Displaying 20 results from an estimated 900 matches similar to: "Asterisk with Euro ISDN, etc"
2006 Jun 26
2
n-way has_mant :through
I''m trying to setup some mildly complex associations for a project we''re
working on and can''t seem to find much documentation on n-way has_many
:through associations.
I have the following models: Person, PhysicalAddress, EmailAddress,
PhoneNumber.
Each person can have multiple PhysicalAddresses, EmailAddresses, and
PhoneNumbers, and multiple people can share the same
2005 Aug 08
1
Same action to multiple numbers
Hey!
I'm going to implement same actions (answer,wait,dial,playback,queue,hangup) for
3 phonenumbers when call is coming. Do I have to write three separate lines for
same actions for each phonenumber or is there any way to write one command
lines for all three lines? Any examples?
Thank you for your answers!
----------------------------------------------------------------
This mail sent
2005 Jan 18
3
Prefered server hardware
What server hardware would you recommend for an Asterisk system which are really critical?
The additional hardware will probably be two digium TE110P cards, and an Adit 600 platform.
If it's possible to run on -48VDC, It would be great!
Are there any experiences with any HP or FujitsuSiemens systems? Or other "complete server" systems?
Thanks!
BR
Daniel Nystr?m
2006 Feb 13
0
Asterisk register ip phone
Hi all
I have a problem to register a cisco 7960 to an asterisk 1.2.2
I defined in sip.conf the next :
["phonenumber"]
type=friend
username="username"
secret="password"
host=dynamic
context=work
I am trying to catch the register requests with
sip debug
with no success (empty screen).
I can only catch the register messages with ngrep on
2006 Feb 19
1
Cisco 7960 Register Problem
Hi all
I have a problem to register a cisco 7960 to an asterisk 1.2.2
I defined in sip.conf the next :
["phonenumber"]
type=friend
username="username"
secret="password"
host=dynamic
context=work
I am trying to catch the register requests with
sip debug
with no success (empty screen).
I can only catch the register messages with ngrep on host it's comming
2005 Feb 15
2
E1 and/or Euro-ISDN specifications?
Where can I get E1 and/or Euro-ISDN specifications/data sheets?
Are there specs for other E./G./Q./etc. protocols as well?
Thanks!
2005 Aug 27
1
SIP Registration failure
Hi list,
I'm in central-europe and signed yesterday a broadvoice account. My
Asterisk box is CVS 2005-08-25.
Problem I face is:
"Failed to authenticate on REGISTER to 'phonenumber@sip.broadvoice.com'
(Tries 2)" then
"Registration for 'phonenumber@sip.broadvoice.com' timed out" and finaly
"Giving up forever to register
2005 Jan 20
2
Some more hardware and E1 questions
Hi again folks! ;)
As before, I will transform one E1 30 Channel PRI into 30 FXS channels using Adit 600.
Now I'm into choosing server platform. And the two opponents are:
* Dell PowerEdge 750 w/ SCSI RAID (or even SATA RAID1)
* FujistuSiemens PRIMERGY RX100 S2 (SATA RAID1)
As I've seen people having problem with HP server, I havn't looked at it at all.
What experience do you
2006 May 09
1
Asterisk settings Net2Phone
Hi,
I?m looking for settings to configure net2phone carrier in my asterisk. I
found this configurations, but it?s not work. I don?t known if this
configuration is for voice line or voice access account.
Anybody can help me, with other configuration?
Thanks.
----
*sip.conf*
[general]
useragent = X-Lite release 1103m
register => PHONENUMBER:PASSWORD@sip.net2phone.com
[net2phone]
type = peer
2006 Nov 13
1
Sending '#' with Dial
Hi!
I have a working asterisk-setup with four sip-clients. Everything works
great but when the users call someone the phonenumber shows up on the
receiving ends callerid-display.
To correct this my provider told me to send #31# before the phonenumber,
tried this with: Dial(SIP/#31#${EXTEN}@provider) but my asterisk tells me
that it isn't a valid extension.
