Re: [asterisk-users] 2 Asterisk Boxes, Single Voicemail
I tried NFS, but must be doing something wrong, as lag times between the two are unacceptably high -- as high as 10 to 15 seconds. If you have any hints about this problem, please let me know. Meantime, I'll pursue the rsync angle. Thanks, G -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Friday, February 05, 2010 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 2 Asterisk Boxes, Single Voicemail On 5 Feb 2010, at 16:55, Greg Blakely wrote: If so, how? NFS or rsync? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 Asterisk Boxes, Single Voicemail
Searching through the archives, I couldn't find an answer for this... I have two asterisk systems, (system A and system B), and would like to use a single voicemail system. Phones on system B are SIP phones, registered at system B. Can the message-waiting indicator be activated on a SIP phone registered to system B, if the voicemail resides on system A? If so, how? Thanks, folks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WARNING[3035]: Invalid priority/label ' ' at line 17
I just recently upgraded to the latest HEAD, and am now getting the following warning: -- Including context 'fromcnet' in context 'pots' Nov 12 18:45:17 WARNING[3035]: pbx_config.c:1697 pbx_load_module: Invalid priority/label '' at line 17 -- Including context 'longdistance' in context 'international' I have added a comment line above and below every config file that I have in /etc/asterisk, and the warning never changes. What's up with this? And will it affect anything? TIA ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] AEL Question
I've been puttering around with extensions.ael, and had a question. (Well, 2 questions, but they're related). First, would asterisk recognize any other .ael files as asterisk extension language? Second, is there a way to #include another file from extensions.ael like there is from extensions.conf? TIA Greg___ 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
[Asterisk-Users] txtcidname usage
I have questions about how to use txtcidname. Specifically, what DNS server does Asterisk use to do its TXT lookup? I cant seem to find a config file for this. Thanks. Greg ___ 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
RE: [Asterisk-Users] txtcidname usage
Errr never mind. In case anyone else was wondering, it uses the enum.conf file to find out who it should query. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Blakely Sent: Monday, August 22, 2005 5:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] txtcidname usage I have questions about how to use txtcidname. Specifically, what DNS server does Asterisk use to do its TXT lookup? I cant seem to find a config file for this. Thanks. Greg ___ 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
[Asterisk-Users] faxdetect config issues
My Asterisk fax detection used to work, but no longer does. OK. So, here's the deal: 1. It appears that the faxdetect command cannot be applied channel-by-channel in zapata.conf anymore, as Asterisk appears to the last faxdetect= command to ALL channels. 2. My stations are detected and sent to the proper extension; i.e., when I send a fax from one zap extension to a zap voice extension, it is intercepted and sent to my fax machine (which is on a SIP ATA). HOWEVER, my ZAP trunks are NOT detected. A call from an outside FAX machine goes to voice mail, and I get a message full of CNG tone. My questions are: 1. How can I make faxdetect apply on a per-channel basis again? (It USED to work that way) 2. How can I make my outside lines have CNG tone detected on them? Here is my config: In ZAPATA.CONF: ; A typical trunk Faxdetect=incoming ; have tried also both and outgoing context = fromqwest group= 9 channel = 1 ; ; A typical station signalling = fxo_ks musiconhold=default usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes group = 1 mailbox = 777 callerid = 777 context = internal channel = 7 ; In EXTENSIONS.CONF [fromqwest] exten = s,1,Answer exten = s,2,Wait(1) ;exten = s,3,Zapateller(answer|nocallerid) exten = s,3,NoOp ;exten = s,4,PrivacyManager exten = s,4,NoOp exten = s,5,Goto(internal,s,1) ; exten = fax,1,NoOp(Fax Detected) exten = fax,2,Dial(SIP/222-5000,20,tr) exten = fax,3,Congestion ; [internal] exten = s,1,Answer exten = s,2,Dial(ZAP/g1ZAP/10ZAP/11ZAP/17ZAP/38SIP/20,16,tr) exten = s,3,Goto(vm,s,1) exten = s,4,Hangup exten = s,103,Playtones(busy) exten = s,104,Wait(20) exten = s,105,Hangup ; ; exten = 10,1,Dial(ZAP/21r2,18,tr) ; exten = fax,1,Dial(SIP/222-5000,20,tr) exten = fax,2,Congestion exten = fax,102,Congestion ; ___ 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
RE: [Asterisk-Users] Vonage --- Asterisk Complete Config
Vonage doesn't sell just a softphone account -- or at least they didn't about six months ago when I was a Vonage customer. But they do allow a softphone as an add-on to an ATA-based account. Because the softphone account works with openly available soft clients, it also works with asterisk. The big secret is that they use port 5061, rather than port 5060. I thought Vonage did not allow this? -Randy Nitesh Divecha wrote: Hello Asterisk Users, After Brain storming for couple of hours, days, and weeks, finally got Asterisk to work with Vonage for Inbound and Outbound calls. Requirement: - 1) Vonage Softphone account 2) Asterisk 3) Couple of SIP Phones [snip] ___ 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
[Asterisk-Users] Traditional Ringback Tone
I am trying to get Asterisk to emulate the sounds of the earlier telephone systems, and the settings in [us-old] are pretty helpful. The only thing lacking is ringback tone, which is not quite as complex as the real phone systems of the day. For example, it is true that a ringback tone commonly used is 420Hz modulated by 40Hz. This is what shows up in [us-old]. But that modulated tone was generally overlaid on top of real ringing, i.e. 20Hz. So, using the Asterisk example of 420*40, it would seem that a decent ringback would be (420*40)*20. But, of course, that doesn't appear to exist. If it does, I am missing the boat on how to do it properly. So, I have a question: Is it possible to either (a) do the double modulation as listed above, or (b) provide recorded wav or gsm sounds as a background fill while a phone is being rung? I have recordings of various types of older central office ringback tones that I'd just love to be able to put into Asterisk. I know this sounds a bit arcane. But Asterisk can do so many things to order that it really ought to be able to do this, don't you think? Thanks, Greg ___ 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
RE: [Asterisk-Users] incoming calls produce multiple quarter rings andasterisk never answers.
Tip side open on the analog line? Have you taken a butt set or normal phone and attached it directly to the outside line to see if you get dial tone? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Gabrielson Sent: Friday, January 28, 2005 11:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] incoming calls produce multiple quarter rings andasterisk never answers. I have an adit 600 connected to a normal analog line. When I try to call that line, the phone rings a quarter ring(almost a beep) instead of a complete ring and keeps ringing and ringing with asterisk never picking up the call. Outgoing calls on those same lines aren't working either. Any suggestions on what might be wrong? Thanks, Jon. ___ 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 ___ 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
Re: [Asterisk-Users] can the dialtone be changed after pressing 9?
Alexander (or anyone),
Can you point me to where this can be done for zap
devices? zonedata.c, perhaps? How?
Thanks,
Greg
Yes you can but it only works for zap devices. IP based would be a
function of the hardware. extensions.conf
has ignorepat = 9 exten =
_9X.,1,Dial(Zap/G2/${EXTEN:1}) The first user to try it
asked if instead of keeping the same dialtone after pressing 9, if
I could play a different dialtone. Can this be done?
