The update worked like a charm! Hold music is as cheesy as ever!
Thanks much, this list is a life saver!
Dan
------------------------------
Message: 2
Date: Fri, 18 Mar 2005 09:16:59 -0600
From: Eric Wieling <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Redhat 9 Music on hold
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii; format=flowed
Jason Becker wrote:
> Daniel Burget wrote:
>
>> I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines
>> connected via TE405P. Everything works great, except MOH. I added an
>> exten with MusicOnHold(30), and it plays just fine. Conferences have
>> music when no one is in. I have SIP phones. When I place a call on hold,
>> the CLI give no indication the call is on hold. I have set
>> musiconhold(default) everywhere, removed it from everywhere, nothing
>> seems to help. I am using 59r of MPG123, and do not have MPG321
>> installed.
>> I did a 'make mpg123' from asterisk, make no difference.
>
>
> I believe it is a bug:
>
> http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/85000
>
> although I don't know if a bug was ever filed. I had a cursory look at
> the time we were bitten by this but couldn't find one. Pulling a newer
> CVS Stable and rebuilding resolved the issue.
And if you are on the asterisk-cvs mailing list, you would have seen a
fix being added yesterday. See: http://www.lists.digium.com/
--Eric
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
------------------------------
Message: 3
Date: Fri, 18 Mar 2005 10:18:26 -0500
From: Giovanni Powell <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Meetme2 compilation problem
To: Anil Kumar K <[EMAIL PROTECTED]>, Asterisk Users Mailing List -
Non-Commercial Discussion <[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
I'm sure there was a patch for meetme2 regarding compilation... google
for meetme2 + patch. It worked for me.
------------------------------
Message: 4
Date: Fri, 18 Mar 2005 09:23:11 -0600
From: Eric Wieling <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Undocumented "exten" syntax?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii; format=flowed
John Goerzen wrote:
> Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
> extensions.conf lines:
>
> exten => s,1,SetVar(SET_EMERG_FLAG=0)
> exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
> exten => s,n,SetGlobalVar(EMERGENCY=1)
> exten => s,n,SetVar(SET_EMERG_FLAG=1)
> exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
> exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)
I hope the wiki page mentions that the "n" priority is only supported
in CVS-HEAD, not 1.0.x.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
------------------------------
Message: 5
Date: Fri, 18 Mar 2005 16:23:35 +0100
From: Martin van den Berg <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] newbe question sip.conf
To: [email protected]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
Dear Gurus,
Just installed Asterisk and it runs just fine. Have made a simple
extension and sip configuration which works nice also. But still a
question.
My (simple) extension is as follows:
extensions.conf:
[default]
extern => 1001,1,Dial(SIP/1001)
extern => 1002,1,Dial(SIP/1002)
and the sip.conf:
[1001]
type=friend
host=dynamic
canreinvite=no
disallow=all
allow=alaw
[1002]
; copy of 1001
type=friend
host=dynamic
canreinvite=no
disallow=all
allow=alaw
So far, so good. But if I would like to deploy e.g. 100 sip phones, I
would have to add 100 sections?
What I would like to do, is group 'm like:
extensions.conf:
[default]
exten => _1XXX,1,Dial(SIP/norm,20)
and in the sip.conf:
[norm]
type=friend
host=dynamic
canreinvite=no
disallow=all
allow=alaw
But this doesn't seem to work. Any suggestions?
Thanks Martin.
------------------------------
Message: 6
Date: Fri, 18 Mar 2005 10:28:09 -0500
From: Noah Miller <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Re: Polycom vs. Cisco IP Phones
To: "[email protected]"
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> If you've considered the Snom, you might also want to test a
> Zultys 4x4 or 4x5. I picked a 4x5 up off of ebay recently and
> have been pleasantly surprised by it. While I don't currently
> have a Polycom to compare it with, I would rank the audio
> quality equal to the Cisco's. It also just 'does the right
> thing' with multiple lines - only one registration, no hints
> needed. Can be configured through TFTP with both default and
> phone specific config files. Software updates are freely
> available from the Zultys website.
I took a look at the Zultys phones when I was first shopping around.
