[Asterisk-Users] Problems with Zpateller on incoming external calls
Title: Problems with Zpateller on incoming external calls I've setup the following in extensions.con: exten = 2200,1,Ringing exten = 2200,2,Wait(2) exten = 2200,3,Answer exten = 2200,4,Zapateller exten = 2200,5,Macro(stdexten,2205,SIP/2205) This works as expected if I dial from a SIP phone on my desk. However, if I dial in from the PSTN (through a SIP provider) it fails while trying to play ths SIT with: Apr 8 18:53:12 WARNING[1209269552]: rtp.c:407 ast_rtp_read: RTP Read error: Resource temporarily unavailable Any idea what's going on? My suspicion is that the PSTN gateway hasn't setup an audio path yet, although I thought Answer would do that. Cheers, Brian
[Asterisk-Users] IAX2 Trunk to PSTN (voicepulse) questions...
All,
I've almost got my Asterisk PBX setup, but I've having some problems with
the VoicePulse IAX trunk.
On outbound calls, when dialing a PSTN number through the IAX2 trunk,
music on hold (moh, using the m option in the dial command) does not work.
The console states that stop sound on IAX2 channel. Ring works, but
only without the r option. MOH works when trying to dial a non-PSTN
terminated IAX2 calls (e.g. a softphone). I've read that with SIP
connetions, the originating line is not held open by the PBX, so the can
be no timing sync with the client, but I don't know if that's also the
case here.
The setup I have is:
[sip softphone Xten] == [ * ] == [IAX2 VoicePulse Trunk] = [PSTN Number
(SprintPCS Cell)]
The relevant iax.conf sections are:
[voicepulse]
context=voicepulse-incoming
dtmfmode=rfc2833
secret=mysecret
auth=md5
type=user
host=gw5.voicepulse.com
[voicepulse-peer]
qualify=yes
trunk=yes
dtmfmode=rfc2833
secret=mysecret
auth=md5
type=peer
host=gw5.voicepulse.com
My extensions.conf has:
TRUNK=IAX2/[EMAIL PROTECTED]
exten = 15,1,Playback(transfer)
exten = 15,2,Dial(IAX2/ckm,20,rt)
exten = 15,3,VoiceMail(u${EXTEN})
exten = 15,4,Hangup
exten = 15,103,Dial(${TRUNK}/1411212,30,t)
exten = 15,104,VoiceMail(u${EXTEN})
exten = 15,105,Hangup
Any ideas, bug?
Thx.
Chris.
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[Asterisk-Users] Who has access numbers in the UK and Germany?
Hello, I need a few access numbers in the UK and Germany. Does anyone have this available right now? I need the incoming calls to be directed through IP to one of my asterisk servers in Europe. Please contact me off the list if you want. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ___ 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] AGI - GET DATA not working on current stable cvs (anyone else?)
Has anyone else had trouble with the AGI command GET DATA on the latest stable cvs? I can't get it to work with asterisk-perl, or by using print statements and reading stdin. I get 200 result= (timeout). (this is from the print statements, and asterisk-perl reports nothing). But asterisk is getting DTMF because my menu in extensions.conf works. I will go through the code Friday, but I just didn't know if anyone else was seeing this? Setup: 23 voice pri from Avaya PBX to Asterisk IVR. Thanks, Jeb Campbell ___ 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] app_queue dialback cdr problem
Hi all, We've been experimenting with the app_queue application, and it works quite well. The only problem we encountered was that outgoing calls (to the operators) aren't logged in CDR. Example, * operators dial a specific number/extension, and AddQueueMember(..) runs (they get added without any problems), and they Hangup. * normal users dial the support/hotline number, get added to the queue, and the app_queue starts dialing to all the available numbers/members in that queue until one picks up The problem (or feature) is that when the application start to dial to all the available members, if one member picks up the phone, those details (nr of operator, calltime, ..) don't get recorded in cdr. Is this intentionally ? Or a bug/feature request that has to be added to the bugtracker ? 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
[Asterisk-Users] Réf. : [Asterisk-Users] RE: [Asterisk-Users] Réf. : Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI
Finally, i will get back to a RedHat 9 distrib as I see that it works with that distribution...[EMAIL PROTECTED])fjåËbú?jË^®+$ºÇ«±:5%H$HJ+ºZµê)¶*'²ø¬Øm¶ÿ+-±Ø é¢oæj)fjåËbú?jË^®+$ºÇ«
[Asterisk-Users] application Directory (Modified by Ryan Thrash)
Sent 12 hours ago and it never showed up (slightly reworded here). Sorry if this is a duplicate: - Scenario: a person selects an Auto Attendant option that fires off the Directory application (CVS circa 3/22). Three questions: 1) How do they escape if they didn't mean to go there in the first place (without having to hang up...)? Config of entry into the vertex directory below: exten = 1,1,Directory(vertex) exten = 1,2,Goto(s,200) 2) Why is there a five second pause before the directory instructions start? 3) Why no option for first name (without recording your own custom message and reversing names in voicemail.conf)? Thanks, Ryan ___ 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 access to voicemail
Use another DID for just voicemial so all users can call into it, enter their extension and then then password to access their own voicemail. I just created this today for a Production system. The extensions.conf looks like this for anyone who call 963-4400: exten = 4400,1,Voicemailmain() exten = 4400,2,Hangup. Upon entering voicemail, the user will be prompted to enter their mailbox number and then their password. -Original Message- From: Keith D'Atrio [mailto:[EMAIL PROTECTED] On Behalf Of Keith D'Atrio Sent: Thursday, April 08, 2004 2:47 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] External access to voicemail Instead of using playback for your vm sounds try the Background command. This command allows interruption by hitting a key. Keith _ From: [EMAIL PROTECTED] on behalf of Steven Kokinos Sent: Thu 4/8/2004 02:41 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] External access to voicemail in my setup i have several users with DID lines coming in from various sip/iax providers. within our old phone system, a user could call their own DID line, then hit the * key when they hear their voicemail greeting and be prompted for their password. is there any way this could be replicated within asterisk? i'm having trouble figuring it out since it steps through things sequentially, whereas i want to scan for input during the playback. any help would be greatly appreciated. regards, -steve --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.645 / Virus Database: 413 - Release Date: 3/28/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.645 / Virus Database: 413 - Release Date: 3/28/2004 attachment: winmail.dat
RE: [Asterisk-Users] Auto Attendant??
You guys are making this way harder than it needs to be. Assume your
main number comes in on 4400, you want to give the receptionist an
opportunity to answer the call, but if s/he's away from the desk or on
another call you want to proceed to an auto-attendant to direct the call
as necessary. In the extensions.conf you will enter the following:
[default]
exten = 4400,1,Dial(SIP/4401,15,r) ; Ring the Receptionist for 15
seconds
exten = 4400,2,GoTo(MainMenu,s,1) ; If the Receptionist doesn't answer
goto the main menu
exten = 4400,102,GoTo(MainMenu,s,1) ; If the Receptionist is on the
phone goto the main menu
[MainMenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,Background(welcome) ; Play Welcome to companyname greeting
exten = s,5,Background(select) ; Play options menu that says select 1
for directory, 2 for sales, etc.
; Provide a directory of users
exten = 1,1,Directory(default) ; Users are listed in the default
context of voicemail.conf
; Go to the sales department
exten = 2,1,Dial(SIP/4402$SIP/4403,20,r) ; Ring the sales department
; Leave a voicemail for the sales department
exten = 2,2,Voicemail(sales) ; If no answer leave a message for the
sales department
; etc.
* End extensions.conf *
You can create a simple extension to record the menu prompts in
extensions.conf like this:
exten = 205,1,Wait(2)
exten = 205,2,Record(/tmp/asterisk-recording:gsm)
exten = 205,3,Wait(2)
exten = 205,4,Playback(/tmp/asterisk-recording)
exten = 205,5,Wait(2)
exten = 205,6,Hangup
When you dial extension 205, you will hear a 'beep', after which you
should begin recording; press the # key when finished. After each
recording is recorded, move it (/tmp/asterisk-recording.gsm) to
/var/lib/asterisk/sounds/filename.gsm so you can find it and access it
via the 'Background(filename)' command issued earlier.
I just did this today to provide a customized menu system for a
Financial Consultant so that users could contact the Consultant, his
assistant, or leave a voicemail for each if they were busy or
unavailable. You can provide a custom menu for each extension by
placing each in its own [context] and jumping to them with the GoTo
command.
Good Luck, and have fun!
Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Sent: Thursday, April 08, 2004 3:38 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Auto Attendant??
If you are refering to the Login Logout of Auto Attendant you can find
an example in the wiki...
But here is an my example of what you will find in the wiki
;Auto Attendant Login Out
exten = *801,1,DBPut(auto/attendant=1)
exten = *801,2,Hangup
exten = *802,1,DBPut(auto/attendant=0)
exten = *802,2,Hangup
;Incoming calls- check if autoattendant is logged in, otherwise goto
main exten = s,1,DBGet(autoattendant=auto/attendant)
exten = s,2,GotoIf($[${autoattendant} = 1]?3:4)
exten = s,3,Dial(SIP/recep,30,t)
exten = s,4,Goto(main,s,1)
[main]
exten = s,1,Answer
exten = s,2,Background(ctm-main-thanks)
exten = 1,1,Goto(default-ctm,3001,1)
exten = 2,1,Goto(default-ctm,3002,1)
exten = 0,1,Goto(default-pb,2002,1)
exten = 3,1,Hangup
Hope this helps
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Moran
Posted At: Thursday, April 08, 2004 1:48 PM Posted To: Asterisk User
Group
Conversation: [Asterisk-Users] Auto Attendant??
