[Asterisk-Users] Anyone run * on OS X ?
ted at indexc.com
ted at indexc.com
Mon Jan 26 18:45:12 MST 2004
With the 1U Apple G5 server at a good reliability/storage/pricepoint
level... got to thinking about compiling Asterisk on OS X.. at least for SIP
phone call switching, voicemail, etc. Has anybody attempted this? Email me
off list if this is too dev-heavy for the user list.
Thanks,
Ted W
-----Original Message-----
From: asterisk-users-request at lists.digium.com
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Sent: Monday, January 26, 2004 11:31 AM
To: asterisk-users at lists.digium.com
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Today's Topics:
1. Re: Has Nufone gone belly-up (Steve Underwood)
2. SIP - fax / voicemail (Dawid Mielnik)
3. Re: Has Nufone gone belly-up (Girish Gopinath)
4. app_queue and dialplan (Anton Yurchenko)
5. Know if a call is answered (Asterisk List)
6. Re: rc.local dont works (Jeroen)
7. Re[2]: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being
passed to caller (Frankie Gravato)
8. Re: OH323 doesnt hear ringing (Michael Manousos)
9. RE: Asterisk Indications (Christopher Lee)
10. Re[2]: [Asterisk-Users] Has Nufone gone belly-up (Frankie Gravato)
11. Questions regarding new echo cancellation features... (john)
--__--__--
Message: 1
Date: Mon, 26 Jan 2004 20:16:12 +0800
From: Steve Underwood <steveu at coppice.org>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Has Nufone gone belly-up
Reply-To: asterisk-users at lists.digium.com
Don Feuer wrote:
>Hi Everybody,
>
>In regards to what I see here, this looks like a whole .com flash back. I
>started a phone company that went belly up (CentreCom, the first Unified
>Communications company) because of customer service issues, lack of on-line
>information, and a lack of caring for the customer. Unfortunately, I had a
>number of big wigs forced on me by vc's and they placed big wigs in the
>company from AT&T who didn't know their head from their a..'s when it came
>to customers, nor did they care.
>
>
I understand your point, but saying you were behind a unified comms
company doesn't make the point awfully well. I was one of the people
behind another unified comms company. Pretty much 100% of them went
bellyup, as it seems nobody would buy the service. Too much free e-mail,
and other free services didn't help, though.
Sad, downcast, bitter, thought unified comms was a good idea :-)
Steve
--__--__--
Message: 2
From: "Dawid Mielnik" <D.Mielnik at elka.pw.edu.pl>
To: <asterisk-users at lists.digium.com>
Date: Mon, 26 Jan 2004 13:20:03 +0100
Subject: [Asterisk-Users] SIP - fax / voicemail
Reply-To: asterisk-users at lists.digium.com
Hi,
Just to clear things out.. Can asterisk transmit faxes over IP ? If not, are
there any works being done towards implementing t.38 on asteisk ?
Also dialing in from a mediatrix fxs sip gateway to voicemail, asterisk does
not see the digits entered after mailbox prompt. I have dtmftone settings
correct - inband (also tried others to make sure), however asterisk shows
'username not entered'. Any clues how to tackle this ? Chenking voicemail
from x-lite for example I dont have problems.
regards,
Dave
--__--__--
Message: 3
From: "Girish Gopinath" <gopinath_girish at hotmail.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Has Nufone gone belly-up
Date: Mon, 26 Jan 2004 17:55:26 +0530
Reply-To: asterisk-users at lists.digium.com
>And I don't care about your network, your services, or your contributions
>to Asterisk. Your behaviour in this matter is like that of a toddler in a
>sandpit, throwing sand back at the other kids then screaming "they started
>it".
>
>Grow up. Your prospective customers have.
>
>echo nufone.net >> killfile
>
But I respect him. I have never seen him or he has never answered my doubts
about Asterisk. Still i do respect him because of his contributions to
Asterisk. And I respect all who contribute to Opensource.
Even Mahatma Gandhi and Albert Einstein had problems in their character.
Don't we respect them?
Please stop this thread. It is not talking about Asterisk or Nufone. It just
says about people's attitudes.