One of their reps was kind enough to lug an entire working phone setup
into our office. He had some 4x4's and 4x5's and also a Cisco 7960
(just to show that their system was open standards compliant). I liked
the 4x5's ease of use, the 4 port network switch, the native PoE, and
the hard buttons for holding and transferring. Much to his chagrin,
though, I was actually much more impressed by the 7960. The 4x4's and
4x5's just looked like lower quality equipment. I suppose it didn't
help that the plastic casing on his 4x4 was cracked and broken.
In the end I went with neither, though, because the Polycom units were
so much cheaper.
------------------------------
Message: 7
Date: Fri, 18 Mar 2005 09:31:55 -0600
From: Eric Wieling <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] leaky reload
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii; format=flowed
Thomas Andrews wrote:
> If I comment out the following line in zapata.conf I would expect
> asterisk to "forget" the cli information for that channel when I reload:
>
> callerid="Uniden Dead" <(256) 428-6125>
>
> ... but it doesn't; I have to restart asterisk for it to take effect.
> The funny thing is that the reverse is *not* true - ie if I uncomment
> the line and "reload" then it learns about the caller id "Uniden Dead".
>
> Why is this a one-way process ?
Issuing a "reload" to asterisk does not correctly reload
/etc/asterisk/zapata.conf. This has been the case forever.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
------------------------------
Message: 8
Date: Fri, 18 Mar 2005 16:35:45 +0100
From: "Florian Overkamp" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] newbe question sip.conf
To: "'Martin van den Berg'" <[EMAIL PROTECTED]>, "'Asterisk Users
Mailing List - Non-Commercial Discussion'"
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
Hi,
> -----Original Message-----
> Just installed Asterisk and it runs just fine. Have made a simple
> extension and sip configuration which works nice also. But still a
> question.
>
> My (simple) extension is as follows:
>
> extensions.conf:
> [default]
> extern => 1001,1,Dial(SIP/1001)
> extern => 1002,1,Dial(SIP/1002)
>
> and the sip.conf:
> [1001]
> type=friend
> host=dynamic
> canreinvite=no
> disallow=all
> allow=alaw
>
> [1002]
> ; copy of 1001
> type=friend
> host=dynamic
> canreinvite=no
> disallow=all
> allow=alaw
>
> So far, so good. But if I would like to deploy e.g. 100 sip phones, I
> would have to add 100 sections?
Yup, you would. Which is why we all download or develop tools to automate
that kind of thing :)
Best regards,
Florian
------------------------------
Message: 9
Date: Fri, 18 Mar 2005 10:48:33 -0500
From: Josh Dady <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Polycom vs. Cisco IP Phones
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
On Mar 18, 2005, at 8:27 AM, Ben Ruset wrote:
>> 3. They don't realy support their phones, unless there is a hardware
>> problem.
>
> They don't support them with Asterisk, but if you don't tell them
> about it, they tend to be very good at working to resolve issues.
You have to know what issues they consider to be related to the
"platform". In general, "copper and plastic" issues (i.e., the phone
is in the wrong number of pieces) the direct customer support people
can help you with. As soon as you start talking about configuration,
though, don't bother trying to get any specifics out of them that
aren't in the admin guide -- you'd be wasting your time (as have many
on this list before you).
--
Joshua P. Dady
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------------------------------
Message: 10
Date: Fri, 18 Mar 2005 16:55:36 +0100
From: Anil Kumar K <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Meetme2 compilation problem
To: Giovanni Powell <[EMAIL PROTECTED]>
Cc: [email protected]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
I did the patch also . That didnt help me. I am using CVS head of 17th March .
Googling didnt give me much info other than this patch.
Thanks
On Fri, 18 Mar 2005 10:18:26 -0500, Giovanni Powell
<[EMAIL PROTECTED]> wrote:
> I'm sure there was a patch for meetme2 regarding compilation... google
> for meetme2 + patch. It worked for me.
>
------------------------------
Message: 11
Date: Fri, 18 Mar 2005 16:59:36 +0100
From: "Reuben Grech" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Group Ring after Timeout
To: <[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
Dear All,
I am listening to blips during conversations when I have an incoming call
from an X100P card. This does not happen on all conversations.