Subject: [Asterisk-Users] Auto Attendant??
I'm having trouble finding documentation for the auto attendant does
anyone have an idea where there might be some???
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Re: [Asterisk-Users] Two operators, 10 rollover lines, Cisco 7960G chanisavail problem
Hi Christian,
On Thu, Apr 08, 2004 at 06:13:50PM -0500, Christian Hoffmeyer wrote:
Here's my situation.
I have two receptionists that answer incoming lines. Each has a 7960G with
5 incoming lines each. I'm trying to set this up so each line on each phone
doesn't utilize call waiting. My problem seems to be that
ChanisAvail(Sip/cisco1Sip/cisco2Sip/cisco3Sip/cisco4Sip/cisco5) always
returns cisco1.
I had tried this back in October and found that ChanisAvail does not work for Sip
channels. I think you can look at the code and confirm this. I have not heard of it
being implemented.
The incominglimit=1 is the solution I used. If you use this feature, it will only
allow one call to the extension. Therefore, you can set your Dial command to ring all
of the extensions at the two front desks (set a global) and it will ring all of the
extensions except the ones that are busy.
Actually, I just set all of the lines on my 7960 phones to the same Sip extension.
That way if they are on a call on the first 7960 appearance, they can just press the
next button to answer the new call while the 7960 holds the one they were talking on
(while playing music on hold!).
email me offline if you want me to call you and discuss it with you more. Also, I
could send my config files for you to take a look at.
Walker
Here are the sip.conf entries: (mind you, there are entries for
frontdesk1-10 and each phone logs in with 1-5 and 6-10 respectively)
[frontdesk10]
type=friend ;Theresa Sprocket
username=frontdesk10
callerid=Cogswell's Coggs 555
secret=asterisk
host=dynamic
transfer=yes
canreinvite=no
incominglimit=1
context=recordings
Here's the dial string I'm trying to use:
exten = 775,1,SetMusicOnHold(default)
exten =
775,2,ChanisAvail(Sip/frontdesk1Sip/frontdesk2Sip/frontdesk3Sip/frontdesk
4Sip/frontdesk5)
exten = 775,3,Cut(DESK1=AVAILCHAN||1)
exten =
775,4,ChanisAvail(Sip/frontdesk6Sip/frontdesk7Sip/frontdesk8Sip/frontdesk
9Sip/frontdesk10)
exten = 775,5,Cut(DESK2=AVAILCHAN||1)
exten = 775,6,Dial(${DESK1}${DESK2},15,tr)
Any tips, tricks or ideas would be greatly appreciated.
Just do this:
FRONTDESK=Sip/frontdesk1Sip/frontdesk2Sip/frontdesk3Sip/frontdesk4Sip/frontdesk5
...
exten = 775,1,Dial(${FRONTDESK})
Thank you,
Christian Hoffmeyer
YottaDot Solutions
Huntsville, AL
(iax) 700.859.4508
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DataCrest, Inc.e-mail: [EMAIL PROTECTED]
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Birmingham, AL 35216 fax: 1-205-823-7838
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Re: [Asterisk-Users] Who has access numbers in the UK and Germany?
I need a few access numbers in the UK and Germany. Does anyone have this available right now? I need the incoming calls to be directed We do. I'll mail you off-list. Linus Magrathea ___ 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] Zaptel/PRI problem
Dear Willy i notice the same problem with my E100P using the latest cvs zaptel driver i have try every type of config in /etc/zaptel.conf to check if i have missed something in timing conf but nothing... Digium help... :-) thanks in advance Dimitri On Thursday 08 April 2004 23:07, [EMAIL PROTECTED] wrote: Chris, Thank you for posting this. Since it concerns my 'production' system, let me comment. After 'downshifting' to a previous release (for no good reason other than desperation and teh fact that an earlier list entry had commented that it cleared up the problems) I am sad to report that the system failed again. Miscellaneous throughout the day: Apr 8 13:41:27 WARNING[-1210639440]: PRI: Read on 32 failed: Unknown error 500 Apr 8 13:41:27 NOTICE[-1210639440]: PRI got event: 8 on span 1 Apr 8 13:41:27 WARNING[-1210639440]: PRI: Read on 32 failed: Unknown error 500 Apr 8 13:41:27 NOTICE[-1210639440]: PRI got event: 6 on span 1 Apr 8 13:42:07 WARNING[-1210639440]: PRI: Read on 32 failed: Unknown error 500 Apr 8 13:42:07 NOTICE[-1210639440]: PRI got event: 8 on span 1 Apr 8 16:44:01 WARNING[-1210631248]: PRI: Read on 34 failed: Unknown error 500 Apr 8 16:44:01 NOTICE[-1210631248]: PRI got event: 6 on span 1 Apr 8 16:44:01 WARNING[-1210631248]: PRI: Read on 34 failed: Unknown error 500 Apr 8 16:44:01 NOTICE[-1210631248]: PRI got event: 6 on span 1 Then this -- possibly not related ? Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Request) Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Request) Apr 8 16:51:46 WARNING[-1137157200]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 105 (Request) And finally, I'll show you a RESTART log Apr 8 17:41:32 WARNING[-1085030272]: Ignoring port for now Apr 8 17:41:33 WARNING[-1085030272]: XXX I don't work right with non-full duplex sound cards XXX Apr 8 17:41:33 WARNING[-1189983312]: Read error on sound device: Resource temporarily unavailable Apr 8 17:41:33 ERROR[-1085030272]: Unable to load config iax1.conf Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 1 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 2 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 3 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 4 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 5 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 6 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 7 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 8 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 9 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 10 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 11 Apr 8 17:41:38 WARNING[-1210963024]: PRI: Read on 32 failed: Unknown error 500 Apr 8 17:41:38 NOTICE[-1210963024]: PRI got event: 5 on span 1 Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 1: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 2: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 3: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 4: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 5: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 6: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 7: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 8: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 9: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 10: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 11: Red Alarm Apr 8 17:49:10 WARNING[-1210963024]: PRI: Read on 32 failed: Unknown error 500 Apr 8 17:49:10 NOTICE[-1210963024]: PRI got event: 4 on span 1 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 1 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 2 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 3 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 4 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 5 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 6 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 7 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 8 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 9 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 10 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 11 Apr 8 17:49:18 WARNING[-1210963024]: PRI: Read on 32 failed: Unknown error
RE: [Asterisk-Users] Who has access numbers in the UK and Germany?
sipgate.de has DIDs in Germany and the UK. -Alfred -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen Karrington Sent: Friday, April 09, 2004 4:08 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Who has access numbers in the UK and Germany? Hello, I need a few access numbers in the UK and Germany. Does anyone have this available right now? I need the incoming calls to be directed through IP to one of my asterisk servers in Europe. Please contact me off the list if you want. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ___ 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] Ignorepat with capi
Hi to all,
I'm trying to make outside call in this way :
ignorepat = 0
exten = _0.,1,Dial(CAPI/xxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
Bye
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Re: [Asterisk-Users] Who has access numbers in the UK and Germany?
I'm afraid I'm just out on a family 'outing', can you give me an overview via email of what you are looking for ? - Original Message - From: Stephen Karrington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, April 09, 2004 12:07 PM Subject: [Asterisk-Users] Who has access numbers in the UK and Germany? Hello, I need a few access numbers in the UK and Germany. Does anyone have this available right now? I need the incoming calls to be directed through IP to one of my asterisk servers in Europe. Please contact me off the list if you want. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ___ 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] Clearpath
Can anyone here help me in getting connected to Clearpath? They have supposedly setup a DID and 800 number for me but not provided login info. They're really hard to reach. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] If you dont really care, dont write. - Mark Bernstein, advice on writing ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] Ignorepat with capi
massimo wrote:
Hi to all,
I'm trying to make outside call in this way :
ignorepat = 0
exten = _0.,1,Dial(CAPI/xxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
Bye
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Try:
exten = _0.,1,Dial(CAPI/xxx:b${exten:1})
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Re: [Asterisk-Users] TigerJet ISDN card
I am using the NetJet cards from www.traverse.com.au, it uses a TigerJet 320 chip on it. It works fine with isdn4linux, you need the netjet driver under the passive cards. Also need the audio option in the kernel to make it work with asterisk. It appears this is already in the 2.4.24 kernel, but in older kernels you need a patch. You will also probably want to try various patches to disable DTMF detection in the kernel. Regards, Adam On Fri, 2004-04-09 at 10:52, Matthew Enger wrote: Arn't TigerJet isdn cards a type of netjet? If so try the netjet hisax drivers under isdn4linux. On Fri, 2004-04-09 at 03:00, Mark Phillips wrote: Is it CAPI compliant? if so yes Is there any Linux/* support for the TigerJet ISDN card? -brian G7LTT/KC2ENI Mark Phillips ___ 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[2]: [Asterisk-Users] Who has access numbers in the UK and Germany?