I'm running asterisk 1.0.0 in case that matters.___
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RE: [Asterisk-Users] UPS for Asterisk
Having worked in the telephone equipment business for years, I've found that there are those customers who want the cheapest possible solution -- a refurbished PBX running on the same circuit breaker that the rest of the stuff in the janitor's closet does. And there are those customers who see that the real cost savings is in having a reliable phone system. Those customers put the PBX into as controlled an environment as possible. At a bare minimum, they purchase a good-quality UPS; preferred would be an environment that would support a finicky main frame computer -- air conditioning, humidity control, etc. Businesses get what they pay for. But, if they use Asterisk, they can take the savings they have realized over buying a traditional PBX, buy a decent UPS, and still have a chunk of change left over. From: [EMAIL PROTECTED] on behalf of David Brodbeck Sent: Mon 1/24/2005 9:49 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] UPS for Asterisk -Original Message- From: Shoval Tomer [mailto:[EMAIL PROTECTED] On the other hand, telephony down time is unacceptable. PBXs have a counter part. Plain old PBXs are expected to run 24x7. real 24x7, with uptimes of 99.999. And if you think about it, they actually do. That would be news to the people who installed our (non-Asterisk) PBX. It has no battery backup at all. When the power goes out, so do all our phones. (Except for the fax machines, which don't go through the PBX.) ___ 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 winmail.dat___ 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
RE: [Asterisk-Users] can the dialtone be changed after pressing 9?
Kinda sorta works. The dialtone changes after dialing 9, but won't go away
when dialing the rest of the call.
The call actually DOES go out, though. I didn't stay on line long enough to
see if the dialtone would go away once answered.
You're on the right track, it would seem, and I have something to work with.
Thanks,
Greg
From: [EMAIL PROTECTED] on behalf of Steve Murphy
Sent: Mon 1/24/2005 12:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] can the dialtone be changed after pressing 9?
On Mon, 2005-01-24 at 11:01 -0600, asterisk-users-
[EMAIL PROTECTED] wrote:
Alexander (or anyone),
Can you point me to where this can be done for zap devices?
zonedata.c, perhaps? How?
Thanks,
Greg
Yes you can but it only works for zap devices. IP based would be a
function of the hardware.
extensions.conf has
ignorepat = 9
exten = _9X.,1,Dial(Zap/G2/${EXTEN:1})
The first user to try it asked if instead of keeping the same
dialtone
after pressing 9, if I could play a different dialtone. Can this
be
done? I'm running asterisk 1.0.0 in case that matters.
I've been thinking about this. The only logical way to change the
dialtone after dialing 9 is to do something like this, I think:
[dialcontext]
;; ignorepat = 9 no more
exten = 9,1,Goto(dialContext9,s,1)
...
[dialContext9]
exten = s,1,PlayTones(dial9)
..
exten = _X.,1,Dial(Zap/G2/${EXTEN})
exten = _X.,2,StopPlaytones
exten = i,1,StopPlaytones
exten = o,1,StopPlaytones
exten = t,1,StopPlaytones
where the dial9 tone is defined in the [us] (or whatever language you
are using) section of the
indications.conf file:
[us]
description = United States / North America
ringcadance = 2000,4000
dial = 350+440
busy = 480+620/500,0/500
ring = 440+480/2000,0/4000
congestion = 480+620/250,0/250
callwaiting = 440/300,0/1
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!
0/100,350+440
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0
dial9 = 337+463 ;; I have no idea how bad this will sound!
Or, you skip the config file stuff, and just use:
exten = s,1,PlayTones(337+463)
The only reason for using the config file, is to be better organized, by
having a single definition,
and possibly many references.
I really pulled the 337+463 out of the hat; my guess is, it should sound
pretty sour, hopefully be distinguished easily as a different dialtone.
I've not tried any of the above. It's just a guess. There may be all
sorts of interesting complications.
There may be more places to stick the StopPlaytones call. Who knows. It
may not work at all. All I can say is, if I wanted to do this myself,
this would be MY first attack... If anybody else has really done it,
it's time to share!
Good luck!
murf
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[Asterisk-Users] SayDigits -- ToneDigits??
I have a user who wants to receive an ANI spitback in DTMF. Right now,
the SayDigits(${CALLERIDNUM}) command works fine with voice. But I'd
like to end up doing both. Something along the lines of:
exten = 34,1,Answer
exten = 34,2,Wait(1)
exten = 34,3,Playback(vm-extension)
exten = 34,4,SayDigits(${CALLERIDNUM})
exten = 34,5,Wait(2)
exten = 34,6,Macro(DTMFDigits,${CALLERIDNUM})
exten = 34,7,Hangup
I've searched the voip-info tiki and google, but haven't seen anything
like this mentioned. Can anyone help?
Thanks,
Greg
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RE: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall
Works for me, too. But I found that the Benito Juarez
International airport was reachable by 9-011-52-5-571-3600.
To get this from my PBX-like setup, I have the following in
extensions.conf:
exten = _901152.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:6},90,Tt)
and the following in iax.conf
disallow=all
allow=GSM
allow=ULAW
allow=ALAW
allow=G726
allow=ILBC
allow=LPC10
allow=SPEEX
(Obviously, anything below allow=GSM isn't necessary for
this particular connection.)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Don Dawson
Sent: Thursday, January 13, 2005 9:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test
chan_unicall
I changed to line to :
exten =
_,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,Tt
and it works fine.
On 13/01/2005, at 9:22 AM, Gary Carr wrote:
I tried to call the mexico city airport and got the following
-- Executing Dial(SIP/9104044010-541d,
IAX2/[EMAIL PROTECTED]/57644910
@guest|90.Tf) in new stack
-- Called
[EMAIL PROTECTED]/57644910 @guest
Jan 13 10:20:59 WARNING[1142135600]:
chan_iax2.c:5339 socket_read:
Call rejected
by 200.53.121.233: No such context/extension
-- Hungup
'IAX2/200.53.121.233:4569/4'
== No one is available to answer at this time
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RE: [Asterisk-Users] RE: R2/MFC Mexico FREE calls to test chan_unicall
If I make it ${EXTEN:8), then it will strip off 90115255, and leave only
6 digits going to your switch. And, as you say below, you use eight
digits there.
Using ${EXTEN:6}, I was successfully able to call the Mexico City
Airport at
5571-3600.
So, I stick by my ${EXTEN:6} configuration, since it works.
;-)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Miguel Ruiz Velasco Sobrino
Sent: Thursday, January 13, 2005 11:48 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: R2/MFC Mexico FREE calls to test
chan_unicall
if you dial this to reach the airport (using international long
distance):
9-011-52-5-571-3600
in extensions.conf
exten = _90115255.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:8},90,Tt)
or
exten = _901152.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:8},90,Tt)
you should configure the extension in this way (adding the 55 in the
exten), AND (this
the important part!) the ${EXTEN:8}, because in mexico city the
numbers
are 8 digits
long, not 10, if you leave the 10 digit numbers you will call all sort
of
wrong places.
Message: 8
Date: Thu, 13 Jan 2005 11:05:30 -0600
From: Greg Blakely [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] R2/MFC Mexico FREE calls to test
chan_unicall
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Works for me, too. But I found that the Benito Juarez International
airport was reachable by 9-011-52-5-571-3600.
To get this from my PBX-like setup, I have the following in
extensions.conf:
exten = _901152.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:6},90,Tt
mailto:IAX2/[EMAIL PROTECTED]/$%7bEXTEN:6%7d,90,Tt )
and the following in iax.conf
disallow=all
allow=GSM
=
Miguel Ruiz Velasco
Version: OpenKeyServer v1.2
Comment: Extracted from belgium.keyserver.net
Signature: 0x59831109
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RE: [Asterisk-Users] Is Busydetect obsolete in the latest CVS?
