didn't know
what caused the 2100 until you said something.On Wed, 2005-03-09 at
09:47, Matt Schulte wrote:
Disabled echo canceller because of tone (tx) on channel 10
I understand that the PSTN companies use their own echo canceller's,
send a tone across 2100hz, the problem we're having
Having problems getting realtime working, I'm trying to use odbc for all
of this. I've got Fedora 3 and have been fighting with odbc for a day
now. I think I got it working correctly, however I can't seem to get the
realtime portion working. In asterisk 'odbc show' shows it connected, I
see it on
things I have found that doesn't work is a) the mailbox
entry for a SIP user doesn't actually light up the MWI (Message Waiting
Indicator); and
b) voicemail passwords cannot begin with a '0' (zero) because its a
numeric field.
Matt Schulte ([EMAIL PROTECTED]) wrote:
Having problems getting realtime
Per Mike's issue here, we're noticing this problem with older versions
of Asterisk (it would seem?), and especially distrib [EMAIL PROTECTED] As
he stated we're seeing 'No Authority Found' coming from the clients, in
[EMAIL PROTECTED] we get see the No Authority found on the server, and the
allow=all
-Original Message-
From: Matt Schulte
Sent: Friday, March 18, 2005 7:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Netlogic inbound DID issue
Per Mike's issue here, we're noticing this problem with older versions
of Asterisk
Has anyone ever heard of this so called Dynamic Impedance matching on
the ADIT 600? I called their support and they've never heard of it. We
are of course having echo problems are on the far end due to
digital/analog conversion on the local end using a channel bank. We have
purchased an ADIT 600
All, I get this whenever trying to dial to a peer when the peer
registered to another server. I'm basically trying to use realtime to
check for the peer and dial it.
Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial(SIP/brak-f69f,
IAX2/brak-test/107) in new stack
Mar 24 09:16:47 DEBUG[4527]:
Flatfile meaning iax.conf? Yes..
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 24, 2005 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime mysql problem?
Matt Schulte wrote:
Mar 24 09:16
|
++---+-+--+---+--+--
---+---+-+---+--+--+
+-+--+-+-+-+
+-+-+++---+-
---+---+--+++---
++
Matt
-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 23, 2005 8:31 AM
On March 23, 2005 08:25 am, Matt Schulte wrote:
Has anyone ever heard of this so called Dynamic Impedance matching on
the ADIT 600? I called their support and they've never heard of it. We
That's odd
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Monday, March 28, 2005 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime mysql problem?
Matt Schulte wrote
I thought the TDM was broke on 1750's...?? I could never get passed
that NMI issue.
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 29, 2005 9:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dell 1750
?
Matt Schulte wrote:
Ok, that was straight from the wiki. Still does not work, I tried it
from the iax.conf, etc files and it works just fine. I even tried
terminating/placing calls on the same server with realtime and it
works fine. Is realtime broken? Is there anything else I can test
Title: Message
I am
having a similar problem, at least trying to access the dynamic user on a second
asterisk machine that pulls from mysql. Are you getting anything in your debug
log? I'm using the same layout as the sample sip users table from the wiki, the
only difference being I added
-Users] Realtime mysql problem?
Matt Schulte wrote:
How do you toggle the realtime cache?
Check in the configs/iax.conf.sample file of a recent CVS download
and it should be in there.
If there were too many fields
in the table, could you foresee this being a problem?
No, because I
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime mysql problem?
What is your problem with IAX in realtime? I have it working (finally).
Wojtek
- Original Message -
From: Matt Schulte [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
do you have any clue when realtime will get added to stable?
It won't.
why not?
Not to mention since
realtime doesn't support qualify= and NAT mode must be manually set,
Have you been using RTC? (RealTime Cache) It fixes the NAT/MWI
problem.
I haven't tried this yet because of
Now, this has been answered many, many, many times...in fact..I
believe Olle answered this in his Welcome to Asterisk post he sent out
over the weekend.
AAHH my bad, I should have asked *when* it will go stable.. ;-)
___
Asterisk-Users mailing
,
this has seemed to address the issue :-)
Matt
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Monday, April 04, 2005 10:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime mysql problem?