I can't read German. Can you outline the cost for me? Thanks. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ===8==Original message text=== sipgate.de has DIDs in Germany and the UK. -Alfred -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen Karrington Sent: Friday, April 09, 2004 4:08 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Who has access numbers in the UK and Germany? Hello, I need a few access numbers in the UK and Germany. Does anyone have this available right now? I need the incoming calls to be directed through IP to one of my asterisk servers in Europe. Please contact me off the list if you want. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ___ 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 ===8===End of original message text=== ___ 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] application Directory (Modified by Ryan Thrash)
On Thursday 08 April 2004 22:41, Ryan Thrash wrote: Scenario: a person selects an Auto Attendant option that fires off the Directory application (CVS circa 3/22). Three questions: 1) How do they escape if they didn't mean to go there in the first place (without having to hang up...)? Config of entry into the vertex directory below: exten = 1,1,Directory(vertex) exten = 1,2,Goto(s,200) If you just wait, Directory will exit if there is no entry. 2) Why is there a five second pause before the directory instructions start? Probably because you have another extension that begins with 1. Since Asterisk has no other way to know if the extension is complete, it waits DigitTimeout seconds (defaults to 5). 3) Why no option for first name (without recording your own custom message and reversing names in voicemail.conf)? Just wasn't written that way. You're welcome to submit a patch to add first name matching on the bugtracker (bugs.digium.com). -Tilghman ___ 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] TigerJet ISDN card
I am using the NetJet cards from www.traverse.com.au, it uses a TigerJet 320 chip on it. It works fine with isdn4linux, you need the netjet driver under the passive cards. Also need the audio option in the kernel to make it work with asterisk. It appears this is already in the 2.4.24 kernel, but in older kernels you need a patch. You will also probably want to try various patches to disable DTMF detection in the kernel. Regards, Adam On Fri, 2004-04-09 at 10:52, Matthew Enger wrote: Arn't TigerJet isdn cards a type of netjet? If so try the netjet hisax drivers under isdn4linux. On Fri, 2004-04-09 at 03:00, Mark Phillips wrote: Is it CAPI compliant? if so yes Is there any Linux/* support for the TigerJet ISDN card? -brian G7LTT/KC2ENI Mark Phillips ___ 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] application Directory
Let's say an unsuspecting soul accidently selects the Directory option from an Auto Attendant (CVS circa 3/22). Three questions: 1) How do they escape if they didn't mean to go there in the first place (without having to hang up...)? exten = 1,1,Directory(vertex) exten = 1,2,Goto(s,200) 2) Why is there a five second pause before the directory instructions start? 3) Why no option for first name (without recording your own custom message and reversing names in voicemail.conf)? Thanks, Ryan ___ 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] Ignorepat with capi
Try this:
exten = _0.,1,Dial(CAPI/xxx:b${EXTEN:1})
The :1 tells it to use everything except the first digit.
Robert Jackson
-Original Message-
From: massimo [mailto:[EMAIL PROTECTED]
Sent: Friday, April 09, 2004 6:59 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Ignorepat with capi
Hi to all,
I'm trying to make outside call in this way :
ignorepat = 0
exten = _0.,1,Dial(CAPI/xxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
Bye
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RE: [Asterisk-Users] Live Music on Hold
Ryan, Did you have to apply a patch to get this to work, or is it in CVS? You mentioned posting some doc of what you did? Thanks in advance, Ed _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan R. Fligg Sent: Thursday, April 08, 2004 3:10 PM To: 'Dan B. Long'; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Live Music on Hold Dan, I played around with this and have also been following the MOH posts. Playing MP3's seems to crash my system but I did get music on hold to stream. Here is my musiconhold.conf file: ; ; Music on hold class definitions ; [classes] ;default = quietmp3:/var/lib/asterisk/mohmp3/mp3 ;loud = mp3:/var/lib/asterisk/mohmp3/mp3 ;random = quietmp3:/var/lib/asterisk/mohmp3/mp3,-z ;stream = quietmp3:/var/lib/asterisk/empty,http://64.236.34.141:80/stream/1006,http:// 64.236.34.161:80/stream/1040,http://64.202.98.33:6230,http://69.10.147.34:80 40,http://64.202.98.75:6180,-z I just used a couple of my favorite internet radio stations. This is a start. I have documentation of exactly what I did to get it to work. Let me find it and I will post a followup reply. Good Luck. Ryan R. Fligg Secured Digital Storage, Inc. 1171 7th St. Suite 100 Des Moines, IA 50314-2525 Phone: (515)-244-6290 ext.205 Fax: (515)-244-6285 Cell: (515)-988-3773 E-Mail: [EMAIL PROTECTED] Website: http://www.dstorage.com _ From: Dan B. Long [mailto:[EMAIL PROTECTED] Sent: Thursday, April 08, 2004 3:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Live Music on Hold I have a small * system in my home (1 U100S, 1 X100P, 1 BT101, and 1 SPA2000) to handle my requirements. I would like to add Music on Hold and have been watching the forum to see if something would come across on this topic. The difference I am interested in is getting the music from a radio or someother external source. All references to MOH up to now have been using MP3 files and going through them. Is there some way to get the music from an external source instead of from files? Dan Long attachment: winmail.dat
RE: [Asterisk-Users] dreaded Caller*ID failed checksum
On Thu, 2004-04-08 at 13:30, Jeremy Hall wrote: I have no idea if that setting affects it or not. Is that a command line switch when starting Asterisk? Woah... I just moved the cards back to my old test box (dual cpu athlon). Guest what? CallerID worked *every time*. I noticed on my main box there seems to be a lot of RF noise getting into the cards. Maybe this is screwing up the DSP code. These are the same cards, same lines, different computers: CallerID doesn't work on: processor : 0 vendor_id : AuthenticAMD cpu family : 6 model : 10 model name : AMD Athlon(tm) XP 2600+ stepping: 0 cpu MHz : 1918.335 cache size : 512 KB CallerID *DOES* work on: processor : 0 vendor_id : AuthenticAMD cpu family : 6 model : 8 model name : AMD Athlon(tm) MP 2400+ stepping: 1 cpu MHz : 2000.103 cache size : 256 KB Also, is yours the true Digium card or is yours a One real, 3 clones. Customers get real cards, my poor testing lab gets clones. ...Jeff Thanks! Jeremy -Original Message- From: Jeff Gustafson [mailto:[EMAIL PROTECTED] Sent: Thursday, April 08, 2004 1:57 PM To: [EMAIL PROTECTED]; Jeremy Hall Subject: RE: [Asterisk-Users] dreaded Caller*ID failed checksum On Thu, 2004-04-08 at 07:03, Jeremy Hall wrote: Jeff, I see the same thing on my FXO card, but it is an Intel modem, not a true Digium X100P. I suspected it was my card, but if you are seeing it on a true card, maybe there is hope for mine yet. I haven't had time to troubleshoot yet as I have been having too much fun playing with other features. Does the -DOLD_DSP_ROUTINES effect Callerid? I tried setting that and I got maybe caller id maybe 5 out of 20 times instead of 0 out of 20 times. ...Jeff Let us know if you find the solution, and I will do the same if I get mine working. I am hoping to be able to do some work on it this weekend to try and see what is going on. In my case I have several other phones plugged into the line as I don't have any FXS ports yet, so eliminating them was going to be one of my first steps. The jack that my * server is attached to is CAT5 run directly from the telco access box. Aside from being a software decoding error or a telco sending error, my first suspects are line noise on the cabling from other devices or devices near the phone cabling. Electrical noise introduced into the signal inside the asterisk system is another failure point I want to try to eliminate. As a last resort, I was thinking of throwing that modem into my Windows PC and loading the drivers and software for it and see if CallerID works in that mode. I don't know if Windows would be able to load modem drivers for the Digium card or not, but that is another idea for you to try. These cards are basically glorified sound cards that attach to a telephone line, so if the Windows software can correctly read the signal, that would maybe point it in the software or driver area. If that turns out to be the case, I may be forced to go ahead and get an actual Digium card sooner than I anticipated in order to prove the theory. Regards, Jeremy -Original Message- From: Jeff Gustafson [mailto:[EMAIL PROTECTED] Sent: Thursday, April 08, 2004 12:06 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dreaded Caller*ID failed checksum Caller*ID used to work as some point, but I can't seem to get it going these days. The card is a x101p. I've tried going up and down the rxgain scale. Can the txgain effect it at all? When I plug in a phone into the line with a splitter it can decode caller id with no problems. Reading through the mailing list archives hasn't given me any move clues. Any ideas? ...Jeff ___ 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] IAX2 DTMF Problem
Hey all, I am dialing a DID through VoicePulse Connect. The number is answered by a main menu type of IVR. The configuration is as specified in both the wiki and VoicePulses documentation. The call comes through without a problem, but when the caller enter any keys they are either not recieved by * or they are ignored. With SIP I would typically put a dtmfmode= line under the peer and everything works great, but I am not sure how to attack this. I found a few items referring to the same issue in the list, but I didn't find any answers. If this is a bug I will create a report on the bugtracker, but I would rather make sure that I am not just completely dense and not seeing the easy answer. I'm trying to replicate the issue with NuFone. CVS from 2004-04-04 stable branch. Thanks, Robert Jackson ___ 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] Ignorepat with capi
On Fri, 2004-04-09 at 12:58, massimo wrote:
Hi to all,
I'm trying to make outside call in this way :
ignorepat = 0
exten = _0.,1,Dial(CAPI/xxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
I don't have CAPI but to get my analog to work I have
ignorepat = 9
exten = _9.,1,Dial(${DIALOUTANALOG}/${EXTEN:1})
--
Dave Cotton [EMAIL PROTECTED]
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Re[2]: [Asterisk-Users] Who has access numbers in the UK and Germany?