Girish
_________________________________________________________________
Add glamour to your desktop. Let your screen sizzle.
http://server1.msn.co.in/msnchannels/Entertainment/wallpaperhome.asp
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--__--__--
Message: 4
Date: Mon, 26 Jan 2004 14:59:38 +0200
From: Anton Yurchenko <phila at dg.net.ua>
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] app_queue and dialplan
Reply-To: asterisk-users at lists.digium.com
Hello,
I`m trying to achive this:
1. when the initial call comes in it is served by a small queue with
short timeout so that at first caller hears only ringing
2. if nobody answers the call at that time or the queue is all full the
call goes to the Playback the message ( "please hold bla bla bla")
3. Then the call goes to another queue and he holds while the
music-on-hold plays a app_queue trys to reach the next free operator
4. after a timeout in second queue there is a Goto to play the message
again and then back into the second queue
I have it like this:
extensions.conf:
exten => 10,1,Queue(q1_short,tn)
exten => 10,2,Answer
exten => 10,3,Playback(please_hold)
exten => 10,4,Queue(q1,t)
exten => 10,5,Goto(3)
queue.conf:
[q1]
music = test
announce = test_anounce
timeout = 40
retry = 3
maxlen = 10
strategy = leastrecent
member => SIP/111
member => SIP/112
member => SIP/113
member => SIP/114
member => SIP/115
[q1_short]
music = test
announce = test_anounce
timeout = 15
retry = 3
maxlen = 3
strategy = leastrecent
member => SIP/111
member => SIP/112
member => SIP/113
member => SIP/114
member => SIP/115
but the broblem is when the q1_short is full, and the call goes to the
q1 it only plays the announce message and and no music on hold is played
and again the announce message is played. somehow the music on lod
doesn start. What am I doing wrong?
I run version CVS-12/01/03-14:50:57
Thanks
--
Anton Yurchenko<phila at dg.net.ua>
Digital Generation
--__--__--
Message: 5
From: "Asterisk List" <asterisklist at hotmail.com>
To: asterisk-users at lists.digium.com
Date: Mon, 26 Jan 2004 13:02:47 +0000
Subject: [Asterisk-Users] Know if a call is answered
Reply-To: asterisk-users at lists.digium.com
Hello:
I have an asterisk server answering SIP calls.
Whenever a call comes, asterisk answers, plays a gsm file (information) and
dials to another SIP phone.
Using asterisk Master.csv file I only have one record, and don't know if the
second call is answered.
I only know this if:
- The called phone is busy
- The called phone doesn't answer in X seconds (the parameter in Dial)
But I can't see a difference between an answerd call and when the caller
hangs up whith the other phone is still ringing. The call is always
"ANSWERED" and the last command is Dial with the SIP address as the
parameter.
How can I do this? Could it be done with an AGI script?
Sorry if this is a basic question, I have been searching for a solution for
weeks and don't know what to do.
Thanks in advance,
Rober T.
_________________________________________________________________
Reserva y planifica tu viaje online. http://www.msn.es/Viajes/
--__--__--
Message: 6
Date: Mon, 26 Jan 2004 14:23:05 +0100
From: Jeroen <linux at comresult.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] rc.local dont works
Reply-To: asterisk-users at lists.digium.com
Hi Miklos,
I have the same problem here in RH90 - have you found any solution?
Or does anybody else know why (safe_)asterisk does not start using
rc.local? (normally I start * as root user)
Cheers
Jeroen
--__--__--
Message: 7
Date: Mon, 26 Jan 2004 08:25:47 -0500
From: Frankie Gravato <nanog at cfsdigital.com>
Organization: Cfsdigital
To: Rich Adamson <asterisk-users at lists.digium.com>
Subject: Re[2]: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being
passed to caller
Reply-To: asterisk-users at lists.digium.com
Hello Rich,
Sunday, January 25, 2004, 8:01:25 PM, you wrote:
RA> It would probably help if you used a packet sniffer (eg, ethereal) to
look
RA> at the traffic, or at least provide the list with a useful clue other
then
RA> it doesn't work.
RA> ------------------------
>> same here, when i recive an incoming call from x100p to line 1 on
>> sipura, i can hear them but people can't hear me im using 1.0.24 on my
>> firmware
>>
>> Miguel
>> On Sun, 2004-01-25 at 20:54, Chris Higgins wrote:
>> > Frankie Gravato wrote:
>> >
>> > >
>> > > I've been beating my head for 5 hours to figure out why my
asterisk
>> > > server or sipura isn't passing my voice over to the caller. It seems
i
>> > > can hear the caller but they can't hear me it seems either
the
>> > > asterisk or the sipura isn't passing this information.