You spelled detect wrong in your config file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garry Taylor Sent: Friday, November 26, 2004 10:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Is Busydetect obsolete in the latest CVS? Hi All, Does anyone know if Busydetect is obsolete in the current CVS? My zapata.conf is - ; ports 3 and 4 on the TDM22B signalling=fxs_ks busydectect=yes busycount=4 group=2 context=incoming channel = 3-4 When reloading asterisk I get the following messages - Nov 27 12:33:34 WARNING[6630]: chan_zap.c:9770 setup_zap: Ignoring signalling Nov 27 12:33:34 WARNING[6630]: chan_zap.c:9770 setup_zap: Ignoring busydectect -- Reconfigured channel 3, FXS Kewlstart signalling -- Reconfigured channel 4, FXS Kewlstart signalling why would it be ignoring the busydetect? Also, why is it sending a warning Ignoring signalling is singnalling=fxs_ks obsolete also? Regards Garry Taylor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] no dial tone when dialing out on vonage
Looks to me like you are telling Asterisk to outpulse 200 on the
vonage line.
If I remember my vonage service correctly, everything (except 911) was
an 11-digit call.
Perhaps it'd look better as:
exten = _1NXXNXX,1,Dial(Zap/2/${EXTEN})
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Imran Akbar
Sent: Thursday, September 02, 2004 7:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] no dial tone when dialing out on vonage
Hi,
I'm trying to dial out on a vonage line connected to a
zap channel using stuff like:
exten = 200,1,Dial(Zap/2/${EXTEN})
but it doesn't work - when i dial in the extension, i can see
on a phone connected to the same line that it's gone active -
but there's no dialtone. also tried adding a wait period
before accessing the line and exten =
_XX,1,Dial(Zap/2/${EXTEN})
to no avail.
what's goin on?
Thanks,
Imran
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RE: [Asterisk-Users] Hard Ground (On Ring)
If you have a single line phone, or better yet, a butt-set, hook it up to the TELCO side of the 66 block, and remove the bridge clips. This effectively isolates the customer equipment from the telco circuit. Then, using that butt set (or phone), go off hook and see if the static is still there. If so, it's telco's problem. If not, it's yours. Generally, if the service works and has static or hum on it, it's a TIP ground, since a RING ground would have the effect of seizing the circuit, thus making it completely unusable. Whatever the cause, though, if it's in telco's stuff, they have to fix it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Franks Sent: Thursday, September 02, 2004 1:27 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Hard Ground (On Ring) We have been experiencing static noise on one of our phone lines connected to our Adtran Total Access 750 which is then connected to a T100P card. I was convinced that, after rain storms, the static would get worse, it was surely a problem of Verizon's. Verizon sent a field rep out today and he told our secretary (I was busy) that there was a hard ground on the Ring portion of the line. Looking at the setup we have the NID connected directly to a 66 Punchdown block which then spans out to an amphenol 50 connector that connects to the Adtran. Could the Adtran be making a hardground anywhere? Has anyone seen/heard of this issue before? Unfortuantely, my background is computers and not electrical, so I am lost... Any help would be greatly appreciated, Brent ___ 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 ___ 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
RE: [Asterisk-Users] FXOs
I have an FXO card in a channel bank, which is run into a Digium TE405P. Works great. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Thursday, September 02, 2004 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXOs On Mon, 30 Aug 2004 17:15:42 -0400, Michael Graves [EMAIL PROTECTED] wrote: I'd really like to see a show of hands with regard to people's experience with FXO interfaces. I own a few X100p cards and have had nothing but problems with them. 2 X100 FXO here and no problems with them here in France. Caller ID, works and both are on lines that have DSL connections as well. ___ 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 ___ 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
RE: [Asterisk-Users] Revert to dial tone?
Thanks. That did the trick.
This is what I ended up with (on extension 45)
exten = 45,1,Dial(Zap/44r1,30,g)
exten = 45,2,System(test ${DIALSTATUS} = NOANSWER)
exten = 45,3,GotoIf($[${DIALSTATUS} = NOANSWER]?4:6)
exten = 45,4,voicemail(u10)
exten = 45,5,Hangup
exten = 45,6,DISA(no-password|internal)
exten = 45,7,NoOp
exten = 45,102,voicemail(b10)
exten = 45,103,Hangup
;
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Peter Svensson
Sent: Monday, August 30, 2004 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Revert to dial tone?
On Mon, 30 Aug 2004, Greg Blakely wrote:
Thanks. That appears to work, but it doesn't appear to work with
voicemail. From what I can see, the next priority can be taken up
either with the DISA command or the unavailable voicemail command.
Any way of separating the two?
Hm, I guess you want to do different things depending on the
reason for terminating the Dial command? I think there is a
variable DIALSTATUS
that you can test in the dialplan.
Peter
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[Asterisk-Users] Revert to dial tone?
I am wondering if it is possible for an extension that is served by a zaptel device to revert to dial tone once a call disconnects. For instance, if I make a call to another extension, talk with them, and THEY hang up, can I then be presented with a new dial tone rather than a congestion tone? Further, can an extension be set up so that, once the call goes back to dial tone, if the user does NOT dial any digits within a timeout period, + the PBX will return 30 seconds of congestion tone, and then + the PBX will return 60 seconds of howler tone, and then + the extension is 'locked out.' ? Thanks. ___ 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
RE: [Asterisk-Users] need help with zaptel.
Look in /usr/src. You should see a directory something similar to linux-2.6.1-1[as an example]. If you DON'T have a directory (or link to a directory) named linux-2.6, you should create one using the 'ln' command. In the case mentioned above, the command would be: ln -s /usr/src/linux-2.6.1-1 /usr/src/linux-2.6 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edward HuittSent: Saturday, August 21, 2004 5:42 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] need help with zaptel. I cant get zaptel to make. I get this error: make: *** [linux26] Error 1. The previous line is: Link /usr/src/linux-2.6 to your kernel sources first! I am running Fedora Core2 Asterisk compiles fine. I am using my SIP phones. I would like to get my TDM400p working.
[Asterisk-Users] AGI Script: calleridnamelookup.agi
Is anyone successfully using the AGI script calleridnamelookup.agi (or anything similar) ? I get both name and number caller ID from my POTS line, but I'd save money if I had them deliver ANI only. I've downloaded and installed the AGI script calleridnamelookup.agi, but I always get -- Executing AGI(SIP/9525485560-5359, calleridnamelookup.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/calleridnamelookup.agi -- AGI Script calleridnamelookup.agi completed, returning 0 I've even received that result calling in to my iconnect account, which delivers only ANI information. I notice that the URL that it queries does not respond when I enter it manually into a browser: http://www.anywho.com/qry/wp_rl/index.html?npa=719telephone=471. A box comes up that says Fetching Results, and then the request times out. Any idea how to structure the query on ANYWHO or how to use the script with another reverse lookup service? Thanks in advance. ___ 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
RE: [Asterisk-Users] External MW Lamp On/Off
Thank you. That will probably get me to where I need to go.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dunc
Sent: Wednesday, August 18, 2004 7:19 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] External MW Lamp On/Off
Greg,
Yes, it helps quite a bit. It shows me where Comedian
Mail spawns the
external app.
Do you have a copy of your SIP MWI script? I may be able
to use it as a
starting point.
FWIW, I've been using my extensions.conf to set/unset MWI on phones
attached to Cisco Call Manager - it's a bit of a hack but I couldn't
find anything better searching around. We've got CM4
interconnected to
* with a SIP trunk.
To change the MWI status I spoof the caller ID and send a
call from * to
the CM set or unset number, which doesn't sound so different
from what
you need to do other than it's a SIP call that changes the
MWI. I guess
you wouldn't need to worry about caller id as you'd be
dialling out on
an analogue line.