Well, I have a few things to discuss. 1. I need a few numbers for the UK and Germany to start. This is for one small project. 2. For my second project I need local access numbers in the major markets. This is for our system at www.diamondcard.us. This is a VOIP services MLM product. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ===8==Original message text=== I'm afraid I'm just out on a family 'outing', can you give me an overview via email of what you are looking for ? - Original Message - From: Stephen Karrington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, April 09, 2004 12:07 PM Subject: [Asterisk-Users] Who has access numbers in the UK and Germany? Hello, I need a few access numbers in the UK and Germany. Does anyone have this available right now? I need the incoming calls to be directed through IP to one of my asterisk servers in Europe. Please contact me off the list if you want. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ___ 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 ===8===End of original message text=== ___ 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] Who has access numbers in the UK and Germany?
Stephen Karrington wrote: I can't read German. Can you outline the cost for me? Thanks. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ===8==Original message text=== sipgate.de has DIDs in Germany and the UK. -Alfred -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen Karrington Sent: Friday, April 09, 2004 4:08 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Who has access numbers in the UK and Germany? Hello, I need a few access numbers in the UK and Germany. Does anyone have this available right now? I need the incoming calls to be directed through IP to one of my asterisk servers in Europe. Please contact me off the list if you want. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us What information are you looking for? They fee's are 1.79 Cent/Minute (euro that is) plus taxes. No monthly fee, no minimum amount of calls required. Their second option is for 8.90Euro you can get 1000 minutes/month. You will receive a free phone number. SIP to SIP is free. ___ 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[2]: [Asterisk-Users] Who has access numbers in the UK and Germany?
On Fri, 9 Apr 2004, Stephen Karrington wrote: I can't read German. Can you outline the cost for me? Thanks. http://www02.sipgate.de/user/tarife.php Tarife NATIONAL Deutschland* - E 1,79Ct/Min (US$ 0.021637) Cellular: E 22,90Ct/Min (US$ 0.276815) They have a plan which includes 1000 minutes (non-cellular), which costs E 8,90 (US$ 10.7583) - or US$ 0,01 per minute. ___ 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] Zaptel/PRI problem
Dimitri, I just got off the phone with digium. Here's what I (from my notes) the event codes mean Event 4: Alarm detected Event 5: Alarm cleared Event 6: Abort HDLC Frams Event 8: Bad HCS The 6 8 which occur sporadically are possibly causing the observed symptoms. Now ... what causes 6 8 is the question. Interrupt conflicts was one suggested possibility. Another possibility is 'stuff' from the Telco which is not understood / mis-understood by the driver. I'll keep the list posted. Willy - Original Message Follows - Dear Willy i notice the same problem with my E100P using the latest cvs zaptel driver i have try every type of config in /etc/zaptel.conf to check if i have missed something in timing conf but nothing... Digium help... :-) thanks in advance Dimitri On Thursday 08 April 2004 23:07, [EMAIL PROTECTED] wrote: Chris, Thank you for posting this. Since it concerns my 'production' system, let me comment. After 'downshifting' to a previous release (for no good reason other than desperation and teh fact that an earlier list entry had commented that it cleared up the problems) I am sad to report that the system failed again. Miscellaneous throughout the day: Apr 8 13:41:27 WARNING[-1210639440]: PRI: Read on 32 failed: Unknown error 500 Apr 8 13:41:27 NOTICE[-1210639440]: PRI got event: 8 on span 1 Apr 8 13:41:27 WARNING[-1210639440]: PRI: Read on 32 failed: Unknown error 500 Apr 8 13:41:27 NOTICE[-1210639440]: PRI got event: 6 on span 1 Apr 8 13:42:07 WARNING[-1210639440]: PRI: Read on 32 failed: Unknown error 500 Apr 8 13:42:07 NOTICE[-1210639440]: PRI got event: 8 on span 1 Apr 8 16:44:01 WARNING[-1210631248]: PRI: Read on 34 failed: Unknown error 500 Apr 8 16:44:01 NOTICE[-1210631248]: PRI got event: 6 on span 1 Apr 8 16:44:01 WARNING[-1210631248]: PRI: Read on 34 failed: Unknown error 500 Apr 8 16:44:01 NOTICE[-1210631248]: PRI got event: 6 on span 1 Then this -- possibly not related ? Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Request) Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Request) Apr 8 16:51:46 WARNING[-1137157200]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 105 (Request) And finally, I'll show you a RESTART log Apr 8 17:41:32 WARNING[-1085030272]: Ignoring port for now Apr 8 17:41:33 WARNING[-1085030272]: XXX I don't work right with non-full duplex sound cards XXX Apr 8 17:41:33 WARNING[-1189983312]: Read error on sound device: Resource temporarily unavailable Apr 8 17:41:33 ERROR[-1085030272]: Unable to load config iax1.conf Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 1 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 2 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 3 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 4 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 5 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 6 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 7 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 8 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 9 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 10 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 11 Apr 8 17:41:38 WARNING[-1210963024]: PRI: Read on 32 failed: Unknown error 500 Apr 8 17:41:38 NOTICE[-1210963024]: PRI got event: 5 on span 1 Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 1: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 2: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 3: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 4: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 5: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 6: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 7: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 8: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 9: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 10: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 11: Red Alarm Apr 8 17:49:10 WARNING[-1210963024]: PRI: Read on 32 failed: Unknown error 500 Apr 8 17:49:10 NOTICE[-1210963024]: PRI got event: 4 on span 1 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 1 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 2 Apr 8 17:49:18
[Asterisk-Users] NuFone and international dialing
Can someone send me a quick snippet of a dialplan for international dialing via NuFone? I'm having a hard time getting any help from them this week. Scott ___ 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] default caller id from X100P
Is there a way to set default caller id info to pass to * when the telco does not provide it? Regards, Victor Perez ___ 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] AGI - GET DATA not working on current stable cvs (anyone else?)
On Fri, 2004-04-09 at 00:59, Jeb Campbell wrote: Has anyone else had trouble with the AGI command GET DATA on the latest stable cvs? I can't get it to work with asterisk-perl, or by using print statements and reading stdin. I get 200 result= (timeout). (this is from the print statements, and asterisk-perl reports nothing). But asterisk is getting DTMF because my menu in extensions.conf works. I will go through the code Friday, but I just didn't know if anyone else was seeing this? Setup: 23 voice pri from Avaya PBX to Asterisk IVR. Make sure you Answer() the line before going into agi. -- Steven Critchfield [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
Re: [Asterisk-Users] IAX2 DTMF Problem
On Fri, 2004-04-09 at 10:12, Robert Jackson wrote: Hey all, I am dialing a DID through VoicePulse Connect. The number is answered by a main menu type of IVR. The configuration is as specified in both the wiki and VoicePulses documentation. The call comes through without a problem, but when the caller enter any keys they are either not recieved by * or they are ignored. With SIP I would typically put a dtmfmode= line under the peer and everything works great, but I am not sure how to attack this. I found a few items referring to the same issue in the list, but I didn't find any answers. If this is a bug I will create a report on the bugtracker, but I would rather make sure that I am not just completely dense and not seeing the easy answer. I'm trying to replicate the issue with NuFone. CVS from 2004-04-04 stable branch. Is this in the extensions.conf file or a agi? either way, maybe you should make sure you Answer() the call before anything else. After that and a clarification of where youa re looking for the DTMF it may be easier to answer your question. -- Steven Critchfield [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] small linux distro to run * in old boxes
Has anybody tried to install * in any of these minimalist linux distros like tinylinux? Which linux distro would you use to run * in old P2, P3 boxes? Regards, Victor Perez [EMAIL PROTECTED] (469) 221-4189 ___ 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] New Zealand indications.conf
Hi Vic,
I hit that same problem! My SIP phones would sound okay when I made
changes to indications.conf but incoming calls in to my TE410P had their
own thing going on!
Have a look at the zaptel source files, there's one called zonedata.c.
You'll see the au settings... replace what's there with this:
{ 1, au, Australia, { 400, 200, 400, 2000 },
{
{ ZT_TONE_DIALTONE, 400+425 },
{ ZT_TONE_BUSY, 425/375,0/375 },
{ ZT_TONE_RINGTONE, 400+425/400,0/200,400+425/400,0/2000 },
/* XXX Congestion: Should reduce by 10 db every other cadence
XXX */
{ ZT_TONE_CONGESTION, 425/375,0/375,420/375,0/375 },
{ ZT_TONE_CALLWAIT, 425/200,0/200,425/200,0/4400 },
{ ZT_TONE_DIALRECALL, 400+425 },
{ ZT_TONE_RECORDTONE, !425/1000,!0/15000,425/360,0/15000 },
{ ZT_TONE_INFO, 400/2500,0/500 },
{ ZT_TONE_STUTTER, 400+425/100,0/40 } },
},
Make clean; make install your zaptel and bam! The world sounds good once
more.
Benjamin Wakefield
[EMAIL PROTECTED]
http://www.dcsi.net.au/
DCSI - We do Internet.
