried removing the oh323 channel driver, so it appears to be something with asterisk. I get crashes either when attempting to start asterisk or when asterisk receives an incoming h323 call. When run on a RedHat 7.3 system (exact same source code) both asterisk and the oh323 channel driver appear to be stable. Does anyone have any advice? I assume this has something to do with incompatible libraries, but have no idea where to start. TIA Chris -- Chris Wik Systems Admin ANU Internet Services http://www.anu.net/
After a recent upgrade to asterisk HEAD, my asterisk startup scripts don't properly start asterisk. They have since May, which is the last time I upgraded. I am on Slackware 9.1, running kernel 2.4.26. After reboot, lsmod shows wct1xxp, then zaptel, which would indicate it now loads out of order? Shouldn't zaptel be loaded first? Maybe my original install is a little hacked. Where do you load all your modules and asterisk from on startup of your server? I have a T100P and a TDM400P installed.
Will Stowe Systems Administrator Promisant (USA) Inc. email:wstowe@promisant.com Office: (770) 913-3723 Mobile: (404) 993-0526
On Feb 1, 2005, at 18:13, Adams, Gavin-ML wrote: <snip> How many more empty test messages are we going to see from you..? jens
I?m a telecommunication engineering student. I?m working on my degree thesis, it?s about Astrerisk . My goal is to estimate the performance of a hybrid platform for the Volp. I?m looking for documentation about: ? Architecture ? Tools for the performances? analysis (to analyse performances) ? Informations about the scheduler ? Informations about the transcoding, to understand how the Volp Protocol (Sip,H.323,IAX) interact If you can help me, please, send me some informations to understand how to start to analyse performances; what I found on internet is not enough!!! Thank you so mutch
Do a search on Empirix. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of gale81@virgilio.it Sent: Saturday, March 19, 2005 1:39 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk I?m a telecommunication engineering student. I?m working on my degree thesis, it?s about Astrerisk . My goal is to estimate the performance of a hybrid platform for the Volp. I?m looking for documentation about: ? Architecture ? Tools for the performances? analysis (to analyse performances) ? Informations about the scheduler ? Informations about the transcoding, to understand how the Volp Protocol (Sip,H.323,IAX) interact If you can help me, please, send me some informations to understand how to start to analyse performances; what I found on internet is not enough!!! Thank you so mutch _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
hi, I have 1.Registered Asterisk with Proxy as bob@proxy1.mydomain.com/5555. 2.Sucessfully registered and got 200 OK. 3.It shows contact as sip:5555@asterisk IP 4.But if i send an INVITE request to Proxy it says 404 not found since that extension does not exist at the Proxy. This is an incoming call Scenario. how the outbound call scenario taking place?that is from Asterisk to SIP entity.What is the configuration? regards, sangeetha Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin@wipro.com immediately and destroy all copies of this message and any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050323/c4efd411/attachment.htm
I love this message, just the perfect one to get some people on the list pi**ed off. The subject is *very* descriptive. The disclaimer just a beauty. On Thu, 24 Mar 2005 10:40:58 +0530, innovation.interops@wipro.com <innovation.interops@wipro.com> wrote:> hi, > > I have > > 1.Registered Asterisk with Proxy as bob@proxy1.mydomain.com/5555. > > 2.Sucessfully registered and got 200 OK. > > 3.It shows contact as sip:5555@asterisk IP > > 4.But if i send an INVITE request to Proxy it says 404 not found since that > extension does not exist at the Proxy. > > > This is an incoming call Scenario. how the outbound call scenario taking > place?that is from Asterisk to SIP entity.What is the configuration? > > > regards, > sangeetha > > > > > > > > > Confidentiality Notice > > The information contained in this electronic message and any attachments to > this message are intended > for the exclusive use of the addressee(s) and may contain confidential or > privileged information. If > you are not the intended recipient, please notify the sender at Wipro or > Mailadmin@wipro.com immediately > and destroy all copies of this message and any attachments. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
I have been trying for days to get an outbound connection to broadvoice
with no luck ..details below ... I have scoured all postings and seem to
get similar responses but none of these seem to help... any help is
appreciated ..
my asterisk@home box is sitting as 192.168.1.106 behind a linksys router
that feeds to comcast as the provider.
trying to make outbound calls from a analog phone extension on a digium
baord to broadvoice ..
system works fine analog phone to analog trunk , but cant get calls out
from analog phone or softphone to broadvoice .
asterisk log throws
-- Executing Dial("Zap/1-1",
"SIP/1770??????@sip.broadvoice.com|30") in new
stack
-- Called 1770??????@sip.broadvoice.com
-- Got SIP response 604 "Does not exist anywhere" back from
147.135.0.128
== No one is available to answer at this time
-- Executing Congestion("Zap/1-1", "") in new stack
== Spawn extension (from-internal, 17705229625, 2) exited non-zero on
'Zap/1-1
sip .conf is
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 192.168.1.106 ; Address to bind to (all addresses on machine)
disallow=all
allow=gsm
allow=ulaw
allow=alaw
;context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
context = from-broadvoice
externip=69.??.??.??
localnet=192.168.1.0/255.255.255.0
===================================
sip_additional.conf shows
register=561???????:91?????????:@sip.broadvoice.com/201
*** i have tried various permutations of this
[bv]
username=5618282155
user=phone
type=peer
secret=myPassword
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=561??????
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-broadvoice
canreinvite=no
authname=561??????
[sip.broadvoice.com]
username=561????????
user=561????????
type=user
secret=91???????
nat=yes
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-broadvoice
canreinvite=no
===========================also , per postings on the boards ..i pasted this to
extensions.conf
..seems that amp had not created an entry for this
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()
=========================outbound routing ...
i have prefix 1 directing to the BV trunk
all (other than general section in sip.conf and the extensions.conf) were
setup using amp ..seems amp does not place the entries in extension.conf ...
==========================trunks in amp is a follows
sip trunk...
outbound caller is is "broadvoice"
max channels is blank
no dial rules
no dial prefix
outgoing settings
trunk name is "bv"
peer details are
authname=561???????
canreinvite=no
context=from-broadvoice
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=561???????
host=sip.broadvoice.com
insecure=very
nat=yes
secret=91??????
type=peer
user=phone
username=561???????
incoming settings
user context "sip.broadvoice.com"
user details
canreinvite=no
context=from-broadvoice
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
insecure=very
nat=yes
secret=91?????
type=user
user=561?????
username=561????
register string ...
561??????:91?????????:@sip.broadvoice.com/201
============================================fyi ...
this is an asterisk@home setup ....
my bv number is shown as 561??????
my bv fancy password is shown as 91??????
i have a outbound rule that says numbers with prefix 1 ..go to sip/bv trunk
any help is MUCH appreciated
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