Matt Schulte wrote
PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 05, 2005 1:26 PM
To: Matt Schulte
Subject: RE: [Asterisk-Users] Realtime mysql problem?
Quoting Matt Schulte [EMAIL PROTECTED]:
Ok,
rtcachefriends=yes seemed to have fixed my problem(s). With both SIP
and IAX2, now the question is why isn't
Is there an SRV bounty out there yet? $500 to the first person who
implements it (correctly :-) )..
Email for details.
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- Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SRV Bounty
Ronald Wiplinger wrote:
Matt Riddell wrote:
Matt Schulte wrote:
Is there an SRV bounty out there yet? $500 to the first person who
implements it (correctly :-) )..
Once somebody told me, if you do not know what it is, you
I've never actually core dumped but I *have* been able to hang asterisk
a couple times, I believed my problem was when I lost my mysql
connection. Why it lost connection is a mystery, the servers are on the
same testswitch. :/
I forgot which head ver it was, a couple weeks ago.
-Original
Word of warning, get the version 5 or higher FXS cards with the ADIT600,
else you will have echo problems. This is just from personal experience.
Supposedly the 5 and higher cards have dynamic impedance adjustment,
it's worth it.
Matt
-Original Message-
From: Peter Hoppe
Does anyone have some decent Nagios scripts out there that do more than
monitor the proc itself? Rather than reinvite the wheel, figured I'd
ask. I already saw the one on the wiki.
Matt
___
Asterisk-Users mailing list
Had a good question for the list, it seems whenever I work in an
Asterisk console or on the machine normally I get jitters on any audio
going through it. Especially if you did file copies or a 'ps ax' for
example. I was wondering if there was a proper way to 'nice' the
asterisk proc's? Cisco does
We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What
we're trying to avoid is hardcoding the IP address in the ACL. We were
trying to match by TOS set by Asterisk however it seems we've run into a
snag where the packet TOS tends to get reset somewhere on our network.
Has anyone
I'm assuming asterisk does not have a SIP jitter buffer in place? Any
ideas on how to help with this going over a data T1 where VoIP is shared
with regular traffic? Problem is when people are downloading the voice
is jittery, even lossy.
Matt
Yes yes, your right. I forget these switches are smart!!! ;-)
-Original Message-
From: Julio Arruda [mailto:[EMAIL PROTECTED]
Sent: Monday, January 03, 2005 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] QOS / Cisco / Asterisk
Matt
What's wrong with doing it by port?
We're actually using SIP to terminate calls, going by rtp.conf the ports
could range several thousand ports. What we're going for is only
honoring TOS for that particular customer, luckily these are T1
customers hosted on our routers. They understand that
Title: Message
Yes
yes, we've been through all that actually :-) We did find out it was one of the
3550's reseting the TOS.
-Original Message-From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent:
Tuesday, January 04, 2005 2:40 PMTo:
Ok, we're trying to use Asterisk as a PBX in our office. Our original
plan was to use a Cisco 7960 with a 7914 attached. Short story is, no
one updated chan_sccp in a long time and the 7914 is questionable at
best anyway from what I've heard. We couldn't ever get chan_sccp to
compile, I went to an
on 'make'
chan_sccp.c: In function `load_module':
chan_sccp.c:653: warning: passing arg 4 of `ast_channel_register_ex'
from incompatible pointer type
Now compiling sccp_actions.c 743 lines
Now compiling sccp_channel.c 279 lines
sccp_channel.c: In function
use it and it works fine.
On Wed, 12 Jan 2005 08:07:11 -0600, Matt Schulte [EMAIL PROTECTED]
wrote:
Ok, we're trying to use Asterisk as a PBX in our office. Our original
plan was to use a Cisco 7960 with a 7914 attached. Short story is, no
one updated chan_sccp in a long time and the 7914
may be outdated though. Anyone
have any thoughts on this?
Matt
-Original Message-
From: Matt Schulte
Sent: Wednesday, January 19, 2005 8:12 AM
To: C F; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Operator Panels?