>> > >
>> > > Here's my setup specs
>> > >
>> > > asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service
-
>> > > Voicepulse Service and DID's
>> > >
>> > > when i get Phone call using the Voicepulse or Pstn the caller
can't
>> > > hear me or barely hear me. The Sipura is running Firmware 1.20
and
>> > > calls are being passed using Ulaw Codec? Anyone out there in
the
>> > > asterisk community please oh please help me before i do something
that
>> > > my asterisk server won't like.
>> > >
>> > >
>> >
>> > I just received my Sipura on Friday and have been testing it
extensively
>> > over the weekend. I have noticed an issue similar to what you mention
>> > above. For the record, the sipura tells me I'm running software
version
>> > 1.0.20. Also, there is NO nat configuration that is causing my
problem.
>> >
>> > When I receive a call over my X100P and dial my 3 SIP phones (one gs
>> > budgetone 100, two analong phones through sipura), if I answer the
>> > analong phone connected to line 1 of the sipura, the caller cannot hear
>> > anything. I've only noticed this problem in this exact scenario. The
>> > other situations listed below have no problems whatsoever and audio
>> > works in both directions:
>> >
>> > 1. Call from sipura line 1 to any internal SIP phone.
>> > 1. Call from any internal SIP phone to sipura line 1.
>> > 2. Call from sipura line 1 out through X100P.
>> > 3. Call into my X100P from outside and answer sipura line 2.
>> > 4. Call into my X100P from outside and answer sipura line 2 and THEN
>> > transfer to sipura line 1.
>> > 5. Call into my X100P from outside and answer sipura line 1 (the caller
>> > cannot hear audio for this leg of the conversation), TRANSFER to any
>> > other line, and transfer back to sipura line 1. After the second
>> > transfer, the caller can hear audio from sipura line 1.
>> >
>> > I don't know what is special about line 1. I've switched my analog
>> > phones across the two ports on the sipura to make sure it wasn't one of
>> > my phones (not that I thought it was anyway).
>> >
>> > Frankie, have you tried the same experiment, but pulled your analog
>> > phone from line 1 and put it in line 2?
>> >
>> > Has anyone else seen issues like this with line 1 on a sipura?
>> >
>> > Thanks..
>> >
>> > -- Chris
>> > _______________________________________________
>> > Asterisk-Users mailing list
>> > Asterisk-Users at 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
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
RA> ---------------End of Original Message-----------------
RA> _______________________________________________
RA> Asterisk-Users mailing list
RA> Asterisk-Users at lists.digium.com
RA> http://lists.digium.com/mailman/listinfo/asterisk-users
RA> To UNSUBSCRIBE or update options visit:
RA> http://lists.digium.com/mailman/listinfo/asterisk-users
I'll be trying that as my next step but it seems that my other fresh -
sipura 2000 unit that was sitting in the box which is running 1.0.15
firmware seems to work seamless so i find it odd ? that brand new unit
works while the upgraded firmware ones don't?
I'm not the only one having this same exact issue I've received 4
emails relating to the same issue from other users. So there's some
kind trend going on with this?
--
Best regards,
Frankie (fgravato at cfsdigital.com)
mailto:nanog at cfsdigital.com
--__--__--
Message: 8
Date: Mon, 26 Jan 2004 15:27:57 +0200
From: Michael Manousos <manousos at inaccessnetworks.com>
Organization: inAccess Networks
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] OH323 doesnt hear ringing
Reply-To: asterisk-users at lists.digium.com
Aaron Martin wrote:
> I have Asterisk running with a combination of SIP and H323 clients. I
> am using the OH323 module instead of the H323 one.
>
> When the SIP clients ring each other, they can hear a ringing noise in
> the ear peice to let them know that the other parties phone is ringing.
> However, when the H323 client rings a SIP client, there is no ringing
> sound at all, although as soon as the called party picks up the phone,
> everything works fine. This is the entry from my extensions.conf:
>
> exten => _7[5-9]X,1,Dial(SIP/${EXTEN},20,rt)
> exten => _7[5-9]X,2,Playback(vm-nobodyavail)
> exten => _7[5-9]X,3,Hangup
>
> I assume that because I havr the 'r' in the dial string, the calling
> party should hear a ringing noice. Any ideas?