Any clues? :)
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------------------------------
Message: 12
Date: Fri, 18 Mar 2005 11:12:00 -0500
From: "dean collins" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Meetme2 compilation problem
To: "Anil Kumar K" <[EMAIL PROTECTED]>, "Asterisk Users Mailing List -
Non-Commercial Discussion" <[email protected]>,
"Giovanni Powell" <[EMAIL PROTECTED]>
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
Just use [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/
Meetme2 is automatically installed
Cheers
dean
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anil Kumar
K
Sent: Friday, March 18, 2005 10:56 AM
To: Giovanni Powell
Cc: [email protected]
Subject: Re: [Asterisk-Users] Meetme2 compilation problem
I did the patch also . That didnt help me. I am using CVS head of 17th
March .
Googling didnt give me much info other than this patch.
Thanks
On Fri, 18 Mar 2005 10:18:26 -0500, Giovanni Powell
<[EMAIL PROTECTED]> wrote:
> I'm sure there was a patch for meetme2 regarding compilation... google
> for meetme2 + patch. It worked for me.
>
_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 13
Date: Fri, 18 Mar 2005 11:04:23 -0500
From: Matt <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Where to place calling rule contexts?
To: Scott Nelson <[EMAIL PROTECTED]>
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
Got it.. thanks that worked...
On Fri, 18 Mar 2005 09:12:34 -0600, Scott Nelson <[EMAIL PROTECTED]> wrote:
> Matt wrote:
> > If I only want to give my sip users say local calling where do I put
> > that in the sip config?
> > ...
> > and the sip.conf looks like:
> > [200]
> > ...
> > context=from-internal
> > ...
>
> The "context=from-internal" is the key. You will need to create a
> context called "from-internal" that only includes local calling.
>
> For example:
>
> [from-internal]
> include => outbound-local
> include => internal-extensions
> include => outbound-emergency
>
>
------------------------------
Message: 14
Date: Fri, 18 Mar 2005 11:14:27 -0500
From: James Murray <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] asterisk reload
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I have read gracefully restarting asterisk on a regular basis is a
good idea. However the problem I have with doing this is that I need to
have all my users log back in , using AgentCallbackLogin and
AddQueueMember. Is there any way anyone has come up with to keep the
state of all users between restarts. I should probably also mention that
all the agents have there own passwords.
------------------------------
Message: 15
Date: Fri, 18 Mar 2005 11:10:14 -0500
From: Scott Griepentrog <[EMAIL PROTECTED]>
Subject: Re: Re[4]: [Asterisk-Users] TDM400P install problems
To: Alessio Focardi <[EMAIL PROTECTED]>, Asterisk Users Mailing
List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
Try using module wctdm instead. That solved a lot of headaches for me.
On Fri, 18 Mar 2005 16:02:23 +0100, Alessio Focardi
<[EMAIL PROTECTED]> wrote:
> Hello Dana,
>
> Friday, March 18, 2005, 3:40:21 PM, you wrote:
>
> DO> Can you run dmesg after that command and tell us what the relevant output
> is?
>
> # modprobe zaptel
> modprobe wcfxs
> FATAL: Could not open '/lib/modules/2.6.9-1.667/misc/wctdm.ko': No such file
> or directory
> # dmesg
> Zapata Telephony Interface Registered on major 196
> #
>
> I have to say that there are 2 cards in this server, this is my
> zaptel.conf
>
> fxoks=32-35
>
> loadzone = us
> defaultzone = us
>
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15
> dchan=16
> bchan=17-31
>
> was running cvs-head, now running 1.0.6
>
> It seems that when I call wcfxs wctdm is called instead.
>
> Any idea ?
>
> TNX !
>
> DO> On Fri, 18 Mar 2005 15:32:02 +0100, Alessio Focardi
> DO> <[EMAIL PROTECTED]> wrote:
> >> Hello Dana,
> >>
> >> Friday, March 18, 2005, 3:23:36 PM, you wrote:
> >>
> >> DO> If you have any FXS ports, use wcfxs.
> >>
> >> No, only green modules.