The problem we're having
There's a MOS scale for this kind of stuff
-Original Message-
From: Paul Fielding [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 19, 2005 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] G.729? Worth it?
Low bandwidth
Low CPU
It's called asternic, www.asternic.org .. The client is based on flash which
connects to a perl daemon on the server. It uses the manager (manager.conf)
interface to determine extension status. Pretty neat :-)
Matt
-Original Message-
From: David John Walsh [mailto:[EMAIL
I couldn't find this option, I'm running the latest stable there is an
unstable version, is it in that one?
-Original Message-
From: Nicolás Gudiño [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 19, 2005 9:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
I'm having the exact same issue on a brand new Dell Poweredge 700, using
FC2. It locks the machine totally.
-Original Message-
From: Michael Loftis [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 20, 2005 7:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
All,
One of our customers is using a Telrad PBX, we are providing
phone server through asterisk via a T1 using em directly connected to
the Telrad system. We're using a T1 cross cable as normal, the T1 part
works great. No alarms. When we try and dial out the Telrad using a
direct trunk
Yes, this is frustrating I know. In fact the wiki could be updated to
provide this info. Basically if you have the phones out of the box
(brand spankin new) then you probly have the SCCP image installed on it
by default. Your tftp server root will need a number of files to start
if this is the
That's very interesting, because we do the exact same thing and all the
phones light up (with line mailbox flashing).. What SIP ver are you
using on the 7960's? However it sounds like 135 isn't registered on all
the phones? What we did is bind the lines to multiple phones, 203 (our
tech mailbox)
Got fed up going round in circles in the end. all for $8 worth of
access :(
Technically, Cisco wants you to pay for those images :)
___
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Anyone have any ideas? I'm bangin my head on the wall over here :(
-Original Message-
From: Matt Schulte
Sent: Wednesday, January 26, 2005 7:24 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Telrad + EM T1 Trunk
All,
One of our customers is using a Telrad PBX
Bueller? Is this a lib of some kind? Google and lists bring up nada,
this is from ast cvs head latest on Fedora Core 3.
/usr/bin/ld: cannot find -lidn
collect2: ld returned 1 exit status
make[1]: *** [app_curl.so] Error 1
make[1]: Leaving directory `/usr/src/asterisk/apps'
make: *** [subdirs]
We have an old CAC and we're trying to get groundstart working on it, we
think it may be a dip switch setting. Does anyone have the config
settings and/or the manual to config this thing? Any help is greatly
appreciated :-)
Matt
___
, 2005 11:11 am, Matt Schulte wrote:
We have an old CAC and we're trying to get groundstart working on it,
we think it may be a dip switch setting. Does anyone have the config
settings and/or the manual to config this thing? Any help is greatly
appreciated :-)
Carrier Access is one of the very
We are a voip terminating company, we're using Channelbank with FXS
modules, Rhino, CAC, etc.. What we're wondering is, is how to would you
echo cancel a channelbank. Of course we're realizing that cancel'ing on
the T1 (on Ast) does no good (we think?) because the analog conversion
is at the
PROTECTED]
Sent: Saturday, January 29, 2005 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Channel Bank Echo
On Sat, 29 Jan 2005, Matt Schulte wrote:
We are a voip terminating company, we're using Channelbank with FXS
modules, Rhino, CAC, etc
Ok, so we went out and bought a 2650 per the lists advice, putting it
plainly
...
SSDD
DD = different dell, I won't even bother with SS :)
Matt
-Original Message-
From: Matt Schulte
Sent: Friday, January 21, 2005 7:23 AM
To: Asterisk Users Mailing List - Non-Commercial
Ok, here's an odd one. I would have opened a bug on this but last time I
tried that I got flamed.. :)
Problem: When proxy requests digest challenge (SIP) Asterisk responds
normally with the exception that for some reason it changes the FROM:
(Also changes Contact: )to what's in the original TO:
, as soon as you
modprobe wctdm, the NMI lights on the server light up and you have about
a minute before the server reboots itself.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Schulte
Sent: Monday, January 31, 2005 4:19 PM
To: Asterisk Users
Just tried this, same deal. A bunch of NMI errors and eventually locks
up.