64 Queen Street
Warragul, VIC 3820 AU
Ph: (+61) 1300 665 575
Fx: (+61) 1300 556 595
-BEGIN GEEK CODE BLOCK-
Version: 3.12
G! d- s: a-- C+ UL++ P+ L++ E W+ N+ o- K- w+$ O--- M-- V? PS !PE Y--
PGP- t 5 X+ R- tv b- DI-- D--- G-- e* h* r- z++
--END GEEK CODE BLOCK--
-Original Message-
From: Vic Cross [mailto:[EMAIL PROTECTED]
Posted At: Thursday, 8 April 2004 5:56 PM
Posted To: Asterisk
Conversation: [Asterisk-Users] New Zealand indications.conf
Subject: Re: [Asterisk-Users] New Zealand indications.conf
On Tue, 6 Apr 2004, Matt Riddell wrote:
Here are the settings for New Zealand indications. I have tested them
and
call progress works...voicemail messages used to contain 50 seconds of
disconnect tones, now just 2.
snipped the detail
So, all you did was update indications.conf with what you posted, and
everything worked? Wow...
Wait a minute... What kind of hardware are you using?
I am fighting with making the Zap stuff recognise and generate proper
tones for AU (on my X100P and TDM cards). Just updating
indications.conf
does not work for me -- the simple switch generates different tones that
are unrelated to what I've coded there (I've tested this with an
extension
that runs a bunch of PlayTones() apps -- PlayTones is correct, but the
simple switch does its own thing). As for analogue call progress,
forget
it -- having read the code, I cannot see how it could work at all on any
service that does not present US tones.
(Digium et.al. -- please don't take this as criticism. You guys have to
scratch the biggest and most annoying itches first! I wish I had the
time
and skill to contribute detection routines for other areas. Skill is
the
main problem for me, since progress tones in the US, based on MF tones
as
they are, are much easier to code to recognise than AU tones which are
all
different cadences - and in some cases, amplitudes - of the same single
frequency tone.)
Matt, good for you!
Cheers,
Vic Cross
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Re: [Asterisk-Users] NuFone and international dialing
Scott Laird wrote:
Can someone send me a quick snippet of a dialplan for international
dialing via NuFone? I'm having a hard time getting any help from them
this week.
exten = _3.,1,Dial,IAX2/[EMAIL PROTECTED]/011${EXTEN:1},60,tr
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
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[Asterisk-Users] Latency and 'Scratchy' Voice...
Title: Latency and 'Scratchy' Voice... Dear All, I have move from the USA to Sydney, Australia. I have gone from a data center environment at work and cable at home to a 513k/128k ADSL line. I'm experiencing two issues; 1) There is a latency of .5 - .8 seconds between me and the USA. 2) I have been in two calls where my voice has been describes as 'Scratchy'? I'm using a SIP Phone from SJ Phone, and a Plantronics USB Headset. In my Asterisk box I'm using the Quad T1 card. Any tips on how I could get around these two issues? I can understand the latency issue, what is contributing to the 'Scratchy' sound? I have not had this issue in the 4 months of running the product. Warm Regards and Thanks Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney
[Asterisk-Users] Asterisk Server Crashing with New Application
Title: Asterisk Server Crashing with New Application Dear All, I have been running a successful and very stable call center PBX based on 0.7.1 release. I need to be on this release because of a number of features that I have complied from 3rd party patches, for the call center. I will not be able to upgrade to release 1 until the patches catch up and I have done the required testing. The system was very stable until two days ago. The changes made were; 1) Installed a Second PBX in my second data center and I am running IAX2. 2) Installed the MySQL module. 3) Installed a copy of the php based CDR reporting. 4) Installed the Flash Operator Panel 5) Installed a modified version of Monastery to show me which agents were logged in and active I only stated having instability around the changes made in 4 and 5. I suspect the problem to be either caused by 4 or 5, in which case they will be very easy to rectify. I would however like to know if anyone else has had a) the same experience and b) has been able to isolate the issue. Warm Regards and Thanks Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Netural Bay Sydney
[Asterisk-Users] Re: [Asterisk-Users]: External access to voicemail
Hello steve. Here is a patch I wrote for app_voicemail.c which does
exactly as you describe. When the outgoing message is playing, if the
listener hits the * key, they're prompted for a mailbox and password,
whereupon they can check their voicemail as if they were using the internal
phone. I found no other way of doing this.
If you patch your app_voicemail.c, I have V1.44 from CVS as of
12/11/2003, with this diff file, and recompile the app_voicemail.so module
and install it in /usr/lib/asterisk/modules and then, from the command line
of Asterisk, do:
unload app_voicemail.so
load app_voicemail.so
you should have the new feature, all without having to stop and restart
asterisk.
Good luck, and let me know if it works for you.
-Brian
--- app_voicemail.c.fcs Thu Dec 11 12:55:25 2003
+++ app_voicemail.c Sat Feb 28 16:21:15 2004
@@ -1083,7 +1083,7 @@
char prefile[256]=;
char fmt[80];
char *context;
- char *ecodes = #;
+ char *ecodes = *#;
char *stringp;
time_t start;
time_t end;
@@ -1117,12 +1117,12 @@
if (mkdir(dir, 0700) (errno != EEXIST))
ast_log(LOG_WARNING, mkdir '%s' failed: %s\n, dir,
strerror(errno));
if (ast_exists_extension(chan, strlen(chan-macrocontext) ?
chan-macrocontext : chan-context, o, 1, chan-callerid))
- ecodes = #0;
+ ecodes = *#0;
/* Play the beginning intro if desired */
if (strlen(prefile)) {
if (ast_fileexists(prefile, NULL, NULL) 0) {
if (ast_streamfile(chan, prefile, chan-language)
-1)
- res = ast_waitstream(chan, #0);
+ res = ast_waitstream(chan, *#0);
} else {
ast_log(LOG_DEBUG, %s doesn't exist, doing what we
can\n, prefile);
res = invent_message(chan, vmu-context, ext, busy,
ecodes);
@@ -1138,6 +1138,10 @@
silent = 1;
res = 0;
}
+ if (res == '*') { /*break out to main vm*/
+ free_user(vmu);
+ return(100);
+ }
if (!res !silent) {
res = ast_streamfile(chan, INTRO, chan-language);
if (!res)
@@ -1156,6 +1160,10 @@
free_user(vmu);
return 0;
}
+ if (res == '*') { /*break out to main vm*/
+ free_user(vmu);
+ return(100);
+ }
if (res = 0) {
/* Unless we're *really* silent, try to send the beep */
res = ast_streamfile(chan, beep, chan-language);
@@ -2678,6 +2686,9 @@
}
res = leave_voicemail(chan, ext, silent, busy, unavail);
LOCAL_USER_REMOVE(u);
+ if (res == 100) { /*The user requested vm main*/
+ res = vm_execmain(chan, NULL);
+ }
return res;
}
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[Asterisk-Users] wcfxo module fail to load (Unable to request IRQ 0)
Hello, just compiled zaptel in mandrake 9.2 and this is what I get when trying
modprobe wcfxo:
Apr 9 11:35:15 localhost kernel: PCI: No IRQ known for interrupt pin A of devic
e 00:05.0. Please try using pci=biosirq.
Apr 9 11:35:15 localhost kernel: Setting hook state to 0 (08)
Apr 9 11:35:15 localhost kernel: Registered Span 1 ('WCFXO/0') with 1 channels
Apr 9 11:35:15 localhost kernel: Span ('WCFXO/0') is new master
Apr 9 11:35:15 localhost kernel: PCI: Setting latency timer of device 00:05.0 t
o 64
Apr 9 11:35:15 localhost kernel: wcfxo: Unable to request IRQ 0
I have this same setup (asterisk on mandrake 9.2) already working in other pc... this
is an old AT pc... any ideas?
Regards,
Victor Perez
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Victor Perez
Sent: Friday, April 09, 2004 11:03 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] small linux distro to run * in old boxes
Has anybody tried to install * in any of these minimalist linux distros like tinylinux?
Which linux distro would you use to run * in old P2, P3 boxes?
