[Asterisk-Users] Would this work?
Title: Message
I am trying to
implement a rollover of extensions.
exten = 3000,1,GotoIf($[${line1} =
Congestion]?3:2)exten = 3000,2,Dial(${line1},15,rt)exten =
3000,3,GotoIf($[${line2} = Congestion]?5:4)exten =
3000,4,Dial(${line2},15,rt)exten = 3000,5,GotoIf($[${line3} =
Congestion]?7:6)exten = 3000,6,Dial(${line3},15,rt)exten =
3000,7,GotoIf($[${line4} = Congestion]?1:8)exten =
3000,8,Dial(${line4},15,rt)exten = 3000,9,Hangup
The $line[x]
represents a Zap Channel.
Thanks,
-gcc
RE: [Asterisk-Users] Would this work?
Thank you... I guess I was making this harder than it need's to be...
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Shaw
Posted At: Monday, June 28, 2004 6:15 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Would this work?
Subject: Re: [Asterisk-Users] Would this work?
MessageIf I am understanding your dialplan snippet correctly, you simply
want * to call extensions in a linear (or even round robin) fashion,
ringing the first one that's not busy correct? This functionality is
built directly into * and needs no special dialplan to implement. Please
check the Wiki or This list about Grouping Zap channels...
- Original Message -
From: AstGrp
To: [EMAIL PROTECTED]
Sent: Monday, June 28, 2004 12:01 PM
Subject: [Asterisk-Users] Would this work?
I am trying to implement a rollover of extensions.
exten = 3000,1,GotoIf($[${line1} = Congestion]?3:2)
exten = 3000,2,Dial(${line1},15,rt)
exten = 3000,3,GotoIf($[${line2} = Congestion]?5:4)
exten = 3000,4,Dial(${line2},15,rt)
exten = 3000,5,GotoIf($[${line3} = Congestion]?7:6)
exten = 3000,6,Dial(${line3},15,rt)
exten = 3000,7,GotoIf($[${line4} = Congestion]?1:8)
exten = 3000,8,Dial(${line4},15,rt)
exten = 3000,9,Hangup
The $line[x] represents a Zap Channel.
Thanks,
-gcc
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[Asterisk-Users] Question - TDM40B - Hunt Group Possibility??
Title: Message I was wondering if this is possible. I have a situation where I am connecting to a third party voicemail system from asterisk. I know this does not make since to everyone, but it has to be this way. Basically - I have an application that runs on the Asterisk system and when an employee calls into this system, they have an option to check there voicemail. This is where it needs to go over to the voicemail system. I would usually use an FXO card for this, but the other phone vendor I am working with is wondering is it possible to put the FXS cards I have in a hunt group - then I could call one of these ports and would ring the other voicemail system. If this can't be done that's fine - I have some FXO cards on order... Just thought I would check if anyone has ever done anything like this before. Thanks, Geoff Clark
RE: [Asterisk-Users] Asterisk + VoiceWorks
I have a customer who has a Comdial Phone System and uses VoiceWorks for it's voicemail. I am installing an IVR / Time Clock system utilizing asterisk. But they want to have an option to hop over to the VoiceWorks system to check voicemail... -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian Posted At: Tuesday, May 11, 2004 2:52 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Asterisk + VoiceWorks Subject: RE: [Asterisk-Users] Asterisk + VoiceWorks Why on earth would you wanna do something like that? Asterisk has voicemail and you even have the src so you can add those nifty features the PHB's like to have but never use! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of AstGrp Sent: Tuesday, May 11, 2004 11:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk + VoiceWorks I have a need to interface Asterisk with a VoiceWorks voicemail system. I was wondering what kind of card would be needed either a FXO or FXS interface? Any help would be appreciated. Thanks, -gcc ___ 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] Asterisk + VoiceWorks
I have a need to interface Asterisk with a VoiceWorks voicemail system. I was wondering what kind of card would be needed either a FXO or FXS interface? Any help would be appreciated. Thanks, -gcc ___ 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 Assitance
I am trying to figure this out... I'm sure it's simple, but I can't
think of it right now
In my AGI Script I am doing this... (This is done in Perl)
$AGI-exec('Record',
/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav);
And after this is done.. I want to get the name of the file it created
so I can store it in a database.
Any thoughts
Thanks,
-gcc
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RE: [Asterisk-Users] AGI Assistance
That is not working...
I tried like you mentioned it and even a few different ways and will not
create the file at all
Am I doing something wrong...
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
Lieberman
Posted At: Sunday, May 09, 2004 4:00 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] AGI Assistance
Subject: RE: [Asterisk-Users] AGI Assistance
Declare the file path before you record it.
$path = /usr/local/apache/htdocs/demo/sound/myapp.$date.wav;
$AGI-exec('Record',$path:wav);
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of AstGrp
Sent: Sunday, May 09, 2004 3:47 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AGI Assitance
I am trying to figure this out... I'm sure it's simple, but I can't
think of it right now
In my AGI Script I am doing this... (This is done in Perl)
$AGI-exec('Record',
/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav);
And after this is done.. I want to get the name of the file it created
so I can store it in a database.
Any thoughts
Thanks,
-gcc
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RE: [Asterisk-Users] AGI Assistance
Never Mind... Figured it out... Thanks...
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
Lieberman
Posted At: Sunday, May 09, 2004 4:00 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] AGI Assistance
Subject: RE: [Asterisk-Users] AGI Assistance
Declare the file path before you record it.
$path = /usr/local/apache/htdocs/demo/sound/myapp.$date.wav;
$AGI-exec('Record',$path:wav);
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of AstGrp
Sent: Sunday, May 09, 2004 3:47 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AGI Assitance
I am trying to figure this out... I'm sure it's simple, but I can't
think of it right now
In my AGI Script I am doing this... (This is done in Perl)
$AGI-exec('Record',
/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav);
And after this is done.. I want to get the name of the file it created
so I can store it in a database.
Any thoughts
Thanks,
-gcc
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RE: [Asterisk-Users] AGI Assistance
The error was in the Quotes and Date Variable
The end result looks as follows...
$Date = time();
$Path = /usr/local/apache/htdocs/demo/sound/$EmpNum.$Date;
$ShortPath = sound/$EmpNum.$Date;
$AGI-exec('Record', $Path:wav);
I needed the $ShortPath variable for some web values... But besides
that This is what did it for me
Thanks,
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Posted At: Sunday, May 09, 2004 6:40 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] AGI Assistance
Subject: RE: [Asterisk-Users] AGI Assistance
Can you post your error to the list so we know what was wrong?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Sent: Monday, 10 May 2004 7:34 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] AGI Assistance
Never Mind... Figured it out... Thanks...
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
Lieberman Posted At: Sunday, May 09, 2004 4:00 PM Posted To: Asterisk
User Group
Conversation: [Asterisk-Users] AGI Assistance
Subject: RE: [Asterisk-Users] AGI Assistance
Declare the file path before you record it.
$path = /usr/local/apache/htdocs/demo/sound/myapp.$date.wav;
$AGI-exec('Record',$path:wav);
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of AstGrp
Sent: Sunday, May 09, 2004 3:47 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AGI Assitance
I am trying to figure this out... I'm sure it's simple, but I can't
think of it right now
In my AGI Script I am doing this... (This is done in Perl)
$AGI-exec('Record',
/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav);
And after this is done.. I want to get the name of the file it created
so I can store it in a database.
Any thoughts
Thanks,
-gcc
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RE: [Asterisk-Users] Voicemail: upgraded?