What are the contents of your oh323.conf? (please hide
passwords/IPs, if any).
Michael.
--__--__--
Message: 9
From: "Christopher Lee" <chris at datachaos.com.au>
To: <asterisk-users at lists.digium.com>
Subject: RE: [Asterisk-Users] Asterisk Indications
Date: Mon, 26 Jan 2004 23:35:31 +1000
Organization: Data Chaos
Reply-To: asterisk-users at lists.digium.com
> I am really having trouble with this. I have been making changes to
> indications.conf, but my changes are not taking effect. I have shut down
> Asterisk, re-run ztcfg, reloaded the zaptel modules, all to no avail --
> I get the same tones consistently...
>
> What should I be doing to bring in a new indications.conf?
No problems here with making changes to indications.conf and then doing a
"reload" on the console, the changes then take effect.
My testing involves calling from a SIP handset to a dummy extension setup to
answer and playback the tones I want to check.
; Test Australian ringing tones - indications
exten => 906,1,Answer
exten => 906,2,Wait(1)
exten => 906,3,Playtones(ring)
exten => 906,4,Wait(12)
exten => 906,5,Playtones(busy)
exten => 906,6,Wait(5)
exten => 906,7,Hangup
It sounds like you must have FXS extensions your trying to test the
indications on? I don't have an FXS card in my machine to test with, so I'm
not sure how it works, but it should still be the same, as a reload
definitely re-reads the indications.conf configuration.
Regards,
Chris Lee
--__--__--
Message: 10
Date: Mon, 26 Jan 2004 08:42:17 -0500
From: Frankie Gravato <nanog at cfsdigital.com>
Organization: Cfsdigital
To: John Baker <asterisk-users at lists.digium.com>
Subject: Re[2]: [Asterisk-Users] Has Nufone gone belly-up
Reply-To: asterisk-users at lists.digium.com
Hello John,
Sunday, January 25, 2004, 11:36:55 PM, you wrote:
JB> I tried a couple times to talk to them about service. How much it
costs,
JB> how it works, etc. Just common stuff you might find on a website. I
left a
JB> message and nobody returned my call; I went with voicepulse instead.
JB> John
JB> ----- Original Message -----
JB> From: "Sean Cheesman" <scheesman at macarthur-group.com>
JB> To: <asterisk-users at lists.digium.com>
JB> Sent: Sunday, January 25, 2004 9:49 PM
JB> Subject: RE: [Asterisk-Users] Has Nufone gone belly-up
JB> funny... I got an immediate response, and within 1 hour had my account
JB> activated. and this was today.....
JB> -----Original Message-----
JB> From: Chris Albertson [mailto:chrisalbertson90278 at yahoo.com]
JB> Sent: Sunday, January 25, 2004 10:36 PM
JB> To: asterisk-users at lists.digium.com
JB> Subject: Re: [Asterisk-Users] Has Nufone gone belly-up
JB> I think discussions about which VOIP service providers are best or "not
JB> best" is a reasonable topic for an asterisk user's list. After all,
JB> selecting a provider is a big deal
JB> Rants about some company just make the "rant-er" look bad and should be
JB> avoided.
JB> Back to NuFone. I've just started testing thier service. So far so
JB> good. The sound quality and pricing is very much like Iconnect but
JB> NuFone will accept IAX2, a big plus.
JB> But my e-mails are ignored. Actually sending e-mails was part of the
JB> test of this vendor. I'm testing to see if they answer e-mail. So far,
JB> not good. Maybe NuPhone needs to loosen up their spam filters or at
JB> least scan the spam folder every few days.
JB> --- Brian West <brian at bkw.org> wrote:
>> Have you tried to call them? Your emails could have been caught up in
>> a spam filer or such.... I use nufone daily for our 888 service. I
>> talk to
>> Jermey daily. So I dont know what your beef is but your rant has no
>> place
>> on this mailing list if you are having problems and have spent any
>> time
>> trying to get someone on the phone you must be doing something wrong.
>> I
>> do know about 2 weeks ago GoDaddy screwed up nufone.net's domain and
>> it
>> was sent off into LALA land for a few days.
>>
>> So please keep your rants off the list.