Regards,
Victor Perez
[EMAIL PROTECTED]
(469) 221-4189
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Re: [Asterisk-Users] small linux distro to run * in old boxes
Victor Perez wrote: Has anybody tried to install * in any of these minimalist linux distros like tinylinux? Which linux distro would you use to run * in old P2, P3 boxes? I have got it to install on Trustix (92MB min install) but I have moved to Fedora now for other reasons.. ___ 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] small linux distro to run * in old boxes
Title: RE: [Asterisk-Users] small linux distro to run * in old boxes I am using * on a RH9 380Mhz AMD K6 processor (With XP100 card), as well as Fedora Core 1 on a PII 333 Mhz machine for a couple of small SIP phone tests. One at work, and one at home. Things seems to be working just fine. -Original Message- From: Victor Perez [mailto:[EMAIL PROTECTED]] Sent: Friday, April 09, 2004 12:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] small linux distro to run * in old boxes Has anybody tried to install * in any of these minimalist linux distros like tinylinux? Which linux distro would you use to run * in old P2, P3 boxes? Regards, Victor Perez [EMAIL PROTECTED] (469) 221-4189 ___ 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] default caller id from X100P
In /etc/asterisk/zapata.conf before the channel=x (where x is the channel assigned to the FXO port) put: callerid=PSTN Call 1234567 You will need to restart * for this change to take effect Andy *** REPLY SEPARATOR *** On 09/04/2004 at 10:56 Victor Perez wrote: Is there a way to set default caller id info to pass to * when the telco does not provide it? Regards, Victor Perez ___ 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] small linux distro to run * in old boxes
I made a custom fedora mini distro, something like 350 megs, including apache,php,mysql webmin of course installable from a cd in 20 minutes, more or less :) at the end you have a fully working asterisk installations, along with some basic tools like webmin and a full webserver Matteo. Il ven, 2004-04-09 alle 18:02, Victor Perez ha scritto: Has anybody tried to install * in any of these minimalist linux distros like tinylinux? Which linux distro would you use to run * in old P2, P3 boxes? Regards, Victor Perez [EMAIL PROTECTED] (469) 221-4189 ___ 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] application Directory (Modified by Ryan Thrash)
On Apr 9, 2004, at 9:52 AM, Tilghman Lesher wrote: On Thursday 08 April 2004 22:41, Ryan Thrash wrote: Scenario: a person selects an Auto Attendant option that fires off the Directory application (CVS circa 3/22). Three questions: 1) How do they escape if they didn't mean to go there in the first place (without having to hang up...)? Config of entry into the vertex directory below: If you just wait, Directory will exit if there is no entry. Ah! So it does in fact. Thanks! Many people get impatient and start getting button-happy, often hanging up in frustration. Time to record a new message with instructions for the escape hatch! 2) Why is there a five second pause before the directory instructions start? Probably because you have another extension that begins with 1. Since Asterisk has no other way to know if the extension is complete, it waits DigitTimeout seconds (defaults to 5). And again, you are correct, sir. Internal extensions start at 100. Thanks. Time to re-record the message and assign new extensions for the prompts. 3) Why no option for first name (without recording your own custom message and reversing names in voicemail.conf)? Just wasn't written that way. You're welcome to submit a patch to add first name matching on the bugtracker (bugs.digium.com). Just signed up on Mantis today. Not being a coder, I'll see if I can poke around and get something to work with some help of some local friends that do have a clue. Again, thanks for your helpful response. : ) rt ___ 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] IAX2 DTMF Problem
I have the same problem, got this from VoicePulse today: Chris, Thank you for contacting VoicePulse. Our engineers are aware of the DTMF problem and are working to have it resolved as quickly as possible. Please reply directly to this email if we can provide any additional assistance. Regards, VoicePulse Customer Support On Fri, 9 Apr 2004, Steven Critchfield wrote: On Fri, 2004-04-09 at 10:12, Robert Jackson wrote: Hey all, I am dialing a DID through VoicePulse Connect. The number is answered by a main menu type of IVR. The configuration is as specified in both the wiki and VoicePulses documentation. The call comes through without a problem, but when the caller enter any keys they are either not recieved by * or they are ignored. With SIP I would typically put a dtmfmode= line under the peer and everything works great, but I am not sure how to attack this. I found a few items referring to the same issue in the list, but I didn't find any answers. If this is a bug I will create a report on the bugtracker, but I would rather make sure that I am not just completely dense and not seeing the easy answer. I'm trying to replicate the issue with NuFone. CVS from 2004-04-04 stable branch. Is this in the extensions.conf file or a agi? either way, maybe you should make sure you Answer() the call before anything else. After that and a clarification of where youa re looking for the DTMF it may be easier to answer your question. -- Steven Critchfield [EMAIL PROTECTED] -- chris maresca senior partner - www.olliancegroup.com linux, up 3 days ___ 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] small linux distro to run * in old boxes
Brancaleoni Matteo wrote: I made a custom fedora mini distro, something like 350 megs, including apache,php,mysql webmin of course installable from a cd in 20 minutes, more or less :) at the end you have a fully working asterisk installations, along with some basic tools like webmin and a full webserver Matteo. Are you going to be making this available or is it something yo created for inhouse use only? ___ 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] Problems with Zpateller on incoming external calls
Brian Cuthie wrote: I've setup the following in extensions.con: exten = 2200,1,Ringing exten = 2200,2,Wait(2) exten = 2200,3,Answer exten = 2200,4,Zapateller exten = 2200,5,Macro(stdexten,2205,SIP/2205) This works as expected if I dial from a SIP phone on my desk. However, if I dial in from the PSTN (through a SIP provider) it fails while trying to play ths SIT with: Apr 8 18:53:12 WARNING[1209269552]: rtp.c:407 ast_rtp_read: RTP Read error: Resource temporarily unavailable Any idea what's going on? My suspicion is that the PSTN gateway hasn't setup an audio path yet, although I thought Answer would do that. Cheers, Brian I don't have a zap device to test on, but can you do Ringing before you Answer? - Andrew Thompson http://aktzero.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] manager api problem
I've got following problem with manager api: In my Asterisk installation when I connect two channels (IAX,SIP) I get following sequence of events(these are events for *single* connection, come one by one without any delay): Event: Link Channel1: [EMAIL PROTECTED]:5036]/3 Channel2: SIP/kamyk-9950 Event: Unlink Channel1: [EMAIL PROTECTED]:5036]/3 Channel2: SIP/kamyk-9950 Event: Link Channel1: [EMAIL PROTECTED]:5036]/3 Channel2: SIP/kamyk-9950 and only then parties may speak to each other. When connection is hungup then another Unlink happens again. How manager client should interpret these events? First Link? Connection seems to has been setup. Following Unlink? Does it mean connection hungup or is it only indication of internal asterisk logic than one should not take care of? Next Link? Does it mean than another connection has been setup again or one should discard this? Maybe other channel events, and timing information should be taken into consideration to tell the difference between inconnection Link/Unlink events and those that mean call setup and hungup? It is possible, but makes manager client much more complicated then necessary. How do You detect call setup/tear down with manager interface? Isn't it sane to expect that for single connection there should be just two events: connected and disconnected (or pair of Link/Unlink)? If I am totally wrong, and miss something fundamental, please point me to relevant source code. Maciej Kaminski ___ 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] Voice mail notifications?
I know an email can be sent when a user get a voicemail message, but is there a way to send a message to a SIP phone to say they have a message? Or how hard would it be to write an app that could popup on a PC when there is a message in the mail box? 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
Re: [Asterisk-Users] Asterisk Server Crashing with New Application
Hi! 4) Installed the Flash Operator Panel 5) Installed a modified version of Monastery to show me which agents were [...] I suspect the problem to be either caused by 4 or 5, in which case they will be very easy to rectify. I would however like to know if anyone else has had a) the same experience and b) has been able to isolate the issue. Most likely this is an issue with the manager API - mattf has reported on this list more than once about problems as soon as more than one manager client is active on an Asterisk server. Cheers, Philipp ___ 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: Sasquatch, the Loch Ness Monster, UFOs and...Speech Recognition