Look at bugs.digium.com - Search for Voicemail... You will find it there... Geoff Clark Network Engineer The Network Essentials [EMAIL PROTECTED] 704-568-0031 (W) 704-622-3905 (C) www.tnessentials.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Elkins Sent: Friday, May 07, 2004 5:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicemail: upgraded? I'm sure I saw a posting about someone updating the CVS with a more richly featured voicemail system. What happened? Am I wrong? Can't seem to find anything on this... -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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] Stripping numbers at the end of a dial pattern = extensions.conf
http://www.voip-info.org/wiki-Asterisk+cmd+StripLSD
gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hermann
Wecke
Posted At: Saturday, May 08, 2004 6:04 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Stripping numbers at the end of a dial
pattern = extensions.conf
Subject: [Asterisk-Users] Stripping numbers at the end of a dial pattern
= extensions.conf
Is it possible to strip some numbers from the *end* of a number?
I know that ${EXTEN:1} will remove 1 position from the beggining... but
how to remove N numbers from the end?
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[Asterisk-Users] Need Help with Dial Plan
Let me lay it out for you
Call comes in over a T1 - Signal is em_w. The extension is seen as
*callerid*last 4 digits of number being called*. Which is fine in
it self.
I have my extension.conf file set up as follows...
[did]
; Receive call as *calling*called
exten = _.,1,Answer
exten = _.,2,Cut(CALLING=EXTEN,*,2)
exten = _.,3,SetCIDNum(${CALLING})
exten = _.,4,Cut(CALLED=EXTEN,*,3)
exten = _.,5,Goto(main,${CALLED},1)
include = main
[main]
exten = 0031,1,Answer
exten = 0031,2,Goto(TNE-SG,s,1)
Include = did
include = TNE-SG
[TNE-SG]
exten = s,1,Answer
;exten = s,2,agi,tne.agi
exten = s,2,Background(tne-main-thanks)
exten = s,3,Background(tne-main-menu)
exten = 1,1,Goto(default-tne,9100,1)
exten = 2,1,Goto(default-tne,4100,1)
exten = 3,1,Goto(default-tne,4200,1)
exten = 4,1,Goto(default-tne,4300,1)
exten = 5,1,Goto(default-tne,4400,1)
exten = 6,1,Goto(tne-main-menu,s,3)
exten = 7,1,Hangup
include = default-tne
include = main
[default-tne]
include = TNE-SG
; Geoff Clark
exten = 4001,1,Macro(stdexten,4001,SIP/gclark)
;exten = 4001,1,Dial(IAX/home:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = 4004,1,Macro(stdexten,4004,SIP/home)
; Kyle Elworthy
exten = 4002,1,Macro(stdexten,4002,SIP/kelworth)
exten = 4003,1,Macro(stdexten,4003,SIP/khome)
; Tech Support Agents
exten = *6,1,AgentCallbackLogin(4001,[EMAIL PROTECTED])
exten = *7,1,AgentCallbackLogin(4002,[EMAIL PROTECTED])
exten = 401,1,Dial(Zap/g1/7046223905)
exten = 402,1,Dial(Zap/g1/7049071514)
exten = 411,1,Answer
exten = 411,2,Wait,2
exten = 411,3,Background(auth-thankyou)
exten = 411,4,Queue(tech-supp)
Where the problem comes in is - I can dial in fine in this scenerio -
but when I go to make an outbound call, it calls the did context and
cut's the call up.
My problem appears to be I need it one way but not the other.. I hope
this makes since...
Thanks,
-gcc
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RE: [Asterisk-Users] Need Help with Dial Plan
Just an update resolved my own issue
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Posted At: Monday, April 19, 2004 9:25 PM
Posted To: Asterisk User Group
Conversation: Need Help with Dial Plan
Subject: [Asterisk-Users] Need Help with Dial Plan
Let me lay it out for you
Call comes in over a T1 - Signal is em_w. The extension is seen as
*callerid*last 4 digits of number being called*. Which is fine in
it self.
I have my extension.conf file set up as follows...
[did]
; Receive call as *calling*called
exten = _.,1,Answer
exten = _.,2,Cut(CALLING=EXTEN,*,2)
exten = _.,3,SetCIDNum(${CALLING})
exten = _.,4,Cut(CALLED=EXTEN,*,3)
exten = _.,5,Goto(main,${CALLED},1)
include = main
[main]
exten = 0031,1,Answer
exten = 0031,2,Goto(TNE-SG,s,1)
Include = did
include = TNE-SG
[TNE-SG]
exten = s,1,Answer
;exten = s,2,agi,tne.agi
exten = s,2,Background(tne-main-thanks)
exten = s,3,Background(tne-main-menu)
exten = 1,1,Goto(default-tne,9100,1)
exten = 2,1,Goto(default-tne,4100,1)
exten = 3,1,Goto(default-tne,4200,1)
exten = 4,1,Goto(default-tne,4300,1)
exten = 5,1,Goto(default-tne,4400,1)
exten = 6,1,Goto(tne-main-menu,s,3)
exten = 7,1,Hangup
include = default-tne
include = main
[default-tne]
include = TNE-SG
; Geoff Clark
exten = 4001,1,Macro(stdexten,4001,SIP/gclark)
;exten = 4001,1,Dial(IAX/home:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = 4004,1,Macro(stdexten,4004,SIP/home)
; Kyle Elworthy
exten = 4002,1,Macro(stdexten,4002,SIP/kelworth)
exten = 4003,1,Macro(stdexten,4003,SIP/khome)
; Tech Support Agents
exten = *6,1,AgentCallbackLogin(4001,[EMAIL PROTECTED])
exten = *7,1,AgentCallbackLogin(4002,[EMAIL PROTECTED])
exten = 401,1,Dial(Zap/g1/7046223905)
exten = 402,1,Dial(Zap/g1/7049071514)
exten = 411,1,Answer
exten = 411,2,Wait,2
exten = 411,3,Background(auth-thankyou)
exten = 411,4,Queue(tech-supp)
Where the problem comes in is - I can dial in fine in this scenerio -
but when I go to make an outbound call, it calls the did context and
cut's the call up.
My problem appears to be I need it one way but not the other.. I hope
this makes since...
Thanks,
-gcc
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RE: [Asterisk-Users] Cisco 7940 no audio
Title: Message Try upgrading to SIP 6.3. I heard from someone on the IRC Channel about this problem and 6.3 resolved it -gcc -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig WaddingtonPosted At: Friday, April 16, 2004 1:04 PMPosted To: Asterisk User GroupConversation: Cisco 7940 no audioSubject: [Asterisk-Users] Cisco 7940 no audio When we receive or make a call to the outside they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks.
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($[${autoattendant} = 1]?3:4)
exten = s,3,Dial(SIP/recep,30,t)
exten = s,4,Goto(main,s,1)
[main]
exten = s,1,Answer
exten = s,2,Background(ctm-main-thanks)
exten = 1,1,Goto(default-ctm,3001,1)
exten = 2,1,Goto(default-ctm,3002,1)
exten = 0,1,Goto(default-pb,2002,1)
exten = 3,1,Hangup
Hope this helps
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Moran
Posted At: Thursday, April 08, 2004 1:48 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Auto Attendant??
Subject: [Asterisk-Users] Auto Attendant??
I'm having trouble finding documentation for the auto attendant does
anyone have an idea where there might be some???
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RE: [Asterisk-Users] Voice Mail Email problem
It's probably sending the domain as the domain setup on the * server...
Change host to somedomain.com and see if that helps...
Thanks,
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan
Posted At: Wednesday, April 07, 2004 7:32 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Voice Mail Email problem
Subject: [Asterisk-Users] Voice Mail Email problem
Ok its probabally something really eaisy im missing. I've searched the
archives and voip-info.