It's not very pretty, but it seems to work OK, the main
drawback is that
if a user retrieves their mail from someone else's phone the
light stays
lit. I think that's fixable, but at the moment it's not a
big deal for me.
Extract below - 100 is the voicemail entry point, and the 600/700
sequences in the h extension deal with figuring out what to do after
exit from voicemail.
Dunc
---
[globals]
VMAIL=0
[local]
; h - hangup
;
exten = h,1,GotoIf($[${VMAIL} != 0]?600)
exten = h,2,Hangup
;
; When exiting voicemail, check for new messages in the recipients
; mailbox and check that their MWI is set accordingly. [EMAIL PROTECTED]
; unsets MWI, [EMAIL PROTECTED] sets. Silly numbers that came about from
; getting the config togther. They need changing.
;
; This stuff actually needs to be in a context of its own, so that
; the h extension doesn't have to have the gotoif stuff. (maybe)
exten = h,600,SetCIDNum(${VMAIL})
exten = h,601,SetGlobalVar(VMAIL=0)
exten = h,602,HasNewVoicemail([EMAIL PROTECTED]:INBOX)
exten = h,603,Dial(SIP/[EMAIL PROTECTED])
exten = h,604,Hangup
exten = h,703,Dial(SIP/[EMAIL PROTECTED])
exten = h,704,Hangup
;
;
;
; Voicemail.
; First, check if the call is a redirection (ie someone
; being transferred in to leave a message) - CM redirects to
1+ccm ext
; to indicate that this is the case. Set $VMAIL to the destination
; mailbox for exit handling (ugly).
;
; If it's not a redirect, send to voicemail with the callerid as the
; mailbox, otherwise use the diversion field.
exten = 100,1,Wait(1)
exten = 100,2,GotoIf($[${RDNIS}:1]?9)
exten = 100,3,SetGlobalVar(VMAIL=${CALLERIDNUM})
exten = 100,4,VoicemailMain(${CALLERIDNUM})
exten = 100,5,Hangup
exten = 100,9,SetGlobalVar(VMAIL=${RDNIS:1})
exten = 100,10,Voicemail(u${RDNIS:1})
exten = 100,11,Hangup
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[Asterisk-Users] Internal Distinctive Ringing + Caller ID
I have set up my asterisk PBX to provide a double-ring for outside calls, and a single ring for station-to-station. (I'm talking about ZAP stations in this email). I had to go into one of the .c files and tell it to expect the Caller ID between the 2nd and 3rd rings in order to get the double-ring scenario to work. My problem is that, in making this change, I now don't see Caller ID on internal calls. Is there a work-around for this? It'd be really handy to have caller ID on both internal and external calls, AND to continue to have the distinctive ringing that I've been using. (I use the 'r5' option in my Dial(ZAP/) statement to get the double ring). ___ 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
RE: [Asterisk-Users] External MW Lamp On/Off
Yes, it helps quite a bit. It shows me where Comedian Mail spawns the external app. Do you have a copy of your SIP MWI script? I may be able to use it as a starting point. Also, can you tell me what variables are passed from asterisk to the app? Thank you very much. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Umar Sear Sent: Saturday, August 14, 2004 7:10 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] External MW Lamp On/Off I have done something simmillar, but not the same. I send mwi notification to our softswitch (SIP). Basically I wrote a small app in pascal that sends a sip message to the softswitch. The app is called everytime a message is left or retrieved, using the extrennotify option in voicemail.conf. You could easily do something simillar, what you need to do, is write a script or app (if one does not already exist) that creates call file based on the parameters passed by externnotify. Hope this helps. Umar --- Greg Blakely [EMAIL PROTECTED] wrote: One of the connections my asterisk PBX has is an analog extension from a Comdial hybrid. On the Comdial system, message waiting is turned on by dialing *3 and then the station number. It is turned off by dialing #3 and the station number. I was wanting to have Asterisk (or Comedian mail) set the message lamp in the Comdial system when a new message arrives for a user, and extinguish the lamp when the message has been played. I understand that this has something to do with a file that is placed in /var/spool/asterisk/outgoing, but I have no idea about + what the contents of that file should be, + how Comedian mail would initiate putting the file into the outgoing queue, and + how Comedian mail would initiate putting the 'extinguish' file into the outgoing queue. Has anyone done this sort of thing already? If so, can you point me in the right direction? As I mentioned in yesterday's post, I did find a question and partial answer to this in the asterisk-users archives, but I need a bit more information before I can make it work for me. Thanks in advance for any help you can give 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 ___ALL -NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com ___ 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 ___ 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
[Asterisk-Users] External MW Lamp On/Off
One of the connections my asterisk PBX has is an analog extension from a Comdial hybrid. On the Comdial system, message waiting is turned on by dialing *3 and then the station number. It is turned off by dialing #3 and the station number. I was wanting to have Asterisk (or Comedian mail) set the message lamp in the Comdial system when a new message arrives for a user, and extinguish the lamp when the message has been played. I understand that this has something to do with a file that is placed in /var/spool/asterisk/outgoing, but I have no idea about + what the contents of that file should be, + how Comedian mail would initiate putting the file into the outgoing queue, and + how Comedian mail would initiate putting the 'extinguish' file into the outgoing queue. Has anyone done this sort of thing already? If so, can you point me in the right direction? As I mentioned in yesterday's post, I did find a question and partial answer to this in the asterisk-users archives, but I need a bit more information before I can make it work for me. Thanks in advance for any help you can give 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
[Asterisk-Users] Message lamp integration with legacy pbx -- revisited
I see from the archives that Siggi Langauf was wanting to do exactly what I want to do back in November 2003. Here is what he asked: I would like to do a pilot with some legacy gear, however. Accordingly, I'd like to be able to have * dial 1000X where X is the box that has a new voicemail message and 1001X when the user of mb X deletes the new message(s). The dialing should occur within the default context. In each case, our legacy gear will turn on/off the message waiting lamp.For example, if I currently dial 1000400 on my * SIP phone, the MW lamp on legacy X 400 is flipped on by the PBX. If I dial 1001400 on my * SIP phone, the MW lamp on legacy X 400 is flipped off. He was told to take a look at /usr/src/asterisk/sample.call, and to put the results into /var/spool/asterisk/outgoing. Apparently, s/he was a little less dense than I am, because s/he never responded, leading me to believe that s/he figured it out. But I need a little help with the process. What would cause Comedian mail to actually use the sample.call file? Would Comedian mail know to use sample2.call to dial out to turn off the MW lamp once the caller picked up their new messages? If so, how? Bottom line, what commands go where? Thanks in advance. Greg
RE: [Asterisk-Users] iconnect inbound - FIXED (kinda)