>>
>> bkw
>>
>> On Sat, 24 Jan 2004, Sathya wrote:
>>
>> > Folks,
>> >
>> > I've ordered a new account from Nufone last month. Transferred
>> money to
>> > Nufone through their paypal account. I had communication with
>> Nufone sales
>> > up until two weeks back. Since then there were no replies to my
>> emails.
>> >
>> > I am afraid with this kind of unresponsiveness how one would run a
>> reliable
>> > service with this company. Have no bad feeling with Jeremy as the
>> author of
>> > widely used h323 channel, but my concern is about the company
>> NuFone. Lot of
>> > newcomers when asked for IAX termination/Origination we say NuFone.
>> I just
>> > want to record my experience so far, as it would help anyone wanted
>> to start
>> > with this company. I can live with the fact that they do not have
>> any web
>> > based interface for customers to do anything with the service as
>> claimed by
>> > the website. But cannot understand taking two weeks to answer a
>> freaking
>> > email. (Well in the absence of trouble ticketing system or web
>> based access
>> > to accounts, email is the only way to contact Nufone)
>> >
>> > I have services running with Iconnect and Voicepulse etc and I was
>> just
>> > trying to use Nufone being well recommended in this list.
>> >
>> > I am not here to tarnish Nufone name but I have no option but to
>> ask the
>> > community since there is no response to my emails or there is no
>> indication
>> > of when my service is available. If they have gone belly-up, well I
>> can then
>> > concentrate on some other company and consider my money as a cost
>> of a bad
>> > choice on my part.
>> >
>> > If I am a very rare case who just had a bad experience with an
>> excellent
>> > company ( I wish ), Nufone please fix this ASSAP.
>> >
>> > Later....
>> >
>> > Sathya
>> >
>> >
>> > _______________________________________________
>> > Asterisk-Users mailing list
>> > Asterisk-Users at 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
>> >
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
JB> =====
JB> Chris Albertson
JB> Home: 310-376-1029 chrisalbertson90278 at yahoo.com
JB> Cell: 310-990-7550
JB> Office: 310-336-5189 Christopher.J.Albertson at aero.org
JB> KG6OMK
JB> __________________________________
JB> Do you Yahoo!?
JB> Yahoo! SiteBuilder - Free web site building tool. Try it!
JB> http://webhosting.yahoo.com/ps/sb/
JB> _______________________________________________
JB> Asterisk-Users mailing list
JB> Asterisk-Users at lists.digium.com
JB> http://lists.digium.com/mailman/listinfo/asterisk-users
JB> To UNSUBSCRIBE or update options visit:
JB> http://lists.digium.com/mailman/listinfo/asterisk-users
JB> _______________________________________________
JB> Asterisk-Users mailing list
JB> Asterisk-Users at lists.digium.com
JB> http://lists.digium.com/mailman/listinfo/asterisk-users
JB> To UNSUBSCRIBE or update options visit:
JB> http://lists.digium.com/mailman/listinfo/asterisk-users
JB> _______________________________________________
JB> Asterisk-Users mailing list
JB> Asterisk-Users at lists.digium.com
JB> http://lists.digium.com/mailman/listinfo/asterisk-users
JB> To UNSUBSCRIBE or update options visit:
JB> http://lists.digium.com/mailman/listinfo/asterisk-users
John Good Luck with Voicepulse they are 0 for 5 this month so far i've
notice 5 outages this month alone and getting any email back from them
is next to impossible..
--
Best regards,
Frankie (fgravato at cfsdigital.com)
mailto:nanog at cfsdigital.com
--__--__--
Message: 11
From: "john" <jharragi at mw.k12.ny.us>
To: <asterisk-users at lists.digium.com>
Date: Mon, 26 Jan 2004 08:53:55 -0500
Subject: [Asterisk-Users] Questions regarding new echo cancellation
features...
Reply-To: asterisk-users at lists.digium.com
I notice the zaptel Makefile option
the mark2 option & KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
is now gone. Does simply adding these options still compile in a certain
echo can - or is there an other method of activating a particular can. I
have not had to update my machine that is connected to pstn for a while & I
don't want to jump into echotraining without a way to quickly enable what
has been working for me.
How do I enable this mode now?
How are people liking conversations with the echotraining enabled on both
ends of connections like...
remote* <> iax <> * pstn whatever
Thanks,
John
This e-mail was scanned and found clean by Monroe-Woodbury's Antivirus.
--__--__--
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