Asterisk is trying to send the email notification for voice mail. But
the log says Invalid sender. Sender = [EMAIL PROTECTED]
and not [EMAIL PROTECTED] as assigned in conf file.
VM Config:
[general]
format=gsm|wav49|wav
[EMAIL PROTECTED] Actual file has a valid
email. attach=no maxmessage=30 silencethreshold=128 maxsilence=10
fromstring=Asterisk Mail
emailbody=${VM_NAME} ${VM_MAILBOX}\n\nYou have received a ${VM_DUR}
long message from ${VM_CALLERID}. The Message was left on ${VM_DATE}
[bell]
100 = 1234,User1,[EMAIL PROTECTED] -- Actual file has a
valid
email.
Thanks in advance.
Kyle
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RE: [Asterisk-Users] CallerID
Resolved the issue... It turned out to be a problem with the ISP -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AstGrp Posted At: Monday, April 05, 2004 2:21 PM Posted To: Asterisk User Group Conversation: CallerID Subject: [Asterisk-Users] CallerID I am having an issue with Callerid (INBOUND). I have a system set up with 4 companies sitting behind the system. On all of the companies except of one of them, it displays callerid withh 'asterisk'. The other company displays the callerid of the person calling. Zapata.conf [channels] musiconhold=default callerid=asreceived threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes jitterbuffers=4 immediate=no context=default-nga signalling=featd group=2 channel = 5-8 context=default-tne signalling=featd group=1 channel = 1-4 context=default-pb signalling=featd group=3 channel = 9-12 context=default=ctm signalling=featd group=3 channel = 13-14 Any thoughts -gcc ___ 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] CallerID
Thank you for the response... After talking with the ISP they did not have CID turned on all of the trunk groups. This has since been resolved. Thanks again, -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Maj Posted At: Tuesday, April 06, 2004 1:41 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] CallerID Subject: Re: [Asterisk-Users] CallerID On Mon, 5 Apr 2004, AstGrp waxed: I am having an issue with Callerid (INBOUND). I have a system set up with 4 companies sitting behind the system. On all of the companies except of one of them, it displays callerid withh 'asterisk'. The other company displays the callerid of the person calling. callerid=asreceived That's a good line to have. context=default-nga signalling=featd group=2 channel = 5-8 context=default-tne signalling=featd group=1 channel = 1-4 context=default-pb signalling=featd group=3 channel = 9-12 context=default=ctm signalling=featd group=3 channel = 13-14 What context is the company in that gets the cid right ? Maybe you are only receiving the cid on certain channels ? Why do you have 3 groups, but 4 contexts ? Is everything hooked up to a channel bank ? What kind of hardware is installed on the box ? Are you explicitly setting the cid in extensions.conf ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] CallerID
I am having an issue with Callerid (INBOUND). I have a system set up with 4 companies sitting behind the system. On all of the companies except of one of them, it displays callerid withh 'asterisk'. The other company displays the callerid of the person calling. Zapata.conf [channels] musiconhold=default callerid=asreceived threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes jitterbuffers=4 immediate=no context=default-nga signalling=featd group=2 channel = 5-8 context=default-tne signalling=featd group=1 channel = 1-4 context=default-pb signalling=featd group=3 channel = 9-12 context=default=ctm signalling=featd group=3 channel = 13-14 Any thoughts -gcc ___ 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 - Cisco 7960 - NAT
Can you post some of your sip configs and your extension configs. Thanks, -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Parlee Posted At: Sunday, April 04, 2004 12:10 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Asterisk - Cisco 7960 - NAT Subject: [Asterisk-Users] Asterisk - Cisco 7960 - NAT We have 10 Cisco 7960 phones at our office and a single static IP. Our asterisk server sits in the colo facility at our ISP. All phones are setup with a unique voip_control_port and they are all able to dial out. However, my phone is the only one that can receive a call. Every phone in the office can dial my extension and it will ring. I can call our main number and my phone will ring. But no other phone will ring! I get a fastbusy signal when trying to dial someone else's extension from my phone or from another phone. Can someone please help! 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 ___ 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] documents
http://www.asteriskdocs.org/ -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Posted At: Monday, March 22, 2004 2:31 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] documents Subject: [Asterisk-Users] documents hi..?? do you know web site where i can download document about install and configure software asterisk and zaptel...?? please.! Cheers. vozip ___ 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] If you know your party's extension # please dial it now ...
If you have your IVR under context [mainmenu] and your extensions under
context [default]. Then make sure you include context default under
context mainmenu...
Because your mainmenu context does not know about any other extensions
if you don't.
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Posted At: Sunday, March 21, 2004 8:37 AM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] If you know your party's extension #
please dial it now ...
Subject: [Asterisk-Users] If you know your party's extension # please
dial it now ...
Hi all,
I've built the usual press one for sales, 2 for support IVR which
works fine but I'm having difficulty in allowing callers to type in
whole extension numbers.
My internal extn ranges are 3xxx and 4xxx. I have pasted the IVR below
(just in case someone wants one). The welcome message states callers
should type in the extension number they want or choose from the
options. It seems though that one can only press one number before the
IVR moves to the next step.
I'm starting to think that if my extn's are 3xxx and 4xxx I can't have
any menu choices beginning with 3 or 4. Would this be correct? If so how
does the received DTMF break out of the IVR and get matched to the
relevant dialplan entry?
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,5
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test exten
= s,5,Background(welcomemsg) exten = s,6,Background(choosemsg)
; Sales
exten = 1,1,Dial,SIP/3400|20
exten = 1,2,Voicemail(3400)
exten = 1,3,Goto(mainmenu,s,60
; Tech support
exten = 2,1,Dial,SIP/3401|20
exten = 2,2,Voicemail(3401)
exten = 2,2,Goto(mainmenu,s,1)
; Echo Test
exten = 3,1,Playback(demo-echotest)
exten = 3,2,Echo
exten = 3,3,Playback(demo-echodone)
exten = 3,4,Goto(mainmenu,s,6)
; Parrot Test
exten = 4,1,Goto(205,1)
; Access VoiceMail
exten = 5,1,VoicemailMain
exten = 5,2,Goto(mainmenu,s,6)
; Play the weasels
exten = 6,1,Wait,3
exten = 6,2,Playback(tt-somethingwrong)
exten = 6,3,Playback(tt-weasels)
exten = 6,4,Wait,2
exten = 6,5,Goto(mainmenu,s,6)
; # to hangup
exten = #,1,Playback(vm-goodbye)
exten = #,2,Hangup
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
Whilst writing this I've had a thought. What would happen if I had an
entry like this?