-Original Message-
From: Matt Schulte
Sent: Tuesday, February 01, 2005 7:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] NMI issues...
Really... I *think* I
Which sip ver are you trying to install. Is it stuck in a loop or
anything?
-Original Message-
From: Nicolas Chabbey [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 03, 2005 7:18 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960G phone crashes during SIP
you say? We upgraded all of ours in our office to 7.3 without
a problem.
-Original Message-
From: Nicolas Chabbey [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 03, 2005 7:51 AM
To: Matt Schulte
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
? What's wrong with the current jitterbuffer..
-Original Message-
From: joachim [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 08, 2005 2:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] jitterbuffers - suggested settings
I recommend
Try README.udev in the zaptel src directory..
-Original Message-
From: Daniel del Castillo [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 10, 2005 8:13 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Configuring Asterisk
Hey list,
I'm having problems to get
I get this when adding a user in ser (using serctl)
[EMAIL PROTECTED] sbin]# ./serctl add +18165551212 blahblah [EMAIL PROTECTED]
MySql password:
error: 400; check if you use aliases in SER
Um error 400?? I'm lost. no docs, frustrated. venting.
Matt
LOL, I'm a dumba$$ please ignore :-)
-Original Message-
From: Matt Schulte
Sent: Tuesday, February 15, 2005 2:41 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Ser 0.9.0 adding a user?
I get this when adding a user in ser (using serctl)
[EMAIL PROTECTED] sbin
: [Asterisk-Users] Ser 0.9.0 adding a user?
Matt Schulte wrote:
LOL, I'm a dumba$$ please ignore :-)
Might help to post what you did wrong for the archives...although, I
guess it isn't really Asterisk related.
:)
--
Cheers,
Matt Riddell
___
http
AT-320EE
Anyone try these? Do they work? any reviews? I couldn't find jack on
google..
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I could host it on my k-rad 56k sportster USR modem!
-Original Message-
From: Sergey Kuznetsov [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 19, 2005 7:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] wiki down?
Or I can host it.
I
I guess it could/would depend on the quality of the codec your using,
which ones are you using? (*not* for phone secks!)
-Original Message-
From: Mark Benson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 22, 2005 6:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
you and everyone else :-)
From: Daiku [mailto:[EMAIL PROTECTED]
But i AM looking for info on another IAX capable device - like the
IAXy, but more user
friendly, as it were...
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Found this on the wiki, is this still true? If so then what's the
alternative?
Default
srvlookup=yes
If srvlookup is turned on, Asterisk supports DNS SRV lookups partially.
Currently, Asterisk only reads the first SRV entry without bothering
with priorities and weights. This option is turned
Anyone have comments on this? ty..
-Original Message-
From: Matt Schulte
Found this on the wiki, is this still true? If so then what's the
alternative?
Default
srvlookup=yes
If srvlookup is turned on, Asterisk supports DNS SRV lookups partially.
Currently, Asterisk only reads
Here's a good one for the group, I have 2 Ast servers behind a NAT
(Sonicwall :-( ) connecting to the same server outside the NAT. Each of
the 2 boxes behind register to the outside server. What I am wondering
is, would there be a problem if both servers behind the NAT were
listening on port 4569,
Disabled echo canceller because of tone (tx) on channel 10
I understand that the PSTN companies use their own echo canceller's,
send a tone across 2100hz, the problem we're having is people are
complaining of echo on random calls. I'm assuming this may be the cause.
Is their anyway to 'ignore'
Title: Message
We use
NI2, it's the "standard" for north american telco. We tried another I can't
remember for the life of me which it was, but we had the least problems with
NI2.
-Original Message-From: Jon Bebeau
[mailto:[EMAIL PROTECTED] Sent: Friday, November 19, 2004
This maybe a simple question however I can't find a way to do this, I'm
wanting to EITHER:
Pass SIP digest authentication via dialplan (extensions.conf)
OR
Make Asterisk realize that the incoming peer in sip.conf doesn't have to
authenticate.