> >>
> >> But this is what I get when loading driver
> >>
> >> modprobe wcfxs
> >> FATAL: Error inserting wctdm
> >> (/lib/modules/2.6.9-1.667/misc/wctdm.ko): Unknown symbol in module,
> >> or unknown parameter (see dmesg)
> >> FATAL: Error running install command for wctdm
> >>
> >> What relates wcfxs to the wctdm that I was using previously ?
> >>
> >> Maybe deleting wctdm ....
> >>
> >> DO> On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi
> >> DO> <[EMAIL PROTECTED]> wrote:
> >> >> Hi,
> >> >>
> >> >> I was using a TDM400P with cvs version of asterisk, loading the driver
> >> >> with "modprobe wctdm".
> >> >>
> >> >> Some days ago I switched to stable version 1.0.6, where I found no
> >> >> trace of such module ... is wcfxo to be used instead ?
> >> >>
> >> >> Do I also have to change something in zaptel.conf ?
> >> >>
> >> >> Tnx for any help!
> >> >>
> >> >> --
> >> >> Best regards,
> >> >> Alessio mailto:[EMAIL PROTECTED]
> >> >>
> >> >> _______________________________________________
> >> >> Asterisk-Users mailing list
> >> >> [email protected]
> >> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> >> To UNSUBSCRIBE or update options visit:
> >> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> >>
> >> DO> _______________________________________________
> >> DO> Asterisk-Users mailing list
> >> DO> [email protected]
> >> DO> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> DO> To UNSUBSCRIBE or update options visit:
> >> DO> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >> --
> >> Best regards,
> >> Alessio mailto:[EMAIL PROTECTED]
> >>
> >>
> DO> _______________________________________________
> DO> Asterisk-Users mailing list
> DO> [email protected]
> DO> http://lists.digium.com/mailman/listinfo/asterisk-users
> DO> To UNSUBSCRIBE or update options visit:
> DO> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Best regards,
> Alessio mailto:[EMAIL PROTECTED]
>
> _______________________________________________
> Asterisk-Users mailing list
> [email protected]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Scott Griepentrog ([EMAIL PROTECTED])
------------------------------
Message: 16
Date: Fri, 18 Mar 2005 10:11:08 -0600
From: Jerry <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] CAC Access Bank Manual
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=US-ASCII; format=flowed
On Mar 18, 2005, at 2:40 AM, George Pajari wrote:
> Vicky Shrestha wrote:
>
>> The asterisk configuration and the channel bank configuration are
>> both set to esf and b8zs. Howerver I am still getting the framing
>> Error "Red and blinking". zttool shows there are no alarms.
>>
>> According to the manual, Framing Error (Red and Blinking )means
>>
>> "Network T1 is out of frame (received signal cannot be framed to ESF
>> or D5 as configured by T1 Option switch 4)"
>>
>> I tried with both DIP switch on and off, but no help.
>>
>> Any ideas ?
>>
>> Is my card or channel bank bad ?
>>
> Probably not. More likely your CAC ABI is set up to run in TR08 mode
> which is incompatible with standard T1 framing.
>
> Check the LIU board (the Line Interface Board -- as opposed to the FXO
> or FXS cards) and look at the PROM. It usually will be marked TR08. If
> that is the case you will need to order a D4/ESF upgrade kit. If you
> have a 1.x revision TR08 chip, you will need P/N 750-0018. If you have
> a 3.x revision TR08 chip, you will need P/N 750-0019.
>
Are you using a terminal to talk to the CAC? Depending on configuration
you may have the DIP switches disabled and changing them will do no
good. Connect to the CAC and ask it for the T1 configuration to verify
what it really is.
Also make sure you rerun ztcfg after any changes to zaptel.conf.
What does zttool tell you?
------------------------------
Message: 17
Date: Fri, 18 Mar 2005 16:15:24 -0000
From: "Kanishka Somaratne" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] reply a post
To: <[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"
Hi
how do i reply a question asked in this mailling list.
tks
Kanishka
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Message: 18
Date: Fri, 18 Mar 2005 16:20:51 +0000
From: Asterisk <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] asterisk reload
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
CVS head has an option to do this. persistentmembers is the option I think.