; transfer to regular extension #
exten = _3XXX,1,Dial(SIP/{EXTN}|20|T)
exten = _4XXX,1,Dial(SIP/{EXTN}|20|T)
Thanks
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/ ___
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RE: [Asterisk-Users] AGI test script
Whats the script doing.. Is the script failing...? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vikram Rangnekar Posted At: Tuesday, March 16, 2004 2:35 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] AGI test script Subject: [Asterisk-Users] AGI test script exten = 666,1,Answer exten = 666,2,AGI(agi-text.agi) exten = 666,103,Hangup iwhy is that not working any idea. Does answer need to be there or does the AGI script answer the call. -- regards Vikram (http://www.vicramresearch.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] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Ok.. After upgrading the PIX to version PIX6.3(3). I can register the phone, but I am having related issue of sorts... Here's the low down.. The outside interface of the PIX is doing PAT. And I have one to one NAT translation for the * Server... But if I configure everything this way... I get an Unreachable... But if I put the PAT IP in for the NAT IP in the SIP file, it registers fine, but then no sound is heard through the phone Any ideas.. gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Monday, March 15, 2004 5:24 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Some firewalls when doing nat will alter the return address (need to make nat work) but not recalculate the header checksum, (Sonic walls come to mind.), Linux will proply delete any tcp/udp packet that fails its checksum at the kernel level, and send an error to the app. If this is happening to you Asterisk should log some kind of error. AstGrp wrote: Update... I did some more testing today.. And with the same setup but one box behind a Linksys router and another box behind a Pix firewall.. The linksys works with no problems... So it appears to be how the PIX is handling NAT SIP... If any one has any thoughts on this , it would be greatly appreciated. And thank you James for the support you have given today. Thanks, gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AstGrp Posted At: Friday, March 12, 2004 4:29 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Do I need to associate the outside interface of the PIX with the phone on the inside.. I don't remember doing this before... Setup * Server --- PIX FW --- WWW CLOUD PIX FW --- IP Phone Again the only difference than before is the First PIX FW Old setup was (Different server though) * Server Linksys Router WWW CLOUD PIX FW IP Phone Any thoughts? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Friday, March 12, 2004 2:58 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call The pings are pinging the out side port on the nat device, You don't have a rule in your nat table to associate it with a device on the inside. You should reset the phone and then see if the qualify shows a return time. You will need to make the phone register every time you change you config till the qualify shows a time. A good way to do this is to reboot the phone. Your nat device will have a default time that it keep nat rules in its table. Your qualify time will need to be lower then this value. AstGrp wrote: ___ 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] I must be an Idiot
If the definition of Insane is doing the same thing over and over again,
but expecting different results.. I must be falling into this category
today... Ok let me lay out the configs first
extension.conf
[macro-stdexten]
exten = s,1,DBget(temp=CFIM/${ARG1})
exten = s,2,Dial(Zap/g2/${temp}1)
exten = s,102,Goto(s|3)
exten = s,3,Dial(${ARG2},15)
exten = s,103,Goto(s|50)
exten = s,4,Voicemail2(u${ARG1})
exten = s,5,Hangup
exten = s,104,Voicemail2(b${ARG1}) ; busy
exten = s,105,Hangup
[default]
; Extension
; Geoff Clark
exten = 4001,1,Macro(stdexten,4001,SIP/gclark)
So far so good
; Call Forward to (Cell, etc.)
exten = _*5X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:2})
exten = _*5X.,2,Hangup
exten = *5,1,DBdel(CFIM/${CALLERIDNUM})
exten = *5,2,Hangup
### Here's where the problem comes in...
I can enter the number to forward to, but that's where the trouble
begins... After that is done, I call my extension from another cell
phone and in the CLI it shows the call going out over Zap/g2 - but
nothing ever happens.. Except that the Cell Phone you are calling in on
is being called back by *. I am completely lost. I am having other
issues with this same function for different apps. So I'm sure once
this get's resolved the rest will fall into place.
Hope somebody has some thoughts...
Thank you,
Geoff
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[Asterisk-Users] AgentCallBackLogin ??
I could use a little assistance.. I am sure I am doing something
stupid. The problem I am having is when the call comes in and runs
the context [411]. The call is generated, but never makes the call. It
rings back the user who is making the call. It works fine if I dial
context [411] from the inside. It sounds like I need to add some
context somewhere just not sure what where?
[agents]
agent = 4001,4001,Geoff Clark
[general]
[default]
[tech]
member = Agent/4001
strategy = roundrobin
timeout = 30
retry = 10
[411]
exten = 411,1,Answer
exten = 411,2,Wait,2
exten = 411,3,Background(auth-thankyou)
exten = 411,4,Queue(tech)
exten = *6,1,AgentCallbackLogin(@411)
exten = *4001,1,Dial(${TRUNK}/${GCELL:${TRUNKMSD}})
Thanks,
gcc
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RE: [Asterisk-Users] AgentCallBackLogin ??
It looks like the AgentCallBackLogin app is not working in the latest
CVS Asterisk CVS-03/15/04. Can someone please verify this. I had
this exact setup running on a different CVS load prior to running the
updates.
Thanks,
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Posted At: Monday, March 15, 2004 3:27 PM
Posted To: Asterisk User Group
Conversation: AgentCallBackLogin ??
Subject: [Asterisk-Users] AgentCallBackLogin ??
I could use a little assistance.. I am sure I am doing something
stupid. The problem I am having is when the call comes in and runs
the context [411]. The call is generated, but never makes the call. It
rings back the user who is making the call. It works fine if I dial
context [411] from the inside. It sounds like I need to add some
context somewhere just not sure what where?
[agents]
agent = 4001,4001,Geoff Clark
[general]
[default]
[tech]
member = Agent/4001
strategy = roundrobin
timeout = 30
retry = 10
[411]
exten = 411,1,Answer
exten = 411,2,Wait,2
exten = 411,3,Background(auth-thankyou)
exten = 411,4,Queue(tech)
exten = *6,1,AgentCallbackLogin(@411)
exten = *4001,1,Dial(${TRUNK}/${GCELL:${TRUNKMSD}})
Thanks,
gcc
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RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Thank you... I found that document last night.. And I have the pix configured this way with fixup sip... But still no go.. I am going to try and upgrade the pix tonight and see if that helps. gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Varga Sent: Saturday, March 13, 2004 10:28 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call On Friday 12 March 2004 09:28 pm, AstGrp wrote: Do I need to associate the outside interface of the PIX with the phone on the inside.. I don't remember doing this before... Setup * Server --- PIX FW --- WWW CLOUD PIX FW --- IP Phone Again the only difference than before is the First PIX FW Old setup was (Different server though) * Server Linksys Router WWW CLOUD PIX FW IP Phone Any thoughts? You may want to look at this page from Cisco http://www.cisco.com/en/US/products/hw/vpndevc/ps2030/ products_configuration_example09186a00801fc74a.shtml It looks like it will take care of the PAT/NATing issues. I have not have the luxury of trying it. HTH, 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] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Ok.. Let me start by saying that SJPhone works fine through NAT and the Cisco phones inside the internal network work fine also... It's just the Cisco phones on the outside using NAT. For Testing I opened the Firewall open on the IP for the * Server. I have done, everything you recommended below, but still no go... When the phone registers with port 2842? Not the standard 5060? Any ideas? I believe this is where my problem sits... Thanks, -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Friday, March 12, 2004 9:03 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Make sure your using qualify=500 in the sip.conf along with nat=yes, make sure any firewalls allow 5060 udp and tcp and random ports above 1 in form your PBX. If you have all that it should work. AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Thursday, March 11, 2004 10:47 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call You do have : nat_enable: 1 nat_received_processing: 1 On the Ciscos? AstGrp wrote: I am having a similar problem... I get the same message, but inbound calls can go through This is only Cisco phones that are behind NAT. I have tried your recommendations from below, but still no luck.. User can make outbound calls, just can't receive any. Any ideas would be greatly appreciated.. I even tried to change the timeout value in chan_sip, but it just waits longer to fail.. Just dosen't seem to want to communicate... Thanks, gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Are you using Cisco phones. ? I had this issue with my cisco phones. I didn't had any issues with dropped calls. All I did to fix this was set a prefered_codex and set proxy_register to 0. I hope this helps. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call *CLI Mar 3 12:55:05 WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) This has been brought up in the previous post but it does not seem to have an answer for it so far. I cvs the stable v1.0 this morning after compiling and installing I have calls drop 1 minutes into the connection with the above message. If anyone has any idea of this occurrence. I have set up sip.conf: canreinvite=no -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Ok... If put in the qualify=500... It says it is unreachable... But ping times Are fine... PING 69.133.182.77 (69.133.182.77) from 10.100.254.21 : 56(84) bytes of data. 64 bytes from 69.133.182.77: icmp_seq=1 ttl=241 time=54.9 ms 64 bytes from 69.133.182.77: icmp_seq=2 ttl=241 time=52.0 ms 64 bytes from 69.133.182.77: icmp_seq=3 ttl=241 time=54.2 ms 64 bytes from 69.133.182.77: icmp_seq=5 ttl=241 time=57.9 ms 64 bytes from 69.133.182.77: icmp_seq=6 ttl=241 time=56.0 ms 64 bytes from 69.133.182.77: icmp_seq=7 ttl=241 time=54.0 ms Any thoughts there? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Friday, March 12, 2004 11:50 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call I have noticed that sometimes you need to comment out profiles with nat=yes on and then reload, then uncomment them and reload, for Asterisk to clean out historical settings. Try that. I have run phones before on odd port with out trouble, so I don't think that is your problem. AstGrp wrote: Ok.. Let me start by saying that SJPhone works fine through NAT and the Cisco phones inside the internal network work fine also... It's just the Cisco phones on the outside using NAT. For Testing I opened the Firewall open on the IP for the * Server. I have done, everything you recommended below, but still no go... When the phone registers with port 2842? Not the standard 5060? Any ideas? I believe this is where my problem sits... Thanks, -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Friday, March 12, 2004 9:03 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Make sure your using qualify=500 in the sip.conf along with nat=yes, make sure any firewalls allow 5060 udp and tcp and random ports above 1 in form your PBX. If you have all that it should work. AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Thursday, March 11, 2004 10:47 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call You do have : nat_enable: 1 nat_received_processing: 1 On the Ciscos? AstGrp wrote: I am having a similar problem... I get the same message, but inbound calls can go through This is only Cisco phones that are behind NAT. I have tried your recommendations from below, but still no luck.. User can make outbound calls, just can't receive any. Any ideas would be greatly appreciated.. I even tried to change the timeout value in chan_sip, but it just waits longer to fail.. Just dosen't seem to want to communicate... Thanks, gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Are you using Cisco phones. ? I had this issue with my cisco phones. I didn't had any issues with dropped calls. All I did to fix this was set a prefered_codex and set proxy_register to 0. I hope this helps. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call *CLI Mar 3 12:55:05 WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) This has been brought up in the previous post but it does not seem to have an answer for it so far. I cvs the stable v1.0 this morning after compiling and installing I have calls drop 1 minutes into the connection with the above message. If anyone has any idea of this occurrence. I have set up sip.conf: canreinvite=no -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 ___ 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] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Update... I did some more testing today.. And with the same setup but one box behind a Linksys router and another box behind a Pix firewall.. The linksys works with no problems... So it appears to be how the PIX is handling NAT SIP... If any one has any thoughts on this , it would be greatly appreciated. And thank you James for the support you have given today. Thanks, gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AstGrp Posted At: Friday, March 12, 2004 4:29 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Do I need to associate the outside interface of the PIX with the phone on the inside.. I don't remember doing this before... Setup * Server --- PIX FW --- WWW CLOUD PIX FW --- IP Phone Again the only difference than before is the First PIX FW Old setup was (Different server though) * Server Linksys Router WWW CLOUD PIX FW IP Phone Any thoughts? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Friday, March 12, 2004 2:58 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call The pings are pinging the out side port on the nat device, You don't have a rule in your nat table to associate it with a device on the inside. You should reset the phone and then see if the qualify shows a return time. You will need to make the phone register every time you change you config till the qualify shows a time. A good way to do this is to reboot the phone. Your nat device will have a default time that it keep nat rules in its table. Your qualify time will need to be lower then this value. AstGrp wrote: Ok... If put in the qualify=500... It says it is unreachable... But ping times Are fine... PING 69.133.182.77 (69.133.182.77) from 10.100.254.21 : 56(84) bytes of data. 64 bytes from 69.133.182.77: icmp_seq=1 ttl=241 time=54.9 ms 64 bytes from 69.133.182.77: icmp_seq=2 ttl=241 time=52.0 ms 64 bytes from 69.133.182.77: icmp_seq=3 ttl=241 time=54.2 ms 64 bytes from 69.133.182.77: icmp_seq=5 ttl=241 time=57.9 ms 64 bytes from 69.133.182.77: icmp_seq=6 ttl=241 time=56.0 ms 64 bytes from 69.133.182.77: icmp_seq=7 ttl=241 time=54.0 ms Any thoughts there? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Friday, March 12, 2004 11:50 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call I have noticed that sometimes you need to comment out profiles with nat=yes on and then reload, then uncomment them and reload, for Asterisk to clean out historical settings. Try that. I have run phones before on odd port with out trouble, so I don't think that is your problem. AstGrp wrote: Ok.. Let me start by saying that SJPhone works fine through NAT and the Cisco phones inside the internal network work fine also... It's just the Cisco phones on the outside using NAT. For Testing I opened the Firewall open on the IP for the * Server. I have done, everything you recommended below, but still no go... When the phone registers with port 2842? Not the standard 5060? Any ideas? I believe this is where my problem sits... Thanks, -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Friday, March 12, 2004 9:03 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Make sure your using qualify=500 in the sip.conf along with nat=yes, make sure any firewalls allow 5060 udp and tcp and random ports above 1 in form your PBX. If you have all that it should work. AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Thursday, March 11, 2004 10:47 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call You do have : nat_enable: 1 nat_received_processing: 1 On the Ciscos? AstGrp wrote: I am having a similar problem... I get the same message, but inbound calls can go through This is only Cisco phones that are behind NAT
RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Here's a copy of the cisco config -- Current *FLASH* Configuration -- Platform : Cisco IP Phone 7940 Elasped Time: 00:01:37 dhcp_server : 10.100.0.2 my_ip_addr : 10.100.0.150 subnet_mask : 255.255.255.0 defaultgw : 10.100.0.2 dyn_dns_addr_1 : 0.0.0.0 dyn_dns_addr_2 : 0.0.0.0 dns_addr : 10.100.254.7 dns_backup_1: 24.93.68.65 tftp_addr : 66.64.246.36 dyn_tftp_addr : 0.0.0.0 my_mac_addr : 000f:23ac:4559 domain_name : tnessentials.com my_name : SIP000F23AC4559 Status Flags : 1230 image_version : P0S3-06-2-00 FirmLoadID : PC030301 DSPLoadID : PS03AT38 network_media_type : Auto network_port2_type : Hub/Switch tos_media : 5 phone_label : TNE PBX VOIP tftp_cfg_dir : phone_password : ** phone_prompt : SIP Phone language : english sntp_mode : DirectedBroadcast sntp_server : time_zone : EST dst_offset : 1 dst_start_month : April dst_start_day : 0 dst_start_day_of_week : Sun dst_start_week_of_month : 1 dst_start_time : 02 dst_stop_month : Oct dst_stop_day : 0 dst_stop_day_of_week : Sunday dst_stop_week_of_month : 8 dst_stop_time : 2 dst_auto_adjust : 1 time_format_24hr : 1 date_format : M/D/Y nat_enable : 1 nat_address : voip_control_port : 5060 start_media_port : 16456 end_media_port : 17456 sync : 1 xml_card_dir : xml_card_file : CARD.XML telnet_level : 2 services_url : directory_url : logo_url : http_proxy_addr : http_proxy_port : 80 enable_vad : 0 dial_template : dialplan callerid_blocking : 0 anonymous_call_block : 0 autocomplete : 1 messages_uri : 55 