Hi, John Todd said: 9) Speech recognition support Nothing towards this yet - sphinx keeps getting mentioned, though I don't know anyone who has had it running in anything other than a crippled test, or at least I don't remember anyone saying anything about it. Which features do Asterisk users a) need and b) desire for a speech recognition solution? Extensions to IVR and Auto Attendant applications are the first couple that spring to mind but what else should/could be included? Thoughts on size of vocabulary and API are of specific interest. Thanks Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: 08 April 2004 15:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Fwd: Sasquatch, the Loch Ness Monster, UFOs and... Every half year or so, I probably will repost this list, adding and subtracting as the community makes advances (or ignores what isn't required.) Date: Thu, 9 Oct 2003 04:51:23 -0400 To: asterisk-users-lists.digium.com From: John Todd [EMAIL PROTECTED] Subject: Sasquatch, the Loch Ness Monster, UFOs and... Mythical Asterisk Creatures, oft-discussed, rarely seen: 1) An advanced graphical user interface We're getting there. There are starting to appear a crop of PHP or in at least one case, Flash-driven front ends for users. These haven't been compiled as part of asterisk-addons, but perhaps sometime in the next month or two the code from the existing various projects can be pushed into the addons directory. 2) An IAX2 hardware device Any Day Now(tm). Wasim has fallen off the face of the Earth, but I've seen with my own two eyes a working copy of the Iaxy from Digium, so this holds promise. My request for a 1u 24-port IAX-based box that takes Digium daughterboards (FXO or FXS) generated some interest when a show of hands was asked for at the VON show... Bob Knight seemed to have an interest and some time on his hands. ;-) 3) A Radius CDR report module This sort-of exists now, but again is not a completely robust solution. I've not implemented it yet (due to other pressing issues of life and profit) but it should hopefully work with some of the traditional billing systems that existing VoIP carriers are using. 4) A live-method, robust SQL-based dialplan Not sure on this one - anyone care to comment? 5) LDAP/SQL/Radius authentication for SIP phones I hear rumors of this existing, but again, I haven't had the time to investigate. The SQL-friends database hacks might be the answer for an SQL system. 6) Robust R2 signalling support Steve Underwood says that he's made advances... has anyone else done any work on R2? 7) Multilingual language recordings of all existing * .gsm files Nothing that I know of towards this end, or at least, nothing that is available on the CVS server. Anyone? 8) Free exchange of PSTN gateways in a centralized routing arbiter model HO ho ho ho ho... that's a funny one. Actually, I have someone working on TRIP now, but I suspect that budget will get cut as soon as another project starts to explode. 9) Speech recognition support Nothing towards this yet - sphinx keeps getting mentioned, though I don't know anyone who has had it running in anything other than a crippled test, or at least I don't remember anyone saying anything about it. Here are this halfyear's additions: 10) Encryption I'd love to see TLS/SRTP built into the SIP stack, to support the Zultys and Sipura devices which now handle crypto natively. More clients will support this functionality; time to start building Asterisk to work with them. Additionally, IAX2 would be much cooler if it had a full-channel encryption method, which I know is at least being thought about (the aes header files have appeared in the CVS distro.) 11) Presence. Support for presence integration into devices would be great, and is this year's hot-button technology. Just simply supporting line appearances would help out quite a bit for business users on newer devices which support that feature, but the same technology (subscribe/notify) could be used for more advanced presence features. My ideas about integration into existing chat services might have some merit, or maybe not. 12) BSD Support We've got Asterisk compiling, now to get Zaptel/libpri working with Digium cards... rumors have someone Almost Done(tm) 13) High-density Zap cards Inexpensive DS3 Zap-driven cards would be a boon for large providers. The cards exist, there are Linux drivers, all that is required is some GPL'ed glue code and hair-pulling to weave it into Zaptel/libpri. With the data mode on Asterisk, it might also be possible to provide the equivalent of a Cisco CT3+ card that does voice as well. That's all I can think of at the moment. Comments are welcome. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To
RE: [Asterisk-Users] Problems with Zpateller on incoming external calls
Tried that, and no go. There's something wrong with Zapteller. It works fine on internal calls, but the only way I can get it to work on external calls (through a SIP/PSTN gateway, no Zap hw necessary) is to first play a message. For instance, this works: exten = 2200,1,Playback(ss-noservice) exten = 2200,2,Zapateller exten = 2200,3,Dial(SIP/2205) -brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Thompson Sent: Friday, April 09, 2004 12:48 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Problems with Zpateller on incoming external calls Brian Cuthie wrote: I've setup the following in extensions.con: exten = 2200,1,Ringing exten = 2200,2,Wait(2) exten = 2200,3,Answer exten = 2200,4,Zapateller exten = 2200,5,Macro(stdexten,2205,SIP/2205) This works as expected if I dial from a SIP phone on my desk. However, if I dial in from the PSTN (through a SIP provider) it fails while trying to play ths SIT with: Apr 8 18:53:12 WARNING[1209269552]: rtp.c:407 ast_rtp_read: RTP Read error: Resource temporarily unavailable Any idea what's going on? My suspicion is that the PSTN gateway hasn't setup an audio path yet, although I thought Answer would do that. Cheers, Brian I don't have a zap device to test on, but can you do Ringing before you Answer? - Andrew Thompson http://aktzero.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
Re: [Asterisk-Users] small linux distro to run * in old boxes
Hi I made a custom fedora mini distro, something like 350 megs, including apache,php,mysql webmin of course installable from a cd in 20 minutes, more or less :) at the end you have a fully working asterisk installations, along with some basic tools like webmin and a full webserver Are you going to be making this available or is it something yo created for inhouse use only? dunno yet. is not to me. the whole packahe contains also our web manager for asterisk (configuration and several tools like call recording,contacts,manager view,blah blah blah) that's not open. as soon as I'll have a fully working stable installer, (now works good, but I have to polish some things) *perhaps* I could arrange to distribute at least a version without the web manager... hope so :) Matteo ___ 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] small linux distro to run * in old boxes
So all you do is pop the CD in and install it and Asterisk is ready to go ? And it only takes up 350 MB ? Is there any way I could get a copy of it ?? Thanks, Paul. --- Original Message --- From: [EMAIL PROTECTED] on behalf of WipeOutPosted At: Fri 09/04/2004 17:55Posted To: Asterisk-UsersConversation: [Asterisk-Users] small linux distro to run * in old boxesSubject: Re: [Asterisk-Users] small linux distro to run * in old boxes Victor Perez wrote:Has anybody tried to install * in any of these minimalist linux distros like tinylinux?Which linux distro would you use to run * in old P2, P3 boxes?I have got it to install on Trustix (92MB min install) but I have movedto Fedora now for other reasons..___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 DTMF Problem
Running straight from extensions.conf, for now. The dialplan looks like this: [voicepulse-incoming] Exten = _NXXNXX,1,Goto(mainmenu,s,1) Exten = _NXXNXX,2,Hangup [mainmenu] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,Background(thankyouforcalling) exten = s,5,Background(mainmenu-prompts) exten = 1,1,VoicemailMain() exten = 1,2,GoTo(s,5) exten = 2,1,Directory exten = 2,2,Goto(s,5) exten = i,1,Playback(invalid) exten = h,1,Hangup exten = t,1,Hangup Thanks for your help, Robert Jackson -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Friday, April 09, 2004 11:59 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 DTMF Problem On Fri, 2004-04-09 at 10:12, Robert Jackson wrote: Hey all, I am dialing a DID through VoicePulse Connect. The number is answered by a main menu type of IVR. The configuration is as specified in both the wiki and VoicePulses documentation. The call comes through without a problem, but when the caller enter any keys they are either not recieved by * or they are ignored. With SIP I would typically put a dtmfmode= line under the peer and everything works great, but I am not sure how to attack this. I found a few items referring to the same issue in the list, but I didn't find any answers. If this is a bug I will create a report on the bugtracker, but I would rather make sure that I am not just completely dense and not seeing the easy answer. I'm trying to replicate the issue with NuFone. CVS from 2004-04-04 stable branch. Is this in the extensions.conf file or a agi? either way, maybe you should make sure you Answer() the call before anything else. After that and a clarification of where youa re looking for the DTMF it may be easier to answer your question. -- Steven Critchfield [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
RE: [Asterisk-Users] Voice mail notifications?
Most sip phones have a message indicator. To use it, just specify mailbox=1234 (or whatever the mailbox number is) is the phone's definition in the sip.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Friday, April 09, 2004 12:59 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voice mail notifications? I know an email can be sent when a user get a voicemail message, but is there a way to send a message to a SIP phone to say they have a message? Or how hard would it be to write an app that could popup on a PC when there is a message in the mail box? 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.645 / Virus Database: 413 - Release Date: 3/28/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.645 / Virus Database: 413 - Release Date: 3/28/2004 ___ 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
Re: [Asterisk-Users] Asterisk Server Crashing with New Application
Shad, I don't remember how far in the past, but a while back at least one person if not more reported instability in asterisk caused by more than one manager client connecting to the Asterisk server at the same time. Your monastery as well as the Flash Panel both access the manager application if my understanding of those applications is correct. The solution the person came up with was to put a single agent in front of the manager port to query the manager application on the Asterisk box and distribute the results to the client programs. You may have run into this same issue by running both flash operator and monastery. -Chris On 09:46 PM 4/8/2004, Shad Mortazavi wrote: snip 4) Installed the Flash Operator Panel 5) Installed a modified version of Monastery to show me which agents were logged in and active I only stated having instability around the changes made in 4 and 5.
Re: [Asterisk-Users] small linux distro to run * in old boxes
Hi Victor I'm currently working in a Linux Distro, it is being internal alpha testing by my self and a couple of me my colleagues, over the next couple of weeks I'm hoping to release a beta version to the asterisk community., I'll keep you posted via asterisk users, about its features as it developed. The first to note is Its currently a 28Mb ISO for installation with asterisk installed with zaptel, and lib pri this includes apache perl PHP, and Mysql I will be producing a web site I post the address when it is ready Regards Robb Victor Perez wrote: Has anybody tried to install * in any of these minimalist linux distros like tinylinux? Which linux distro would you use to run * in old P2, P3 boxes? Regards, Victor Perez [EMAIL PROTECTED] (469) 221-4189 ___ 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] Analogue telephone cards for the UK
Hi, Does anyone know where I can get a telephone card that will fit into the PCI slot on my PC and work with the UK telephone system (BT) ? I would really like the retailer to be based in the UK if at all possible ? Also, is there any way to set up asterisk so that only certain phones can make external calls, but all phones can receive incoming calls if they are routed to that phones by some sort of auto attendant ? Thanks in advance, Paul.
