[asterisk-users] asterisk-users Digest, Vol 81, Issue 27
darin iv
adariniv at gmail.com
Sat Apr 9 03:31:17 CDT 2011
I need to change the sip port from 5060 to 5061 actually we already
used 5060 for proxy to sip any idea to change 5060 to 5061 so all can
acces the sip using this port please help........................
On 4/8/11, asterisk-users-request at lists.digium.com
<asterisk-users-request at lists.digium.com> wrote:
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> Today's Topics:
>
> 1. Re: IAX2/0.0.29.199 (satish patel)
> 2. Re: Variable inheritance with dialplan command Originate
> (Naomi Rosenberg)
> 3. Re: CRC Zaptel.conf (Shaun Ruffell)
> 4. Re: Variable inheritance with dialplan command Originate
> (Jim Dickenson)
> 5. Re: Variable inheritance with dialplan command Originate
> (Sherwood McGowan)
> 6. Re: Variable inheritance with dialplan command Originate
> (Sherwood McGowan)
> 7. Re: IAX2/0.0.29.199 (satish patel)
> 8. Re: asterisk login to voicemail (vip killa)
> 9. Re: asterisk login to voicemail (satish patel)
> 10. Re: IAX2/0.0.29.199 (Paul Belanger)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Fri, 8 Apr 2011 15:55:39 +0000
> From: satish patel <satish_lx at hotmail.com>
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> To: asterisk-users <asterisk-users at lists.digium.com>
> Message-ID: <BLU159-w50125FF32BB6ED7ACBCBC390A70 at phx.gbl>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
> @Paul - many time i am gettting following SIP error when channel isn't
> available. I want to get rid on this revers thing. I tried all version
> 1.8.1,1.8.2,1.8.3 but not fix :(
>
>
> [Apr 8 11:52:22] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> 0x2f40580 (len 793) to 0.0.29.200:5060 returned -1: Invalid argument
> [Apr 8 11:52:26] NOTICE[13912]: chan_iax2.c:4643 __auto_congest:
> Auto-congesting call due to slow response
>
> -Satish
>
>> Date: Fri, 8 Apr 2011 11:12:59 -0400
>> From: pabelanger at digium.com
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] IAX2/0.0.29.199
>>
>> On 11-04-08 10:48 AM, satish patel wrote:
>> >
>> > Where this revers IP comes from ?
>> >
>> > == Using SIP RTP CoS mark 5
>> > -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000006b",
>> > "stdexten,7623,SIP/7623") in new stack
>> > -- Executing [s at macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b",
>> > "SIP/7623&IAX2/7623,20,t") in new stack
>> > -- Hungup 'IAX2/0.0.29.199:4569-5255'
>> > -- Executing [s at macro-stdexten:2] NoOp("SIP/7527-0000006b",
>> > "IAX2/0.0.29.199:4569-5255") in new stack
>> > -- Executing [s at macro-stdexten:3] NoOp("SIP/7527-0000006b", "0&0")
>> > in new stack
>> > -- Auto fallthrough, channel 'SIP/7527-0000006b' status is
>> > 'UNKNOWN'
>> >
>> Asterisk 1.8? Are you using realtime? Looks to be an issue with
>> netsock2.c.
>>
>> --
>> Paul Belanger
>> Digium, Inc. | Software Developer
>> twitter: pabelanger | IRC: pabelanger (Freenode)
>> Check us out at: http://digium.com & http://asterisk.org
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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> ------------------------------
>
> Message: 2
> Date: Fri, 8 Apr 2011 16:57:27 +0100 (BST)
> From: Naomi Rosenberg <naomi at dmcip.com>
> Subject: Re: [asterisk-users] Variable inheritance with dialplan
> command Originate
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <28369896.3418.1302278246954.JavaMail.root at pomona>
> Content-Type: text/plain; charset=utf-8
>
> Thanks. That's as I thought (feared). Dial is not an option in this case but
> I have come up with a workaround involving using a reference number as the
> extension and then doing a database call. Not pretty but it works!