This appears to have been the magic bullet for me. Thank you very much. So, the bottom line is that there is a bug that ends up making inbound calls use type=peer rather than type=user. Correct? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Cheng Sent: Tuesday, August 10, 2004 8:35 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iconnect inbound - so do we know how to fix it All, This is related to the following bug reports: http://bugs.digium.com/bug_view_page.php?bug_id=0002024 http://bugs.digium.com/bug_view_page.php?bug_id=0002017 This is not an iConnect specific problem, but a chan_sip change. As it turns out, type=user does not seem to work in the latest CVS (since June) for authentication against inbound--at least not in the way the documentation describes. What we have resorted to in our offices is have two type=peer contexts in sip.conf defined, the first being outbound, the second being inbound. Believe it or not, the order of these peers matters as chan_sip appears to take the LAST defined context as the authentication peer if you have a static host or dns name in the host= field. So, to summarize: 1) chan_sip.c has changed recently the authentication against type=peer and type=user 2) Registration statements that used to work now need to have a matching peer in sip.conf, however, the documentation states that chan_sip will first match type=user and if none is found type=peer. In real life testing with CVS-HEAD from August 4, this did not work. 3) What did work is creating an outbound peer with the authentication information such as username, secret, etc. (same as before) and a separate inbound peer with just the context,type=peer, host=, and any other codec preferences etc for the inbound leg. 4) This inbound peer has to be AFTER the outbound context otherwise, chan_sip will authenticate against the outbound peer instead of the inbound peer. 5) NOTE that the syntax of the registration statement has changed slightly as well (see wiki) and may need to be modified. On Aug 9, 2004, at 8:57 PM, Sathya Weerasooriya wrote: Raj, yes your post helped me. Just to complete the whole thing and clarify the problem that was posted by Greg Blakely; First, if there is no outbound iconnect section in sip.conf, my incoming calls work fine (as long as my register statements exist in the top section). But, when I add an outbound section, using either 'peer' or 'friend,' my incoming calls begin to fail again with the '407 Proxy Authentication' error. When there is a context created in SIP.CONF for iconnect outgoing, we should point it correctly to extensions.conf. Reason is now the incoming too land in this context. Thanks Sathya -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Raj Sent: Monday, August 09, 2004 5:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iconnect inbound - so do we know how to fix it May be you can find the solution in my post: http://lists.digium.com/pipermail/asterisk-users/2004-August/ 058014.html Raj --- Vladyslav [EMAIL PROTECTED] wrote: Try to comment out in your sip.conf ;qualify=yes On Mon, 2004-08-09 at 06:52, Sathya Weerasooriya wrote: Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW -- Best regards Vlad ___ 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 you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] iconnect inbound - so do we know how to fix it
I still have the problem, but have done a little further isolation. First, if there is no outbound iconnect section in sip.conf, my incoming calls work fine (as long as my "register" statements exist in the top section). But, when I add an outbound section, using either 'peer' or 'friend,' my incoming calls begin to fail again with the '407 Proxy Authentication' error. I've copied the section from /usr/src/asterisk/configs/sip.conf.example into my own sip.conf, and that makes no difference. Bottom line: I can have inbound or I can have outbound, but not both. One thing I've not tried is using the natrelay.deltathree.com for outbound, and sipauth.deltathree.com for inbound. Maybe that will 'fool' asterisk into thinking that they are two separate accounts. Obviously, I'm missing something here. But I've decided to lurk on the list, waiting for an answer -- with my outbound going on voicepulse, and inbound-only on iconnect. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sathya WeerasooriyaSent: Sunday, August 08, 2004 10:53 PMTo: [EMAIL PROTECTED] Digium. ComSubject: [Asterisk-Users] iconnect inbound - so do we know how to fix it Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW
RE: [Asterisk-Users] Voicepulse problems?
I'm not having the problem, but then I updated my IAX configuration per the instructions they sent out a couple of weeks ago, and that fixed a lot of the connectivity problems that I was having. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Wiesner Sent: Sunday, August 08, 2004 11:18 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voicepulse problems? Bruce, Yeah I'm having the same problems with VoicePulse. It's getting to be ridiculous because this happens all the time now. There's a new voip provider coming out that is working with some of the larger telcos. It will be offering similar quick turn up of services like voicepulse but much better service. In fact, there is even a phone number that you can call if you have problems where a person actually answers! AMAZING CONCEPT for a phone company! :-) Soon as it's up and out of beta I'm giving voicepulse the boot! ~Ken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito Sent: Sunday, August 08, 2004 10:29 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicepulse problems? Is any one else having problems with Voicepulse today? Suddenly, I can't register and calls to my Voicepulse numbers get a fast busy. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.734 / Virus Database: 488 - Release Date: 8/4/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.734 / Virus Database: 488 - Release Date: 8/4/2004 ___ 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 ___ 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
RE: [Asterisk-Users] Voicepulse problems?
In iax.conf
[vpconnect-t01]
type=peer
secret=jlz84532s
host=gwiaxt01.voicepulse.com
auth=md5
qualify=yes
[vpconnect-t02]
type=peer
secret=jlz84532s
host=gwiaxt02.voicepulse.com
auth=md5
qualify=yes
**
In extensions.conf:
[voicepulseld]
;
; Domestic Long Distance via VoicePulse
;
exten = _1NXXNXX,1,SetCallerID(2024561414)
exten = _1NXXNXX,2,SetCIDName(KERRY JOHN)
exten = _1NXXNXX,3,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,4,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,5,Goto(trunkld,${EXTEN},1)
exten = _1NXXNXX,6,Congestion
;
***
They don't have a Minnesota area code, so I don't use them for inbound
calls.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Totaro
Sent: Sunday, August 08, 2004 12:03 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Voicepulse problems?
can you post the instructions i think they went to my old
email and i dont see where to update email address on the wesite.
- Original Message -
From: Greg Blakely [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, August 08, 2004 12:41 PM
Subject: RE: [Asterisk-Users] Voicepulse problems?
I'm not having the problem, but then I updated my IAX
configuration per
the instructions they sent out a couple of weeks ago, and that fixed a
lot of the connectivity problems that I was having.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ken Wiesner
Sent: Sunday, August 08, 2004 11:18 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Voicepulse problems?
Bruce,
Yeah I'm having the same problems with VoicePulse. It's
getting to be ridiculous because this happens all the time now.
There's a new voip provider coming out that is working with
some of the larger telcos. It will be offering similar quick
turn up of services like voicepulse but much better service.
In fact, there is even a phone number that you can call if
you have problems where a person actually answers! AMAZING
CONCEPT for a phone company! :-) Soon as it's up and out of
beta I'm giving voicepulse the boot!
~Ken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bruce Komito
Sent: Sunday, August 08, 2004 10:29 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Voicepulse problems?
Is any one else having problems with Voicepulse today?