Julian LS.
James Murray wrote:
>
> I have read gracefully restarting asterisk on a regular basis is a
> good idea. However the problem I have with doing this is that I need
> to have all my users log back in , using AgentCallbackLogin and
> AddQueueMember. Is there any way anyone has come up with to keep the
> state of all users between restarts. I should probably also mention
> that all the agents have there own passwords.
>
>
> _______________________________________________
> Asterisk-Users mailing list
> [email protected]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
------------------------------
Message: 19
Date: Fri, 18 Mar 2005 11:18:27 -0500
From: "Kanuri, Seshu (Company IT)" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] reply a post
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
Do you know how to hit the reply button on the Outlook menu?
Just hit the reply button.
If you dont know this, send an email to [email protected]
if it is a user related topic.
Dont post business topics here.
Seshu
_____
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanishka
Somaratne
Sent: Friday, March 18, 2005 11:15 AM
To: [email protected]
Subject: [Asterisk-Users] reply a post
Hi
how do i reply a question asked in this mailling list.
tks
Kanishka
--------------------------------------------------------
NOTICE: If received in error, please destroy and notify sender. Sender does
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------------------------------
Message: 20
Date: Fri, 18 Mar 2005 09:19:33 -0700
From: "Kevin P. Fleming" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] asterisk reload
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Asterisk wrote:
>
> CVS head has an option to do this. persistentmembers is the option I think.
Yes, CVS HEAD has both persistent dynamic members and persistent agent
logins.
------------------------------
Message: 21
Date: Fri, 18 Mar 2005 17:19:55 +0100
From: Asterisk <[EMAIL PROTECTED]>
Subject: Re: Re: [Asterisk-Users] Meetme2 compilation problem
To: Anil Kumar K <[EMAIL PROTECTED]>, Giovanni Powell
<[EMAIL PROTECTED]>
Cc: "[email protected]"
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="utf-8"
Giovanni,
on ftp://ftp.vinkconsult.com/downloads
is a patched version of app_meetme2.c.
I patched and compiled it against the CVS unstable from today
Andre
----- Oorspronkelijk Bericht -----
ONDERWERP: Re: [Asterisk-Users] Meetme2 compilation problem
AFZENDER: Anil Kumar K
AAN: "Giovanni Powell"
CC:[EMAIL PROTECTED]
DATUM: 18-03-2005 16:56
I did the patch also . That didnt help me. I am using CVS head of
17th March .
Googling didnt give me much info other than this patch.
Thanks
On Fri, 18 Mar 2005 10:18:26 -0500, Giovanni Powell
wrote:
> I'm sure there was a patch for meetme2 regarding compilation...
google
> for meetme2 + patch. It worked for me.
>
_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
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------------------------------
Message: 22
Date: Fri, 18 Mar 2005 16:22:11 +0000
From: "Barry FAWTHROP" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] echo / delay problem
To: [email protected]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; format=flowed
I'm having with an echo or delay
I connect to the PSTN with a x100p and then connect a std. phone
to a FXS module on a TDM10B.
The std phone is only 2-wire so I know this is not helping.
(yes I have read the 2-wire 4-wire issue)
I have tried many echocancel values. The best thing to help was
rxgain and txgain. below is my current zapata.conf file
All help would be grateful. I have tried and tried for 2 weeks
it is rather annoying and irating to hear this delay/echo
I would call it a delay since you can hear the end of the sentence repeat
over
and over. Also every now and again it sounds like a underwater submarine
with ping and all.
Thanks in advance
Barry
[channels]
language = en
context = inbound
signalling = fxs_ks
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
echocancel = 16
echocancelwhenbridged = yes
echotraining = no ;; yes
rxgain = -2.0
txgain = -2.0
musiconhold = default
channel => 1
context = intern
signalling = fxo_ks
callwaiting = yes
usecallerid = yes
echotraining = no ;; yes
echocancel = 16
channel => 2
------------------------------
Message: 23
Date: Fri, 18 Mar 2005 11:22:44 -0500
From: Seth Remington <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Group Ring after Timeout
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion <[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain
On Fri, 2005-03-18 at 16:59 +0100, Reuben Grech wrote:
> Dear All,
>
> I am listening to blips during conversations when I have an incoming
> call from an X100P card. This does not happen on all conversations.