The reason I have this is because I'm connecting
All,
We are using a SIP provider that is expecting 0-15 response for
fmtp. Our CVS Head asterisk server is sending 0-16, I looked up an rfc
and it stated:
RTP Payloads for Telephone Signal Events
RFC 2833
Henning Schulzrinne, Scott Petrack.
May 2000
Implementation
Does anyone know if Asterisk can accept cleartext auth (SIP), as in it
recv's a call destined to:
1234:[EMAIL PROTECTED]
The problem I'm having is simply for faxing, normal calls come in as
g729 and of course we need ULAW for faxes.
sip.conf snippet
[sipfarm]
insecure=very
I found a few mentions of the 7914 being used with Asterisk, these all
covered SCCP/skinny though. Does anyone know if the 7914 can even be
used with SIP? If so, any pointers? Is it a services thing? Anyone get
the operator (line/extension status) to work with it. Thanks for the
help, Cisco
Thanks for the info
-Original Message-
From: Jeffrey C. Ollie [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 15, 2004 12:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco 7960 SIP + 7914
On Wed, 2004-12-15 at 11:54 -0600, Matt
Any ideas? I edited the Makefile as instructed, ty.
Now compiling sccp_channel.c 279 lines
sccp_channel.c: In function `sccp_channel_send_callinfo':
sccp_channel.c:48: structure has no member named `callerid'
sccp_channel.c:49: structure has no member named `callerid'
Subject: Re: [Asterisk-Users] chan_sccp compile problem w/ CVS head?
Seems that the author of sccp_channel.c hasn't upgraded his code.
You can fix this by replacing all instances of chan-callerid with
chan-cid.cid_num
-Matthew
- Original Message -
From: Matt Schulte [EMAIL PROTECTED
Has anyone had problems with using hold on a 7960 SIP firmware? The
problem is when the 7960 puts a call on hold and you take it off hold
again, the 7960 outbound audio is delayed on the other end. Sometimes up
to a few seconds. I've tried a couple different things, making the
other end a diff
The first example wasn't even touching SER..
7960sip -- asterisk -- IAX2 -- PRI
:/
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 16, 2004 9:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco
ala cisco 7960
-Original Message-
From: Matt Schulte
Sent: Thursday, December 16, 2004 10:34 AM
To: 'Paul A Brown'
Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems
Sure thing, the biggest problem I had was getting the SIP filenames
working correctly for updating
Anyone???
-Original Message-
From: Matt Schulte
Sent: Thursday, December 16, 2004 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems
The first example wasn't even touching SER..
7960sip -- asterisk -- IAX2
I asked this question once before with no answer. Hopefully someone can
help me as I cannot see a way to do this. I am wanting to differentiate
inbound calls voice from FAX. The purpose of course voice gets g729 and
FAX gets 711 (ulaw). The problem I'm having is everytime it matches the
SIP peer
diff codecs?
Matt Schulte wrote:
I asked this question once before with no answer. Hopefully someone can
help me as I cannot see a way to do this. I am wanting to differentiate
inbound calls voice from FAX. The purpose of course voice gets g729 and
FAX gets 711 (ulaw). The problem I'm having
: [Asterisk-Users] One SIP peer use 2 diff codecs?
Matt Schulte wrote:
So what's the work around? Have faxes come from a diff IP?
Well have them come into a different user/friend at least. The IP can
be the same if you are authenticating on username/secret rather than IP
Has anyone had any success with the Rhino CB-24? I can't get mine to
work, I tried all the obvious settings. The cb-24 gets stuck at init ESF
framing, as if it's not seeing the t1 card at all. It does get a t1
carrier (detecting voltage??)
Help!
Thanks..