Suddenly, I can't register and calls to my Voicepulse numbers
get a fast busy.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
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RE: [Asterisk-Users] Vonage working with asterisk
Not officially. Several months ago, I talked with a 'rogue' tech support individual who told me that all I needed to get my Cisco 7940 working as a soft phone was to use port 5061 instead of port 5060. This translated well into asterisk, and I used it successfully for a couple of months. The problem is that is has to be a SOFT PHONE account, and Vonage does not sell JUST softphone accounts. They have to be attached to an existing account that uses their box. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Saturday, August 07, 2004 12:46 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Vonage working with asterisk did vonage finally allow there service to work with asterisk? when I was with them they wouldn't give out there server info. thanks hank - Original Message - ___ 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
RE: [Asterisk-Users] Inbound not working with iconnect
Voicepulse is working fine, as are all my IAX devices/connections. It's inbound SIP from iConnectHere that is the problem. And they appear to be the only company (other than vonage) that has local numbers in the Minneapolis / Saint Paul metro area. So, if they don't work, I'm left holding the bag, AFAIK. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis Sent: Saturday, August 07, 2004 10:53 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Inbound not working with iconnect Did you change the voicepulse context to user in you iax.conf? This is what they told me to do when I had the same problem 3 weeks ago: [voicepulse] type=user context=voicepulse-in ;auth=md5 ;secret=mysecret host=gw5.voicepulse.com qualify=yes This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raj Sent: Saturday, August 07, 2004 11:35 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Inbound not working with iconnect You are right Greg, I do get a 407 error when I sniff the packets and it's the same problem now. I'm able to receive calls from Broadvoice but not from iConnect. Any help would be appreciated in this regard. Thanks, Raj Greg Blakely [EMAIL PROTECTED] wrote:This may be what I experienced in my thread New Head Appears to break SIP to iConnect. Maybe it WASN'T the fact that I upgraded my asterisk software. But, yes. I noticed the problem day before yesterday. - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raj Sent: Friday, August 06, 2004 12:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Inbound not working with iconnect Hi there, Since last 2 days iconnect's incoming is not working. Is it the same with everybody? For the past 5 months I've been using this service perfectly in two boxes and suddenly it stopped functioning. I'm able to call out, the version is 0.9.1. Any help is appreciated Thanks, Raj - Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ 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 ___ 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 ___ 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
[Asterisk-Users] New Head Appears to Break SIP to iConnect
Folks, I have to admit that I MAY have changed something (at someone's advice) on a previous CVS head (May 28), but I'm not sure. I think that it had to do with changing digest realm, but that may be a different issue. At any rate, I had both incoming and outgoing with iConnectHere. Now, I made exactly ONE change: I upgraded to the CVS head dated 7/30. I still have outgoing SIP via iconnect, but the incoming just hangs, and finally times out to an iconnect intercept recording (Your Call Cannot be completed). I've done a 'sip debug,' and it appears that I've got the old 407 Error: (209.98.47.209 is me; 213.xxx.xxx.xxx is iconnect) lizzie*CLI Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.183 Via: SIP/2.0/UDP 213.137.73.41:5060;branch=22b27932-ad4142ba-5399fdb0-571ef11f-1 Via: SIP/2.0/UDP 213.137.65.239:5060;received=213.137.65.239 To: sip:[EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=31062DE8-21CE Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Record-Route: sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.183 Record-Route: sip:[EMAIL PROTECTED]:5060; maddr=213.137.73.41 Content-Type: application/sdp Content-Length: 148 v=0 o=CiscoSystemsSIP-GW-UserAgent 5851 2446 IN IP4 213.137.65.239 s=SIP Call c=IN IP4 213.137.65.239 t=0 0 m=audio 18698 RTP/AVP 4 0 8 2 101 13 headers, 6 lines Using latest request as basis request Sending to 213.137.73.140 : 5060 (non-NAT) Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Found RTP audio format 101 Peer RTP is at port 213.137.65.239:0 Capabilities: us - 0xc(ULAW|ALAW), peer - audio=0x1d(G723|ULAW|ALAW|G726)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer 'iconnect' Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.183 Via: SIP/2.0/UDP 213.137.73.41:5060;branch=22b27932-ad4142ba-5399fdb0-571ef11f-1 Via: SIP/2.0/UDP 213.137.65.239:5060;received=213.137.65.239 From: sip:[EMAIL PROTECTED];tag=31062DE8-21CE To: sip:[EMAIL PROTECTED];tag=as702accc9 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=60c6600e Content-Length: 0 to 213.137.73.140:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms lizzie*CLI Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.183 Via: SIP/2.0/UDP 213.137.73.41:5060;branch=22b27932-ad4142ba-5399fdb0-571ef11f-1 From: sip:[EMAIL PROTECTED];tag=31062DE8-21CE To: sip:[EMAIL PROTECTED];tag=as702accc9 Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Content-Length: 0 8 headers, 0 lines lizzie*CLI [EMAIL PROTECTED] asterisk]# ___ 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
RE: [Asterisk-Users] FCC Rules VoIP Must Be Tappable
Does the FCC honestly expect that criminals are going to stop using encrypted point-to-point VOIP connections just so that they won't be breaking the law? Yeah, right. I'm sure they'll all erase their encrypted IM clients so that the FCC will be happy. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang S. Rupprecht Sent: Wednesday, August 04, 2004 9:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FCC Rules VoIP Must Be Tappable Me raises his hand. All in favor of IAX with native encrypted tunneling say Aye :-) Now I'm likely in the target rings of Big Brother :-) If the voice data passed through a service provider run asterisk system, I'd imagine they'd just get a court order to force IAX encryption to be turned off. (Or try to pull some strings if the service provider was in a foreign country.) The question I have of this ruling is does this make end-to-End RTP encryption illegal? Ditto for re-invites that cut out all the middlemen? How are they planning in getting the two endpoints to stop encrypting things without tipping off the same two endpoints? What about VPN tunnels? Are they illegal now by the same logic? -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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 ___ 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
RE: [Asterisk-Users] 10:10am CST - VoicePulse appears to be down
Same here in Minnesota, USA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 30, 2004 10:46 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 10:10am CST - VoicePulse appears to be down On 30 Jun 2004 at 10:17, Michael Graves wrote: ...my VoicePulse Connect account is timing out on its login requests. Was working fine a hour ago. Michael I can confirm that this is the case from here in New Zealand too. Matt ___ 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 ___ 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
RE: [Asterisk-Users] Failover Trunking Won't Fail Over
Thank you. That did the trick. Once you execute Congestion, everything stops. You need something like exten = _NXX,1,Dial(SIP/) exten = _NXX,2,Dial(ZAP/26/...) exten = _NXX,3,Congestion Steve ___ 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
[Asterisk-Users] Failover Trunking Won't Fail Over
Hello, all.
In section 4.3.10 of the Asterisk Handbook, there is an example of an
LCR/Failover Trunking scenario. I've tried it, and it works, as long as
I fail over from something else to ZAP, but I can't get it to hunt to
the other context if the zapata channel (or group) is used first.
Can anyone help? Here is my extensions.conf, and the error message I
get.
[trunklocal]
;
; Local area dialing accessed through trunk interface
;
exten = _NXX,1,Dial(ZAP/26/952${EXTEN})
exten = _NXX,2,Congestion
;
exten = _612NXX,1,Dial(ZAP/26/${EXTEN})
exten = _612NXX,2,Congestion
;
exten = _651NXX,1,Dial(ZAP/26/${EXTEN})
exten = _651NXX,2,Congestion
;
exten = _763NXX,1,Dial(ZAP/26/${EXTEN})
exten = _763NXX,2,Congestion
;
exten = _952NXX,1,Dial(ZAP/26/${EXTEN})
exten = _952NXX,2,Congestion
[iconnectlocal]
;
; Local calls routed through iConnectHere
;
exten = _NXX,1,Macro(dialiconnect,1952${EXTEN},70)
exten = _NXX,2,Macro(fastbusy)
;
exten = _612NXX,1,Macro(dialiconnect,1${EXTEN},70)
exten = _612NXX,2,Macro(fastbusy)
;
exten = _651NXX,1,Macro(dialiconnect,1${EXTEN},70)
exten = _651NXX,2,Macro(fastbusy)
;
exten = _763NXX,1,Macro(dialiconnect,1${EXTEN},70)
exten = _763NXX,2,Macro(fastbusy)
;
exten = _952NXX,1,Macro(dialiconnect,1${EXTEN},70)
exten = _952NXX,2,Macro(fastbusy)
;
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
include = parkedcalls
include = trunklocal
include = iconnectlocal
[extensions]
;
include = local
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
[EMAIL PROTECTED] asterisk]# asterisk -vvr
== Parsing '/etc/asterisk/asterisk.conf': Found
Asterisk CVS-HEAD-05/28/04-09:54:40, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-HEAD-05/28/04-09:54:40 currently running on
lizzie (pid = 31313)
-- Starting simple switch on 'Zap/3-1'
-- Executing Dial(Zap/3-1, ZAP/26/763512) in new stack
-- Called 26/763512
-- Zap/26-1 answered Zap/3-1
-- Attempting native bridge of Zap/3-1 and Zap/26-1
=-=-=-=-=-=-=--=-=-=--=-=-=-=-=-=-=-=-=-=-=-=-=-
-- Starting simple switch on 'Zap/2-1'
-- Executing Dial(Zap/2-1, ZAP/26/763512) in new stack
Jun 21 23:00:23 NOTICE[1242952640]: app_dial.c:674 dial_exec: Unable to
create channel of type 'ZAP'
== Everyone is busy at this time
-- Executing Congestion(Zap/2-1, ) in new stack
lizzie*CLI
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[Asterisk-Users] Hard Coded CLASS Codes (was 11 instead of Star)
In May, I posted an inquiry to the list concerning my desire to configure my own CLASS codes in extensions.conf rather than having them hard coded into the channel drivers. I have a number of old rotary dial telephones that (obviously) can't dial *. Traditionally in the US, 11 can be dialed in place of * as the first digit dialed. Many people mentioned that this would be very useful to them, especially those whose Asterisk PBXes are not located in the US, where their countrys' CLASS codes are different than the US standard. I guess I have to ask: If I want this change, as per the bug at http://bugs.digium.com/bug_view_page.php?bug_id=071 , What is the procedure? Do I just ask nicely and hope for the best? Or would funding be required to pay for having the code rewritten? If payment is necessary, how many of you members on the list would be willing to pitch in your own money to fund the effort? Thanks! ___ 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
RE: [Asterisk-Users] FWD network from Asterisk through NAT
I use the new IAX service at FWD. Much easier than trying to sort out the whole proxy thing with SIP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank smith Sent: Saturday, June 05, 2004 8:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FWD network from Asterisk through NAT Hi there, I'm trying to dial into the FWD network using Asterisk, though a NAT. The sources I've read say that it's unconfirmed to work through a NAT, but I'm wondering if anyone's done it anyway. So, anyone got a clue how to do this? Hank ___ 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 ___ 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
RE: [Asterisk-Users] * to Vonage Connection anyone?