dnd_control : 0 preferred_codec : g711ulaw dtmf_outofband : avt dtmf_avt_payload : 101 dtmf_db_level : 3 dtmf_inband : 1 line1_name : khome line2_name : UNPROVISIONED line1_authname : khome line2_authname : UNPROVISIONED line1_password : ** line2_password : ** line1_shortname : UNPROVISIONED line2_shortname : UNPROVISIONED line1_displayname : Kyle Elworthy line2_displayname : proxy1_address : 66.64.246.36 proxy2_address : proxy1_port : 5060 proxy2_port : 5060 sip_retx : 10 sip_invite_retx : 6 timer_t1 : 500 timer_t2 : 4000 timer_invite_expires : 180 timer_register_expires : 3600 proxy_register : 1 proxy_backup : proxy_emergency : proxy_backup_port : 5060 proxy_emergency_port : 5060 outbound_proxy : outbound_proxy_port : 5060 nat_received_processing : 1 mwi_status : 0 call_waiting : 1 user_info : none cnf_join_enable : 1 remote_party_id : 0 semi_attended_transfer : 1 call_hold_ringback : 0 stutter_msg_waiting : 0 cfwd_url : call_stats : 1 auto_answer : 0 local_cfwd_enable : 1 timer_register_delta : 5 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Thursday, March 11, 2004 10:47 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call You do have : nat_enable: 1 nat_received_processing: 1 On the Ciscos? AstGrp wrote: I am having a similar problem... I get the same message, but inbound calls can go through This is only Cisco phones that are behind NAT. I have tried your recommendations from below, but still no luck.. User can make outbound calls, just can't receive any. Any ideas would be greatly appreciated.. I even tried to change the timeout value in chan_sip, but it just waits longer to fail.. Just dosen't seem to want to communicate... Thanks, gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Are you using Cisco phones. ? I had this issue with my cisco phones. I didn't had any issues with dropped calls. All I did to fix this was set a prefered_codex and set proxy_register to 0. I hope this helps. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call *CLI Mar 3 12:55:05 WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) This has been brought up in the previous post but it does not seem to have an answer for it so far. I cvs the stable v1.0 this morning after compiling and installing I have calls drop 1 minutes into the connection with the above message. If anyone has any idea of this occurrence. I have set up sip.conf: canreinvite=no -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman
RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
I am having a similar problem... I get the same message, but inbound calls can go through This is only Cisco phones that are behind NAT. I have tried your recommendations from below, but still no luck.. User can make outbound calls, just can't receive any. Any ideas would be greatly appreciated.. I even tried to change the timeout value in chan_sip, but it just waits longer to fail.. Just dosen't seem to want to communicate... Thanks, gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Are you using Cisco phones. ? I had this issue with my cisco phones. I didn't had any issues with dropped calls. All I did to fix this was set a prefered_codex and set proxy_register to 0. I hope this helps. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call *CLI Mar 3 12:55:05 WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) This has been brought up in the previous post but it does not seem to have an answer for it so far. I cvs the stable v1.0 this morning after compiling and installing I have calls drop 1 minutes into the connection with the above message. If anyone has any idea of this occurrence. I have set up sip.conf: canreinvite=no -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 ___ 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] Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk
I have this working, with not much work... SIP CONF [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; address to bind to externip = NAT IP / Outside IP ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 10.100.254.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context=default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc allow=alaw [travel] type=friend username=travel secret=password host=dynamic nat=yes context=local mailbox=4003 Ports in the Firewall Port 5060 UDP Ports 16456 - 17456 UDP RTP Conf rtpstart=16456 rtpend=17456 -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Beaumont Posted At: Sunday, February 29, 2004 4:12 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk Subject: [Asterisk-Users] Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk On the wiki pages it suggests that clients on the outside of NAT can connect to an Asterisk server behind nat. (option no 3). The note suggests that this can work with port forwarding and some 'header mangling magic'. I have the port forwarding configured however, when I try to connect an external client through the firewall the client does not correctly register. The REGISTER message is received, the server responds with Status 100 trying, followed by Status 407 Proxy Authentication required. This repeated several times. I guessing but could this be where the 'header mangling magic' is required. ? Does anyone know how this magic can be applied. Many thanks Steve Beaumont ___ 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] Macro Forward Calls
Having a problem with call forwarding If I call into the main number
go through the auto attendant and choose the persons extension it
forwards out to there alt number they specified. But if call them
directly via there DID... The call rings back the person calling the
DID.. Dosen't make since to me any ideas
-gcc
exten = s,1,DBget(temp=CFIM/${ARG1})
exten = s,2,Dial(Zap/g2/${temp}1)
exten = s,102,Goto(s|3)
exten = s,3,Dial(${ARG2},15)
exten = s,103,Goto(s|50)
exten = s,4,Voicemail2(u${ARG1})
exten = s,5,Hangup
exten = s,104,Voicemail2(b${ARG1}) ; busy
exten = s,105,Hangup
exten = 2564,1,Answer
exten = 2564,2,Goto(default,3001,1)
include = default
;Todd Greene
exten = 3001,1,Macro(stdexten,tgreene,SIP/tgreene)
; Call Forward to (Cell, etc.)
exten = _*5X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:2})
exten = _*5X.,2,Hangup
exten = *5,1,DBdel(CFIM/${CALLERIDNUM})
exten = *5,2,Hangup
___
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[Asterisk-Users] app_directory.c
I was wondering how difficult it would be to add a 2-3 sec delay before the Name or Extension is said. Some of our customers who call in are complaining that when they search for an employee by name by the time they have put there phone back to there ear the name has already been said. Hope my rambling makes since. I am not a C programer by any stretch... Any assistance would be greatly appreciated. Thanks, -gcc ___ 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] Sip Register Fail - NAT
I am having an issue with registering SIP client w/ NAT. I have set this up before on other boxes... But for some reason this one is not acting the same... I have attached a sip debug from the registration... For what ever reason it does not appear to be setting up the nat session correctly Am I seeing something wrong or even doing something wrong -gcc SIP CONFIG ## ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = nat ip ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 10.100.254.0; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context=default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc allow=alaw [4003] type=friend username=4003 secret=4003 host=dynamic qualify=500 context=local nat=yes mailbox=4003 ## SIP DEBUG #3 Sip read: REGISTER sip:10.100.254.21 SIP/2.0 Content-Length: 0 Contact: sip:192.168.1.10 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=10990022 CSeq: 87 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.10:5060 8 headers, 0 lines Using latest request as basis request Sending to 192.168.1.10 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990022 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 87 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 69.132.68.17:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990022 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 87 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=267e89bd Content-Length: 0 to 69.132.68.17:5060 ^Dtnevoip*CLI Sip read: REGISTER sip:10.100.254.21 SIP/2.0 Content-Length: 0 Contact: sip:192.168.1.10 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=10990413 CSeq: 88 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.10:5060 Proxy-Authorization: Digest username=4003,realm=asterisk,nonce=267e89bd,uri=sip:10.100.254.21 ,response=fb30e53fffc30ea15fc97acf7d82322f 9 headers, 0 lines Using latest request as basis request Sending to 192.168.1.10 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990413 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 88 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 69.132.68.17:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990413 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 88 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 69.132.68.17:5060 Feb 22 19:33:23 [1;33;40mNOTICE[0;37;40m[-1147384912]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m5577[0;37;40m [1;37;40mhandle_request[0;37;40m: Registration from 'sip:[EMAIL PROTECTED]' failed for '69.132.68.17' ^Dtnevoip*CLI Sip read: REGISTER sip:10.100.254.21 SIP/2.0 Content-Length: 0 Contact: sip:192.168.1.10 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=10990613 CSeq: 89 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.10:5060 8 headers, 0 lines Using latest request as basis request Sending to 192.168.1.10 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990613 To: sip:[EMAIL PROTECTED];tag=as42b62c4b Call-ID: [EMAIL PROTECTED] CSeq: 89 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content- ___ Asterisk-Users mailing list [EMAIL PROTECTED]