>
> Any clues? :)
Turn off call waiting in zapata.conf
callwaiting=no
-Seth
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
------------------------------
Message: 24
Date: Fri, 18 Mar 2005 13:26:41 -0300
From: "Matias G." <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] call a url and get a result in the dialplan
To: <[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
can a call a php script wich is located in a remote server, someting like
calling www.theserveraddress.com/scripts/validate?code=234234swq and get the
result which this script generates (a 0 or a 1) back in the dial plan in a
direct way or should I create a script which in turn does this?
I'm using * CVS HEAD.
Also I searched for this for I while but didn't manage to find anything but
SendUrl and PHP AGI
thanks
M.
------------------------------
Message: 25
Date: Fri, 18 Mar 2005 17:23:56 +0100
From: Alessio Focardi <[EMAIL PROTECTED]>
Subject: Re[6]: [Asterisk-Users] TDM400P install problems
To: Scott Griepentrog <[EMAIL PROTECTED]>
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii
Hello Scott,
Friday, March 18, 2005, 5:10:14 PM, you wrote:
SG> Try using module wctdm instead. That solved a lot of headaches for me.
There is no wctdm module in zaptel-1.0.6.tar.gz .....
So why when I call wcfxs ...
modprobe wcfxs
FATAL: Could not open '/lib/modules/2.6.10-1.770_FC3/misc/wctdm.ko': No such
file or directory
That does not look normal to me, I have built another kernel to try to
make this behavior go away, still no luck ....
Tnx anyway ...
SG> On Fri, 18 Mar 2005 16:02:23 +0100, Alessio Focardi
SG> <[EMAIL PROTECTED]> wrote:
>> Hello Dana,
>>
>> Friday, March 18, 2005, 3:40:21 PM, you wrote:
>>
>> DO> Can you run dmesg after that command and tell us what the relevant
>> output is?
>>
>> # modprobe zaptel
>> modprobe wcfxs
>> FATAL: Could not open '/lib/modules/2.6.9-1.667/misc/wctdm.ko': No such file
>> or directory
>> # dmesg
>> Zapata Telephony Interface Registered on major 196
>> #
>>
>> I have to say that there are 2 cards in this server, this is my
>> zaptel.conf
>>
>> fxoks=32-35
>>
>> loadzone = us
>> defaultzone = us
>>
>> span=1,1,0,ccs,hdb3,crc4
>> bchan=1-15
>> dchan=16
>> bchan=17-31
>>
>> was running cvs-head, now running 1.0.6
>>
>> It seems that when I call wcfxs wctdm is called instead.
>>
>> Any idea ?
>>
>> TNX !
>>
>> DO> On Fri, 18 Mar 2005 15:32:02 +0100, Alessio Focardi
>> DO> <[EMAIL PROTECTED]> wrote:
>> >> Hello Dana,
>> >>
>> >> Friday, March 18, 2005, 3:23:36 PM, you wrote:
>> >>
>> >> DO> If you have any FXS ports, use wcfxs.
>> >>
>> >> No, only green modules.
>> >>
>> >> But this is what I get when loading driver
>> >>
>> >> modprobe wcfxs
>> >> FATAL: Error inserting wctdm
>> >> (/lib/modules/2.6.9-1.667/misc/wctdm.ko): Unknown symbol in module,
>> >> or unknown parameter (see dmesg)
>> >> FATAL: Error running install command for wctdm
>> >>
>> >> What relates wcfxs to the wctdm that I was using previously ?
>> >>
>> >> Maybe deleting wctdm ....
>> >>
>> >> DO> On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi
>> >> DO> <[EMAIL PROTECTED]> wrote:
>> >> >> Hi,
>> >> >>
>> >> >> I was using a TDM400P with cvs version of asterisk, loading the driver
>> >> >> with "modprobe wctdm".
>> >> >>
>> >> >> Some days ago I switched to stable version 1.0.6, where I found no
>> >> >> trace of such module ... is wcfxo to be used instead ?