For incoming, it's a simple entry in sip.conf: [general] register = 16126051544:[EMAIL PROTECTED]:5061/200 ; This will register username 16126051544 with password of QjrT56svW to server atlas3.atlas.vonage.net on port 5061. Incoming calls will ring to extension 200, as defined in extensions.conf ; Outgoing is a little trickier. I've had better luck with SIP using iConnectHere. And IAX providers make the easiest of all outgoing connections. (I use voicepulse). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Saturday, June 05, 2004 7:56 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] * to Vonage Connection anyone? Does anyone have any configuration info for the Vonage sip client? -Original Message- From: Greg Blakely [mailto:[EMAIL PROTECTED] Sent: Friday, June 04, 2004 11:59 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] * to Vonage Connection anyone? If you use their soft phone, it will work with Asterisk if you use port 5061 rather than port 5060. Incoming works well all the time; outgoing is somewhat problematic, especially if you are using Asterisk to proxy for one of your internal SIP phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Roy Sent: Friday, June 04, 2004 10:09 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] * to Vonage Connection anyone? Listonians, Anyone get * to work together with Vonage? Thanks, Jerry ___ 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 ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] * to Vonage Connection anyone?
If you use their soft phone, it will work with Asterisk if you use port 5061 rather than port 5060. Incoming works well all the time; outgoing is somewhat problematic, especially if you are using Asterisk to proxy for one of your internal SIP phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Roy Sent: Friday, June 04, 2004 10:09 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] * to Vonage Connection anyone? Listonians, Anyone get * to work together with Vonage? Thanks, Jerry ___ 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 ___ 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
[Asterisk-Users] CFDA from cell phone to SIP line in Asterisk PBX
I have something I'd like to try, but don't know if enough information would be delivered to Asterisk to make it work. I have a cell phone that I would like to forward on Don't Answer to an iConnect phone number that is then delivered SIP to my Asterisk PBX. I'd like to be able to route the call when it is received by Asterisk based on the number that the caller dialed, ie, the cell phone number. So, my question is, of all the ANIs delivered to Asterisk by a SIP call, would the number orignally dialed be included in the mix? If so, what variable (if any) does Asterisk assign to it? TIA -- Greg -- ___ 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
RE: [Asterisk-Users] Asterisk Receptionist manager program.
I had a similar result. The buttons work fine for transferring calls, but there was no dial pad shown. (Is there supposed to be?) Also, it would be VERY handy if it didn't have to take up the whole screen. I've taken to clicking on the icon in the upper left corner and choosing restore just so that I don't end up having to devote an entire workstation to nothing but Asterisk Receptionist. From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED] Sent: Thu 6/3/2004 1:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Receptionist manager program. It didn't work for me, didn't show me the keypad and my extention (IAX extention), below is a copy of the debug window. ---start--- Asterisk Call Manager/1.0 ---stop--- ---start--- Response: Success ---stop--- ---start--- Message: Authentication accepted ---stop--- ---start--- Response: Error ---stop--- ---start--- ActionID: 1 ---stop--- ---start--- Message: Permission denied ---stop--- --__--__-- Message: 2 Date: Thu, 03 Jun 2004 09:27:44 -0700 From: Kyle Hagan [EMAIL PROTECTED] Organization: Nuvo Technologies To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Receptionist manager program. Reply-To: [EMAIL PROTECTED] I put a new version up last night. Caller ID shows up on the buttons. This time IAX is fixed. Works at home and at work through FWD. http://www.easyhomenetworks.com/AstRec/ Has anyone had anyother bugs popup other than the IAX problem? Some people are asking why the screen shot has more buttons than the alpha version. We are going to get the bugs worked out of the existing buttons before we add more features. Kyle ___ 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 winmail.dat
[Asterisk-Users] Voicetronix OpenLine4 -- Help Needed
Hi. I need help with my brand new Voicetronix OpenLine4 board that I
installed into Asterisk.
After building the Linux device driver and inserting the module, I
modified the /usr/src/asterisk/channels/chan_vpb.c file to uncomment the
US settings and comment out the Austrailian ones.
I made the appropriate entries for routing in vpb.conf and
extensions.conf All appears to be well, except that the volume is
VERY low.
On incoming calls, I can barely hear the call being answered by voice
mail.
On outgoing calls, only some of the digits are being heard by the CO,
with the call eventually timing out.
I've tried inserting w in front of the ${EXTEN} in extensions.conf,
but that doesn't do anything.
I've tried dialtone instead of fxo for the mode in vpb.conf, but
that makes it unable to dial out at all.
Can anyone help?
(Config files posted below)
; - vpb.conf
[general]
type = v4pci
cards = 1
[interfaces]
board = 1
echocancel = on
txgain = 12
txhwgain=12
txswgain=12
;
; For OpenLine4 cards
context = fromvpb
mode = fxo
channel = 1
channel = 2
channel = 3
channel = 4
; ===
; in extensions.conf
exten = _NXX,1,Dial(vpb/1-1/${EXTEN})
(which should be correct, since I don't have a leading 9 or 8 in my
dial plan. All I'm trying to do here is send 7-digit calls out over a
loop-start CO line.)
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RE: [Asterisk-Users] Voicetronix OpenLine4 -- Help Needed
I had been using the HEAD branch, but went back to the STABLE version
just so I could use the drivers that I downloaded from Voicetronix.
Curiously, a majority of my issues were solved when I reversed the
polarity on the incoming line.
But I am now having an error appears indicating that it can't read the
FSK caller ID signal. I have verified that there is indeed a caller ID
squawk between the first and second rings, and I have told Asterisk to
wait 6 seconds before answering. Additionally, I have callerid = on
as a statement in vpb.conf.