RE: [Asterisk-Users] Sip Register Fail - NAT
I was able to resolve the issue... Me being stupid... Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AstGrp Posted At: Sunday, February 22, 2004 7:45 PM Posted To: Asterisk User Group Conversation: Sip Register Fail - NAT Subject: [Asterisk-Users] Sip Register Fail - NAT I am having an issue with registering SIP client w/ NAT. I have set this up before on other boxes... But for some reason this one is not acting the same... I have attached a sip debug from the registration... For what ever reason it does not appear to be setting up the nat session correctly Am I seeing something wrong or even doing something wrong -gcc SIP CONFIG ## ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = nat ip ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 10.100.254.0; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context=default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc allow=alaw [4003] type=friend username=4003 secret=4003 host=dynamic qualify=500 context=local nat=yes mailbox=4003 ## SIP DEBUG #3 Sip read: REGISTER sip:10.100.254.21 SIP/2.0 Content-Length: 0 Contact: sip:192.168.1.10 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=10990022 CSeq: 87 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.10:5060 8 headers, 0 lines Using latest request as basis request Sending to 192.168.1.10 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990022 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 87 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 69.132.68.17:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990022 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 87 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=267e89bd Content-Length: 0 to 69.132.68.17:5060 ^Dtnevoip*CLI Sip read: REGISTER sip:10.100.254.21 SIP/2.0 Content-Length: 0 Contact: sip:192.168.1.10 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=10990413 CSeq: 88 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.10:5060 Proxy-Authorization: Digest username=4003,realm=asterisk,nonce=267e89bd,uri=sip:10.100.254.21 ,response=fb30e53fffc30ea15fc97acf7d82322f 9 headers, 0 lines Using latest request as basis request Sending to 192.168.1.10 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990413 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 88 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 69.132.68.17:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990413 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 88 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 69.132.68.17:5060 Feb 22 19:33:23 [1;33;40mNOTICE[0;37;40m[-1147384912]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m5577[0;37;40m [1;37;40mhandle_request[0;37;40m: Registration from 'sip:[EMAIL PROTECTED]' failed for '69.132.68.17' ^Dtnevoip*CLI Sip read: REGISTER sip:10.100.254.21 SIP/2.0 Content-Length: 0 Contact: sip:192.168.1.10 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=10990613 CSeq: 89 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.10:5060 8 headers, 0 lines Using latest request as basis request Sending to 192.168.1.10 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip
RE: [Asterisk-Users] Call Redirection
Do you have any examples of this? -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew B Marlowe Posted At: Saturday, February 21, 2004 7:34 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Call Redirection Subject: RE: [Asterisk-Users] Call Redirection What I have is it rings all of our desk phones obviously in the office for however long. Afterwards it places the call in a queue that doesn't retry on timeout and all of the people with cell phones are logged in via AgentCallBackLogin. I place the call in the same queue 3 times in a row (hence, calling all of the cell phones at the same time for ~ 15 seconds and hanging up and trying again. Reason for this is because within 15 seconds if no one picked up the cell phone is probably out of service or that person isn't taking the call) The person being called is required to dial a # to accept the call and they can hit a * to reject the call. Works extremely well. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FastJack Sent: Saturday, February 21, 2004 7:25 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call Redirection hi, I think it should be even great, to have an ack-password so if the phone is answered by someone unexpected (e.g. your wife!!) the person CAN_NOT answer this call! any thoughts? - Original Message - From: AstGrp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, February 21, 2004 7:09 AM Subject: [Asterisk-Users] Call Redirection I have a question regarding call redirection. Example call comes in to a extension. No one answers then call gets redirected out to cell phone. I need to implement something like for our tech support line. Call rings multiple extensions then if no one answers it gets forwarded out to a cell. I have tried the following : [FWD] exten = s,1,Dial(Zap/g2/7041234567) ;Tech Support exten = 4200,1,Dial,SIP/gclarkSIP/kelworth|15 exten = 4200,2,Goto(FWD,s,1) But everytime I try this - the phone that is generating the call receives another call from the pbx. It appears that the call is going out but, calls back the same user who is making the call. Thanks, -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiceMail
To add to the comments... You can apply the following patch for advanced features that you are looking for http://bugs.digium.com/bug_view_page.php?bug_id=156 -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Posted At: Saturday, February 21, 2004 4:34 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] VoiceMail Subject: Re: [Asterisk-Users] VoiceMail On Saturday 21 February 2004 04:24, Chad Brown wrote: Is something different about the latest implementation of voicemail in asterisk? Yes, it's been rewritten to allow multiple organizations within the same Asterisk system. There are also multiple other features that you can check out by looking at the ChangeLog. 1. I can leave but cannot retrieve my voicemails. I call my extension number but cant find a combination that allows me to break into an admin menu for retrieval. Are you dialling an extension which calls VoiceMailMain() ? 2. I could be dreaming but I thought I remember more advanced features available after leaving a VM. (Review, delete, etc.) I could be remembering incorrectly as well. Nope. You're welcome to write and contribute that addition, though. -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 ___ 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] multiple lines on 7960's
Not if I understand but with the 7960 you can have one exten for the primary line and then have the other 5 softkeys register different sip extensions. Then just choose that extension and dial out. We do this for people you share 1 phone. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Netlabz - Chris Clifton Posted At: Friday, February 20, 2004 2:39 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] multiple lines on 7960's Subject: [Asterisk-Users] multiple lines on 7960's I'm assuming this works on the 7960's with * from looking at the wiki and reading other posts. (user has primary ext. for themselves, but can pick up and dial multiple other lines on the 7960, place these lines on hold, transfer, etc.) Can someone verify ? How does this look in extensions.conf ? Thanks, Chris Clifton ___ 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] Call Redirection
I have a question regarding call redirection. Example call comes in to a extension. No one answers then call gets redirected out to cell phone. I need to implement something like for our tech support line. Call rings multiple extensions then if no one answers it gets forwarded out to a cell. I have tried the following : [FWD] exten = s,1,Dial(Zap/g2/7041234567) ;Tech Support exten = 4200,1,Dial,SIP/gclarkSIP/kelworth|15 exten = 4200,2,Goto(FWD,s,1) But everytime I try this - the phone that is generating the call receives another call from the pbx. It appears that the call is going out but, calls back the same user who is making the call. Thanks, -gcc ___ 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] Callerid AGI Thougts
I have a system put together with 3 Companies behind it and want the
Receptionist to Answer the phone Accordingly. Here are my thoughts.
DID Number - 704-123-0031
Extension.conf
[main]
exten = 0031,1,Goto(test,s,1)
[test]
exten = s,1,Answer
exten = s,2,agi,callid.agi
Exten = s,3,Dial(SIP/Recp)
AGI.conf - Brief Descrip.
$AGI-set_callerid('CompanyA')
The concept behind this is for the Receptionist to know who is being
called and not who is calling.. Would this work?