The INVITE looks fine,
2004 Dec 12
1
I'm stumped
I am trying to use the simple CID name management script on the wiki.
http://www.voip-info.org/wiki-Asterisk+tips+managing+CID+names I can not
see what is wrong. The values never get entered in the database. Here are
the files: I have asterisk running as the user asterisk also.
---cid-store.php----
<HTML>
<HEAD>
<TITLE>Storing Asterisk CID data</TITLE>
</HEAD>
2003 Nov 03
1
one way sound with x-lite (sip) -3rd attempt !
Hi List,
Additional with the latest tries from the below
I get a nice random seg fault when I hangup on PSTN.
(With obviously no sound on x-lite, still!)
asterisk -vvvvgc
results after hanging up the pstn line in:
-- Executing Hangup("SIP/1087997-d79f", "") in new stack
== Spawn extension (sip-phone-out, h, 2) exited non-zero on
'SIP/phonenumber-d79f'
Segmentation
2005 Jan 14
1
Grouping lines pending on Called ID
I will be using one E1 to the telco, and there will be 4 static phone numbers, and one number serie with 1000 numbers ranging from e.g. 555-1000 to 555-1999.
There will be 30 FSX lines on the other side of Asterisk.
Is it possible to "group" those 30 FSX lines pending on which number was dialed?
Let's say line 1-4 are for the static numbers, and 5-30 for the other 1000 numbers.
Are
2005 Jan 17
2
CAS voice signalling?
According to CarrierAccess, the Adit 600 uses CAS for voice signalling. What is this?
This should not be a problem for Asterisk?
Does the Asterisk server need to reencode CAS into aLaw when going to Euro ISDN?
BR
Daniel Nystr?m
2010 Jul 26
0
Adit 600 over MGCP.
Hi,
Anybody out there running Adit600s?
I have in my care an Adit600 channel bank connected to an old (version
1.0.6) Asterisk instance with MGCP. When trying a more recent Asterisk
(1.4.21.2~dfsg-3+lenny1, Stock current Debian) calls fail.
I have attempted to add the "slowsequence = yes" line to mgcp.conf. (It
seemed to be the only likely candidate in the example files I found
2006 Jan 17
2
change error messages for Validation helpers?
Is it possible to change error messages for Validation helpers? I am
writing an app against a existing database (so no control over column
names), but when there is validation error (e.g. with
validate_presence_of) I would like to customize the field name. For
example for telephone whose field name is PhoneNumber I would like to
chnage it to "Telephone Number cannot be empty" rather
2014 Aug 11
1
401 Unathorized
I have an asterisk 1.8.x box that intermittently returns a 401. Calls come
through the same peer all the time, from the same carrier. However
intermittently the asterisk box returns a 401.
Below is the output of a failed call (1st) and a successful call (2nd). I
can't see any difference until we get to these lines.
Bad call:
--- (17 headers 14 lines) ---
Sending to carrierIP:5060 (no NAT)
2006 Oct 18
0
[OT] Nokia E60/61/70 and SIP
Martin Joseph wrote:
>
>
> For all of us using these devices, I have some good news. There is a
> self installable firmware update available from Nokia here (requires
> windows box to install):
>
> http://www.nokia.co.uk/nokia/0,1522,,00.html?orig=/softwareupdate
>
> This seems to radically improve the behavior of the SIP client on my
> E60. It seems to have
2008 Jan 18
1
Automatic call-out problem
Hello!
My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on
Fedora Core 4. I am making automatic call-out campaign with this setup
on 4 PRI. The scripts for this:
====================================================================
caller php script write this to outgoung folder:
fwrite($outfile,"Channel: Zap/g1/$phonenumber\n");
fwrite($outfile,"MaxRetries:
2010 Sep 02
0
asterisk 1.6.2.11 freezes the server
Hi,
I have a problem that the machine running asterisk 1.6.2.11 freezes
unexpectly time to time. Sometimes it runs for 4 weeks without any
problem, sometimes after a free it freezes again in 24 hours. But
usually it runs normally for 1 month or so before it freezes again.
I could not find any additional info in any file located in /var/log/
including asterisk's messages file. Verbosity