Here is the error:
ERROR[1175660480]: chan_vpb.c:437 void get_callerid(vpb_pvt*): Failed to
decode caller id on vpb/1-1 - VPBAPI_CID_NO_SIGNAL
Any ideas?
In case you haven't, did you download the latest drivers for
OpenLine4 from Voicetronix. We used their latest drivers and
patches for chan_vpb.c and have elinimated the DTMF being
dialed out too early and also improved sound quality when
multiple lines are in use.
David
- Original Message -
From: Greg Blakely [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, May 27, 2004 2:14 PM
Subject: [Asterisk-Users] Voicetronix OpenLine4 -- Help Needed
Hi. I need help with my brand new Voicetronix OpenLine4 board that I
installed into Asterisk.
After building the Linux device driver and inserting the module, I
modified the /usr/src/asterisk/channels/chan_vpb.c file to
uncomment the
US settings and comment out the Austrailian ones.
I made the appropriate entries for routing in vpb.conf and
extensions.conf All appears to be well, except that the
volume is
VERY low.
On incoming calls, I can barely hear the call being answered by voice
mail.
On outgoing calls, only some of the digits are being heard by the CO,
with the call eventually timing out.
I've tried inserting w in front of the ${EXTEN} in extensions.conf,
but that doesn't do anything.
I've tried dialtone instead of fxo for the mode in vpb.conf, but
that makes it unable to dial out at all.
Can anyone help?
(Config files posted below)
; - vpb.conf
[general]
type = v4pci
cards = 1
[interfaces]
board = 1
echocancel = on
txgain = 12
txhwgain=12
txswgain=12
;
; For OpenLine4 cards
context = fromvpb
mode = fxo
channel = 1
channel = 2
channel = 3
channel = 4
; ===
; in extensions.conf
exten = _NXX,1,Dial(vpb/1-1/${EXTEN})
(which should be correct, since I don't have a leading 9 or
8 in my
dial plan. All I'm trying to do here is send 7-digit calls out over a
loop-start CO line.)
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RE: [Asterisk-Users] 11 instead of Star
OK. Well, here are a couple of newbie-type thoughts on the whole Vertical Service Code (CLASS) hard-codings. + First: let's get some diffs done to allow either 11 or * to be used as the leading digit(s). This is industry standard. So, if we're hard-coding these assignments in an effort to be CLASS compliant, let's be fully compliant. I am NOT a coder, so anything I do will be really ugly, but I will do SOMETHING just to arouse horror and guffaws, and to create a starting point. + Second: *8 just ain't gonna work for call pickup unless we want to blow away the usefulness of any *8X CLASS codes, unless (as is probably true) I am missing something. So, it would seem that a different code would be desirable -- maybe one of the locally-assignable *94 - *99? + It's just as well that *8# isn't used for call pickup anymore. The # on the end really SHOULD mean end of dialing, and not have any other significance. Or has this already been discussed to death? ___ 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
RE: [Asterisk-Users] 11 instead of Star
[...] I think the way it was going to go was a flag which would allow you to disable all channel driver features like this and rely on the dial plan to implement the features. This is very much my preferred solution. If there is still some bizarre obligation to support alien phone standards in the channel drivers, we should have the option of disabling this undesired behaviour. Or... What about having something similar to the tone plans in indications.conf that would allow someone to either choose one of several canned Vertical Service Code plans or roll their own? ___ 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
[Asterisk-Users] 11 instead of Star
I have several older telephones with rotary dials that I would like to use a working museum pieces. I have everything working well except for those hard-coded codes that start with *. In the traditional phone world, dialing 11 in place of * works fine, ie, someone could dial 1172 in place of *72 and so forth. I'm thinking that a simple entry in extensions.conf ought to do the trick, but I get nowhere using various configurations. One such is: exten = 1172,Goto(default,*72,1) But, obviously, that didn't do the trick. A quick search of the archives didn't turn up anything obvious. So, I have to ask: Does anyone know if (and how) this can be done? Thank you. ___ 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
RE: [Asterisk-Users] 11 instead of Star
Hmmm... Well, I did a search and replace on chan_zap.c, and got most of it converted to 11XX instead of *XX, but the call pickup code still eludes me. Is it set somewhere else? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick Sent: Monday, May 24, 2004 9:58 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 11 instead of Star I'm thinking that a simple entry in extensions.conf ought to do the trick, but I get nowhere using various configurations. One such is: exten = 1172,Goto(default,*72,1) But, obviously, that didn't do the trick. *sighs* Yeah, that won't work.. which is a shame.. this goes back to the whole debate of should CLASS service codes be implemented in the dial plan or the channel driver? From memory and reading the mailing list for a while now, I think Mark's dead against having these features in the dial plan, but I can't remember why. All the CLASS features are currently hard coded in the Zap channel driver so you could probably hack that around a bit to solve your problem, and a patch to implement 1172 = *72 etc for all the codes would probably be gratefully received in to CVS for the purposes of making things nice and complete. I think the way it was going to go was a flag which would allow you to disable all channel driver features like this and rely on the dial plan to implement the features. Searching back through the mailing list will shed some light on the pros, cons and the ongoing debate, but I think for now you're going to have to get involved with changing channel driver source to do what you want to do. Cheers Paul ___ 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 ___ 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
RE: [Asterisk-Users] RE: Caller ID
I've come late into this thread, so I risk saying things that you all will just shake your heads at and say, Duh! Historically, though, from WAY back in the days of electromechanical switches, reverse battery mainly provided answer supervision. Its usefulness pretty much went away with the advent of SS7, except for those cases where end users resold their POTS services (such as hotels and motels, which usually paid extra for the service). The battery would then reverse BACK to normal again after the call was terminated. During this reversal (obviously), the voltage would transition past zero, and it would also suffice for disconnect supervision. Aside from the hotel/motel scenario, telcos have recently been providing disconnect supervision solely by means of removal of battery from the circuit. This feature continues to be of value in situations where analog CPE would continue to keep the line seized were it not for the removal of the battery -- key systems having lines on hold, answering machines, etc. Personally, I would very much like to see the reverse battery feature built in to the FXS cards that work on asterisk. I say this because I am starting to go back to my roots in the industry by looking for old step-by-step line finders, selectors, and connectors. Answer supervision via some electromechanical means would be preferable than trying to cobble an ISDN D channel over to that old stuff. Just my 2 cents. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: Sunday, May 02, 2004 1:05 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: Caller ID Steve Underwood [EMAIL PROTECTED] wrote: Wake up. Sorry, I must have drifted off for a while. Thanks for the alarm call. The reversal detection is a complete waste of time. Totally unnecessary. Pointless. A line break detector would have much more use, as it would give a reliable disconnect detection on many lines. (Actually, reversal detection would have years ago, but its not much use any more). Perhaps both would be good then: A polarity reversal detector for determining the start of a Caller*ID sequence and a line break detector for, err, detecting line breaks. Actually, my X101P seems to detect hangups just fine, so I've not had cause to check whether the detection is done in the hardware or in the driver. If you say that it's not done in hardware then I'll take your word for it for the moment. All you need for these CLI requirements is to monitor for some energy on the line. Since these FXOs are not being used in banks of hundreds, you will never notice this MIPs this uses. I'd still prefer to see this done in hardware, rather than in some sort of idle loop in the driver or the application. Call me old fashioned, but I prefer it when unnecessary overheads are not measured in MIPs. :-) Perhaps the new FXO module for the TDMxxB has, or will have, hardware support for the above. If it has, and I'm sure I heard somewhere that it does, then that's great. An X102P, with similar support, would no doubt be welcome too. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 ___ 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