Thanks,
-gcc
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[Asterisk-Users] VoicemailMain2
I have question regarding recording the Unavail message. I remember in previous versions - when recording the message it played back the message you recorded and you had a chance to re-record or save it. Was this taken out and if so why? CVS-02/14/04-11:26:25 Thanks -gcc ___ 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] T1 Help
I have a question. We have been using Asterisk for a few months with POT's lines. And have just implemented a T1 Circuit. My problem is I can receive inbound calls but can't make any outbound calls. We have Cisco 7940G phones. You will find my config below - if you can find anything I am doing wrong please let me know. -gcc Zapata.conf [channels] context=default group=1 signalling=featd musiconhold=default immediate=no channel = 1-6 zaptel.conf loadzone=us defaultzone=us span=1,0,0,esf,b8zs em=1-6 sip.conf ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to ;externip = 200.201.202.203 ; Address that we're going to put in SIP messages if we're behind a NAT ;localnet = 192.168.1.0 ; Internal NETWORK address ;localmask = 255.255.255.0 ; Internal netmask context=default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc allow=alaw [tgreene] type=friend username=tgreene fromuser=Todd Greene secret=dickslap host=dynamic canreinvite=no mailbox=3001 [rnewton] type=friend username=rnewton fromuser=Randy Newton secret=dickslap host=dynamic canreinvite=no mailbox=3002 ___ 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] Music on Hold - Context
I was thinking about that... But here is my problem. We have 6 DID lines. We have it set up that all three companies share all lines.. Based off of the DNIS states what AutoAttendant they hit. So if I were to specify what channels the played certain MOH. Then that would mean Company 1 would have to come over on Channels 1-2 and so on. Any other thoughts. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matteo Brancaleoni Posted At: Sunday, February 15, 2004 8:27 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Music on Hold - Context Subject: Re: [Asterisk-Users] Music on Hold - Context hi I have set up a * box supporting 3 different companies but have some questions regarding MOH. Can MOH support multiple context or classes. Reason I ask each company would like to have different MOH sound files. Is this possible? yes, just specify multiple moh classes in musiconhold.conf and use each moh class for each company. example: company1 = mp3:/var/lib/asterisk/somemoh1 company2 = mp3:/var/lib/asterisk/somemoh2 company3 = mp3:/var/lib/asterisk/somemoh3 and now assign each moh class on your users/ivr/channels... 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 ___ 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] Music on Hold - Context
I have set up a * box supporting 3 different companies but have some questions regarding MOH. Can MOH support multiple context or classes. Reason I ask each company would like to have different MOH sound files. Is this possible? Thanks, -gcc ___ 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] CVS Changes (NAT-SIP)
I was able to resolve this problem, after removing and adding back the port settings in the firewall. I changed hardware and IP's. So I can only guess that arp table was messed up. I'm sure rebooting the firewall would have given me the same result. But everything has been working fine since then. Not sure if this helps. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Flagg Posted At: Friday, February 06, 2004 1:27 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] CVS Changes (NAT-SIP) Subject: Re: [Asterisk-Users] CVS Changes (NAT-SIP) I am having the same problem with a new CVS. Patrick also has the problem here http://lists.digium.com/pipermail/asterisk-users/2004-January/035114.htm l Keven had a problem here http://lists.digium.com/pipermail/asterisk-users/2004-January/035262.htm l but was able to get it fixed. Can you post a patch?. My asterisk computer is multi-homed behind NAT so maybe that is a factor? Is Asterisk behind NAT working with a new CVS for anybody? Thanks, - Original Message - From: Asterisk User Group [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 19, 2004 10:16 PM Subject: [Asterisk-Users] CVS Changes (NAT-SIP) I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc [1001] type=friend secret=1001 host=dynamic username=1001 mailbox=1001 context=local nat=no [1006] type=friend secret=oicu812 host=dynamic username=1006 mailbox=1006 context=local nat=yes canreinvite=no qualify=500 Internal SIP users can register it just the outside users. -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AutoAttendent ON/OFF control by Attendent
I know this is fairly old thread, but I have a question regarding this.
The following line:
exten = s,2,GotoIf($[${autoattendant} = 1]?auto|1)
Is basically saying goto context priority 1. So the last line also has
a goto to statement. When is this being trigered. So could you use the
same line but instead say:
exten = s,2,GotoIf($[${autoattendant} = 1]?4:3)
Just curious
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Posted At: Monday, December 15, 2003 12:45 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] AutoAttendent ON/OFF control by Attendent
Subject: Re: [Asterisk-Users] AutoAttendent ON/OFF control by Attendent
On Monday 15 December 2003 10:57, Sri wrote:
Hi All
This is one scenario I would like to have some help. I have searched
the digium lists and could not find any posts on this.
How can an Attendant switch on or off the AutoAttendant from her
phone? Eg. 8am - Attendent enters office - switches OFF auto
attendent. He/She takes in all the incoming calls and answers.
12pm - out of lunch. Needs to put the system back into Auto.
1 pm - return from lunch. Needs to switch OFF auto attendent
5 pm- Puts Auto attendent ON.
I am sure there can be a script built that should change
extensions.conf. and reloading asterisk on the attendent activating
based on a clock that kicks in 8 am, 12 pm, 1 pm and 5 pm. I dont want
this to be time restricted. the attendent should have control. Is
there a better way ? this could be even done through the phone of the
attendent eg, like *80-1 (ON) *80 - 2 (OFF)...
exten = *801,1,DBPut(auto/attendant=1)
exten = *802,1,DBPut(auto/attendant=0)
exten = s,1,DBGet(autoattendant=auto/attendant)
exten = s,2,GotoIf($[${autoattendant} = 1]?auto|1)
exten = s,3,Dial(Zap/23,30,t)
exten = s,4,Goto(auto|1)
-Tilghman
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RE: [Asterisk-Users] ZAP Problems
I would say it might be this... n zapata.conf language=en contect=default - should be context=default switchtype-euroisdn signaling=fxs_ks rxwink=300 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Posted At: Monday, January 26, 2004 6:12 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] ZAP Problems Subject: [Asterisk-Users] ZAP Problems Hi all, Since my upgrade to CVS dated 14-01-2004 I am unable to call or receive calls through my ZAP channel. When calling out I get the following message: - WARNING [155667]:app_dial.c:527 dial_exec: Unable to create channel of type ZAP In zaptel.conf fxsks=1 loadzone=uk defaultzone=uk In zapata.conf language=en contect=default switchtype-euroisdn signaling=fxs_ks rxwink=300 I have done: - modprobe zaptel modprobe wcfxo ztcfg -vv results: - Zaptel Configuration Channel Map: Channel 01: FXS Kewlstart (Default) (Salves:01) 1 Channels configured Any help to resolve would be appreciated. Regards Dave ___ 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] CVS Changes (NAT-SIP)
It is not working. Need HELP -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Posted At: Tuesday, January 20, 2004 1:08 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] CVS Changes (NAT-SIP) Subject: Re: [Asterisk-Users] CVS Changes (NAT-SIP) Can you clarify this? Does it or doesn't it work? bkw On Mon, 19 Jan 2004, Asterisk User Group wrote: I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc [1001] type=friend secret=1001 host=dynamic username=1001 mailbox=1001 context=local nat=no [1006] type=friend secret=oicu812 host=dynamic username=1006 mailbox=1006 context=local nat=yes canreinvite=no qualify=500 Internal SIP users can register it just the outside users. -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Enter Pin followed by Pound key
This has worked for me
my $empid = $AGI-get_data('employee',-1,5);
It is set to accept 5 digits but hitting pound before the fifth digit
works to satisfy.
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
Franczyk
Posted At: Tuesday, January 20, 2004 11:52 AM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Enter Pin followed by Pound key
Subject: [Asterisk-Users] Enter Pin followed by Pound key
Im trying to create a custom application via the AGI. I want to
authenticate the users that dial in with a userid and pin. However, the
number of digits in the PIN and userid are variable, and therefore I
need to allow the user to press enter by hitting the pound key. How
would I accomplish this in the AGI?
stream_file doesnt seem to work, since it only allows one digit to be
pressed. get_data seems to only allow a fixed number of digits to be
entered.
Thanks
Gary Franczyk
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