RE: [Asterisk-Users] Analogue telephone cards for the UK
Paul Tyreman [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) Does anyone know where I can get a telephone card that will fit into the PCI slot on my PC and work with the UK telephone system (BT) ? I would really like the retailer to be based in the UK if at all possible The nice people at TelAppliant will sell you an analogue FXO card, and are based in London, England. See here: http://www.voiptalk.org/ The Digium X100P (well, the X101P now) works in England with the notable exception of support for BT's caller ID. Also, is there any way to set up asterisk so that only certain phones can make external calls, but all phones can receive incoming calls if they are routed to that phones by some sort of auto attendant ? All phones should be allocated a context from which to begin their search of the dialplan (extensions.conf). Phones can all be told to start from the same context or can start from different contexts, as you see fit. You can then include other contexts into the various top-level contexts you set up, as appropriate for your application. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ 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
RE: [Asterisk-Users] small linux distro to run * in old boxes
Robert Boardman [EMAIL PROTECTED] wrote: The first to note is Its currently a 28Mb ISO for installation with asterisk installed with zaptel, and lib pri this includes apache perl PHP, and Mysql That's impressive. My MySQL installation has munched its way through 48MB of disk space on its own - and that's without a database. Asterisk uses a further 16MB on my setup. Did you mean 280MB by any chance? :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ 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
Re: [Asterisk-Users] Analogue telephone cards for the UK
Kevin Walsh wrote: The nice people at TelAppliant will sell you an analogue FXO card, and are based in London, England. See here: http://www.voiptalk.org/ The Digium X100P (well, the X101P now) works in England with the notable exception of support for BT's caller ID. Some UK cable companies (eg NTL or Telewest) use bellcore (US) caller id in certain areas but they use BT standard in others. The only way to be certain to get caller id with * at the moment is to use an ISDN line (this will require an ISDN line card, not the x101p). The new FXO (external line) ports (available soon) for the TDM400P will be _capable_ of receiving BT's caller id but whether support for it gets added into the driver is a different matter. regards Julien ___ 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] RE: Asterisk Server Crashing with New Application
Chris, This does sound like my scenario. Do you remember how they achieved this? Now that I have removed these components Im stable again. Thanks for the feedback and help. Warm Regards Shad From: Chris A. Icide [mailto:[EMAIL PROTECTED] Sent: Saturday, April 10, 2004 4:46 AM To: [EMAIL PROTECTED] Cc: Shad Mortazavi Subject: Re: [Asterisk-Users] Asterisk Server Crashing with New Application Shad, I don't remember how far in the past, but a while back at least one person if not more reported instability in asterisk caused by more than one manager client connecting to the Asterisk server at the same time. Your monastery as well as the Flash Panel both access the manager application if my understanding of those applications is correct. The solution the person came up with was to put a single agent in front of the manager port to query the manager application on the Asterisk box and distribute the results to the client programs. You may have run into this same issue by running both flash operator and monastery. -Chris On 09:46 PM 4/8/2004, Shad Mortazavi wrote: snip 4) Installed the Flash Operator Panel 5) Installed a modified version of Monastery to show me which agents were logged in and active I only stated having instability around the changes made in 4 and 5.
[Asterisk-Users] IAX phone for Pocket PC
Hello all, Does anyone know of a good IAX softphone for Pocket PC's? 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] small linux distro to run * in old boxes
Kevin Walsh wrote: Did you mean 280MB by any chance? :-) He said that was the iso size, I managed to get debian installed down to about 32megs, but this was minus apache, php, mysql... but you can compress the installer files on the iso and then have it extract them as it installs... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ 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] syslog error
Hello, I have been running into a problem on my server (which I believe was the cause of an O/S crash earlier today). I am consistently seeing the following messages in /var/log/messages: Apr 8 05:24:04 east insmod: /lib/modules/2.4.20-8/misc/torisa.o: insmod char-major-196 failed Apr 8 05:24:04 east kernel: Zapata Telephony Interface Registered on major 196 Apr 8 05:24:04 east kernel: No ISA tormenta card found at d Apr 8 05:24:04 east kernel: Zapata Telephony Interface Unloaded Apr 8 05:24:04 east kernel: Zapata Telephony Interface Registered on major 196 Apr 8 05:24:04 east kernel: No ISA tormenta card found at d Apr 8 05:24:04 east kernel: Zapata Telephony Interface Unloaded Apr 8 05:24:04 east insmod: /lib/modules/2.4.20-8/misc/torisa.o: init_module: Input/output error Apr 8 05:24:04 east insmod: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg I have put the following line in modules.conf (under [modules]): noload = chan_zap.so And I am also receiving the following error when doing a modprobe of ztdummy: /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed Does anyone have any ideas on why this might be happening? It looks to me like either something is missing on my system or I did something incorrectly at compile time. Regards, -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
Re: [Asterisk-Users] small linux distro to run * in old boxes
At 08:01 PM 4/9/2004 +0100, Robb wrote: I'm currently working in a Linux Distro, it is being internal alpha ... The first to note is Its currently a 28Mb ISO for installation with asterisk installed with zaptel, and lib pri this includes apache perl PHP, and Mysql Is there anything in apache thats actually NEEDED? Perhaps a nice straight forward boa web server can be used? It would also be nice to have a super scaled down dedicated box that would rely on a 2nd box for the database, web support, and such, thus you're fully dedicating the machine to just *. I guess it depends how many phones you want to run, and how streamlined the machine/os needs to be. As a side note, I've used MeshAP, which is a wireless mesh server box that runs on a small scaled down linux distro. It has X, web browser, webcam server, and a bunch of networking type stuff, and it is around 28 meg as well. You can actually boot from CD, and a unique ID derived from the machine, and the box gets it config via http, and runs without needing write access to the drive. Too bad theres not alot of hardware support for FreeBSD, we'd probably just need a couple of megs. A fully functioning base system would only be 1.44 meg. - - - Jon ___ 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] RedHat/Fedora RPMS Update
Greetings folks, I have updated our asterisk RPM repository with CVS builds for RH 7.3, 9 and FC1. The zaptel package is compiled against the supplied kerenel rpm and SRPMS are supplied for those wishing to rebuild. Other than the CVS update no other changes are made from previous releases... Features Security: runs as user asterisk not root Convienance: Console automatically runs on tty8 Newb friendly: Lots of links to documentation / hints Smarter: SRPM CVS update now packages the updated source code Compatable: Redhat style configuration (/etc/sysconfig/asterisk) Please don't scream at Digium or the list if our package don works for you. contact me. Also I am off the list for the time being so if you post regarding these packages please copy my email addy. ftp://ftp.linuxsys.com/pub/LSE/packages/ Cheers! -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. PO BOX 3791 Tallahassee, FL 32315 (850)224-5737 (850)294-7567 ___ 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] MeetMe conference
Hi !! all !! My MeetMe is moving by SIP. Does Ztdummy load to the kernel? - Original Message - From: Jain, Sonal [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, April 09, 2004 4:53 AM Subject: [Asterisk-Users] MeetMe conference I am trying to setup MeetMe conference. In my MeetMe.conf file I have [rooms] conf = 4001,4001 In my extension.conf file I have the following: exten =4001,1,MeetMe(4001|p|4001) When I try to call the extension 4001 it gives me the following error message. I am using SIP and I have not created 4001 in my Sip.conf file. Do I need to create this extension and also how do I fix this error. Apr 8 15:49:16 NOTICE[-1394906192]: sched.c:218 sched_settime: Request to schedule in the past?!?! Apr 8 15:49:16 WARNING[-1394906192]: file.c:521 ast_readaudio_callback: Failed to write frame ___ 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] small linux distro to run * in old boxes
Just disable (via the .conf file) and remove most of the apache modules -- its very small then. Small enough to go on a LEAF/LRP to drive the control interface, anyway. There are a lot of modules that only make sense on a full web serving sitution (like mod_speling for example) but if you get rid of them things get really light, apache-wise. Cheers, Mathew - Original Message - From: Jon Myers [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, April 10, 2004 10:36 AM Subject: Re: [Asterisk-Users] small linux distro to run * in old boxes At 08:01 PM 4/9/2004 +0100, Robb wrote: I'm currently working in a Linux Distro, it is being internal alpha ... The first to note is Its currently a 28Mb ISO for installation with asterisk installed with zaptel, and lib pri this includes apache perl PHP, and Mysql Is there anything in apache thats actually NEEDED? Perhaps a nice straight forward boa web server can be used? It would also be nice to have a super scaled down dedicated box that would rely on a 2nd box for the database, web support, and such, thus you're fully dedicating the machine to just *. I guess it depends how many phones you want to run, and how streamlined the machine/os needs to be. As a side note, I've used MeshAP, which is a wireless mesh server box that runs on a small scaled down linux distro. It has X, web browser, webcam server, and a bunch of networking type stuff, and it is around 28 meg as well. You can actually boot from CD, and a unique ID derived from the machine, and the box gets it config via http, and runs without needing write access to the drive. Too bad theres not alot of hardware support for FreeBSD, we'd probably just need a couple of megs. A fully functioning base system would only be 1.44 meg. - - - Jon ___ 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] small linux distro to run * in old boxes
Yes !! My Asterisk is working by the spec of the P2 average. - Original Message - From: Victor Perez [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, April 10, 2004 1:02 AM Subject: [Asterisk-Users] small linux distro to run * in old boxes Has anybody tried to install * in any of these minimalist linux distros like tinylinux? Which linux distro would you use to run * in old P2, P3 boxes? Regards, Victor Perez [EMAIL PROTECTED] (469) 221-4189 ___ 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] New Zealand indications.conf
On Sat, 10 Apr 2004, Benjamin Wakefield wrote: Have a look at the zaptel source files, there's one called zonedata.c. You'll see the au settings... replace what's there with this: detail snipped Benjamin, LEGEND! ;) Don't know why I didn't see this sooner -- thanks indeed! For my ear 412+437 works better than 400+425, but only because I'm really fussy (it makes the 'main' tone 425Hz, like the spec says; plus, it sounds identical to my POTS line). Cheers, Vic Cross ___ 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] vm e-mail notification stopped
After rebooting my asteriks server, e-mail notifications are no longer being sent after a voice-mail is left. I can see the messages in /var/spool/asterisk/vm. has anybody had the same experience? how was it resolved? Uri