Everything appears to look good on the
I agree, I got my first IAXy yesterday. I couldn't for the life of me
get DHCP to work, then I remember it was plugged into a VLAN on a Cisco
Switch (3550). Spanning tree always waits to bring up the vlan on
ports, unless specified otherwise. It would appear the IAXy only sends
an initial DHCP
Ok, I searched the lists and found no definitive answer. I'm assuming
the IAXy has some primitive form of echo cancel, is there anyway to
adjust this? Or any ideas on what to do instead. Here's the setup, this
will not be a typical setup for our company however, well whatever.
Anyway it looks
I am connecting Asterisk to Asterisk to PSTN (Either by SIP or PRI) and
am having some issues dealing with busy signals. I have the HANGUPCAUSE
dial result macro in place to generate my hangup causes. I get a
hangupcause on my gateway machine with a code of 34, here's the code:
... -snip-
exten =
Interesting, would this be considered a bug or is it rather intentional?
Or is that a dumb question ;-)
-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED]
IAX does not correctly set the HANGUPCAUSE for a LOT of things. Look at
DIALSTATUS or look at the dial-result macro
Now if one could only find a way to adapt an FXS module! :-)
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 28, 2004 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device
00101/2 UNKN
Format unknown??!! Ideas?
sip.conf
-snip-
[811]
host=dynamic
type=friend
context=matt-desk
videosupport=no
username=811
secret=xxx
[EMAIL PROTECTED]
callerid=Matt Schulte +1314xxx
reinvite=no
canreinvite=no
disallow=all
allow=g729
ty.
I purchased yesterday two G729
:17, Matt Schulte wrote:
I am having the same problem, it doesn't work on my SNOM either. Below
is my sip.conf .. On both sipura and SNOM I get same results, I can
hear voice but not send voice.
When you do a show g729 on the CLI do you get that the license is intalled?
You should
I use VMware + (Generic linux flavor) + Asterisk, for testing. Works
great, sound and mic work even. Kind of a bloated aproach seeing you
need ~128meg ram to even boot the OS but still it's fun to play with..
Matt
-Original Message-
From: Michael Giagnocavo [mailto:[EMAIL
This is still broken, I updated to the latest CVS. Flashed the Sipura and still no
dice, does anyone out there have any ideas? Thanks.
-Original Message-
From: Matt Schulte
Sent: Friday, October 29, 2004 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE
This may be a no brainer for some of you out there, simply put it seems
that we have a problem passing DTMF from IAX to SIP. The digits cannot
be heard coming from the IAX side nor do they seem to register in
Asterisk. This seems to happen with any Codec we use so that part has
been ruled out.
Here's a thought, anyone have ideas on how you could take registrations
from SIP/IAX users and run an AGI command using Asterisk? My goal would
be to enter the user/IP (after user reg's) into a MySQL database then
have other asterisk servers read from the same db. This would be for the
sake of
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Greenberg
Sent: Wednesday, January 25, 2006 6:30 AM
To: Asterisk Developers Mailing List; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-dev] No audio? Update
Could anyone either recommend a website or howto on optimizing Linux to
run asterisk. Such examples of what I mean are..
Renice of asterisk pid's
Forcing irq smp_affinity (For interupt hogging T1 cards)
.. That kind of stuff, I looked on the wiki and nothing directly
mentions server
All, I'm not sure how to word this question but we're noticing a lot of
our asterisk boxes no longer have multiple asterisk child processes.
i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used
to seeing 8+ .. There is no rhyme or reason to it, and we're using the
safe_asterisk
All,
I'm having a heck of a time getting hdlc to work on kernel
2.6.11.9 .. I compiled hdlc, hdlc_gen, hdlc_cisco, hdlc_raw, into the
kernel (note into, and not 'modules').
System comes up, I configured zaptel.conf
span=1,0,0,esf,b8zs
nethdlc=1-24
modprobe wct4xxp
ztcfg
sethdlc hdlc0
: [Asterisk-Users] HDLC/Zaptel/Kernel 2.6.11(.9)
Matt Schulte wrote:
All,
I'm having a heck of a time getting hdlc to work on kernel
2.6.11.9 .. I compiled hdlc, hdlc_gen, hdlc_cisco, hdlc_raw, into the
kernel (note into, and not 'modules').
System comes up, I configured zaptel.conf
span
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