>> >> >>
>> >> >> Do I also have to change something in zaptel.conf ?
>> >> >>
>> >> >> Tnx for any help!
>> >> >>
>> >> >> --
>> >> >> Best regards,
>> >> >> Alessio mailto:[EMAIL PROTECTED]
>> >> >>
>> >> >> _______________________________________________
>> >> >> Asterisk-Users mailing list
>> >> >> [email protected]
>> >> >> http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> >> To UNSUBSCRIBE or update options visit:
>> >> >> http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> >>
>> >> DO> _______________________________________________
>> >> DO> Asterisk-Users mailing list
>> >> DO> [email protected]
>> >> DO> http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> DO> To UNSUBSCRIBE or update options visit:
>> >> DO> http://lists.digium.com/mailman/listinfo/asterisk-users
>> >>
>> >> --
>> >> Best regards,
>> >> Alessio mailto:[EMAIL PROTECTED]
>> >>
>> >>
>> DO> _______________________________________________
>> DO> Asterisk-Users mailing list
>> DO> [email protected]
>> DO> http://lists.digium.com/mailman/listinfo/asterisk-users
>> DO> To UNSUBSCRIBE or update options visit:
>> DO> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> Best regards,
>> Alessio mailto:[EMAIL PROTECTED]
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> [email protected]
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
------------------------------
Message: 26
Date: Fri, 18 Mar 2005 17:28:36 +0100
From: Giovanni Miano <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] I4l + HiSax
To: [email protected]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
I need HELP pls!
BRISTUFF: Bad Sound quality
CAPI : PTP Mode dont supported
mISDN : kernel is 2.4.x and not 2.6.x
HISAX : PTMP ok, PTP incoming ok but in outgoing asterisk dont
compose number(i listen dial tone and than i can compose number via
dtmf)
Asterisk CLI (g3 is group of Modem[i4l]/ttyI0 and ttyI1):
Called g3: 3453444444
(Channel is used but dont compose number)
Asterisk Log:
VERBOSE[10406]: -- Called g3:3453444444
Mar 18 23:41:01 DEBUG[10406]: Detecting DTMF inband with sw DSP on /dev/ttyI1
Mar 18 23:41:01 DEBUG[10406]: Dropping duplicate answer!
Mar 18 23:41:01 VERBOSE[10406]: -- Modem[i4l]/ttyI1 answered SIP/201-3e46
Mar 18 23:41:01 DEBUG[10406]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 55994:
Found
Mar 18 23:41:01 DEBUG[10406]: Ooh, format changed from unknown to alaw
Mar 18 23:41:06 DEBUG[10406]: Didn't get a frame from channel: SIP/201-3e46
Mar 18 23:41:06 DEBUG[10406]: Bridge stops bridging channels
SIP/201-3e46 and Modem[i4l]/ttyI1
------------------------------
Message: 27
Date: Fri, 18 Mar 2005 08:28:27 -0800
From: Trevor Peirce <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] PRI Cause Code Help
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Peter Svensson wrote:
>The two issues are only somewhat related. The RELEASE COMPLETE as an reply
>to a SETUP after having sent a CALL PROCEEDING is probably not allowed by
>the state transitions listen in q.931.
>
>
I've commented out a few lines of code to make sure * sends DISCONNECT
but I'm getting identical results. Seems like it doesn't matter if I
skip to RELEASE_COMPLETE or not.
>The in-band announcement is more related to whether we have sent a
>progress information element which states that in-band audio is available.
>I think Asterisk sends such a progress message almost as soon as possible.
>However, in this case the problem is a CALL PROCEEDING before the
>RELEASE_COMPLETE answering teh SETUP. The fact that the CALL PROCEEDING
>also includes a PROGRESS element is incidental.
>
>
Are you suggesting that * is telling the other side that we are making
noise and they shouldn't? My intent here is to have the telco say "I'm
sorry the numbern is not in service" instead of tying up one of our
lines for the duration of such a message. Likewise with Congestion....
odd part is that Busy works fine, as I mentioned in my original post.
Thanks for the posts Peter and Eric. Good to know I am going about
this in the right direction.
Trevor
------------------------------
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