>
> Naomi
> ----- Original Message -----
> From: "Sherwood McGowan" <sherwood.mcgowan at gmail.com>
> To: asterisk-users at lists.digium.com
> Sent: Friday, 8 April, 2011 4:35:43 PM
> Subject: Re: [asterisk-users] Variable inheritance with dialplan command
> Originate
>
> On 4/8/2011 4:57 AM, Naomi Rosenberg wrote:
>> Hi,
>>
>> I would have thought that when spawning a channel using the
>> Originate() dialplan command, variables prefixed with two underscores
>> would be preserved.
>>
>> However this does not work in the following case.
>>
>> Dialplan code:
>>
>> [intern]
>> exten => 200,1,Set(__myvar="foo")
>> exten => 200,n,Originate(Local/123 at test_orig,exten,dummy)
>>
>> [test_orig]
>> exten => 123,1,NoOp(${myvar})
>> exten => 123,n,Hangup()
>>
>> [dummy]
>>
>> /end dialplan code.
>>
>> Console output:
>>
>> -- Executing [200 at intern:1] Set("SIP/200-00000018",
>> "__myvar="foo"") in new stack
>> -- Executing [200 at intern:2] Originate("SIP/200-00000018",
>> "Local/123 at test_orig,exten,dummy") in new stack
>> -- Executing [123 at test_orig:1] NoOp("Local/123 at test_orig-cbab;2",
>> "") in new stack
>> -- Executing [123 at test_orig:2]
>> Hangup("Local/123 at test_orig-cbab;2", "") in new stack
>>
>>
>> /end console output.
>>
>> This is in Asterisk 1.8.3.
>>
>> Is this expected behaviour or a bug, or am I just confused? I would
>> appreciate your thoughts on the matter.
>>
>> Thank you,
>>
>> Naomi
>
> I believe that it's expected behavior because you're not creating a
> "child" channel, you're originating a different set. Try using Dial
> instead of Originate, and you'll get the inheritance behavior you
> expected.
>
> -- Sherwood McGowan <sherwood.mcgowan at gmail.com>
> Carrier, ITSP, Call Center, and PBX Solutions Consultant
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> ------------------------------
>
> Message: 3
> Date: Fri, 8 Apr 2011 10:59:45 -0500
> From: Shaun Ruffell <sruffell at digium.com>
> Subject: Re: [asterisk-users] CRC Zaptel.conf
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <20110408155945.GA24160 at digium.com>
> Content-Type: text/plain; charset=us-ascii
>
> On Fri, Apr 08, 2011 at 09:11:20AM +0000, salaheddine elharit wrote:
>> i have a question related to CRC, yesterday i had an issue in our span
>> when
>> i verify with zttool i found recovering instead ok
>>
>> i verify the zaptel.conf and i found
>>
>> # Autogenerated by /usr/sbin/zapconf on Thu Apr 7 17:19:52 2011 -- do not
>> hand edit
>> # Zaptel Configuration File
>> #
>> # This file is parsed by the Zaptel Configurator, ztcfg
>> #
>>
>> # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS
>> span=1,1,0,ccs,hdb3,*crc4*
>> # termtype: te
>> bchan=1-15,17-31
>> dchan=16
>>
>
> ...
>
>> in the normal zaptel there is no *crc4 * i delete the CRC4 from the 2
>> spans
>> and all works perfectly.
>>
>> i want to know how the file was generated automatically with CRC4 and if
>> there is any way to disable this option
>
> I am not familiar with how the /usr/sbin/zapconf tools works, but with
> dahdi_genconf in DAHDI, E1 spans default to use CRC4. This covers the
> majority of users.
>
> I did not see any way in /etc/dahdi/genconf_parameters to control the
> output of the CRC4 line.
>
> My opinion is that you did exactly what you needed: run zapconf to get
> sensible defaults then edit the output file for your specific provider.
>
> Cheers,
> Shaun
>
> --
> Shaun Ruffell
> Digium, Inc. | Linux Kernel Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org
>
>
>
> ------------------------------
>
> Message: 4
> Date: Fri, 8 Apr 2011 09:05:00 -0700
> From: Jim Dickenson <dickenson at cfmc.com>
> Subject: Re: [asterisk-users] Variable inheritance with dialplan
> command Originate
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <C5D367A3-9C44-4FE2-9576-E652C413B040 at cfmc.com>
> Content-Type: text/plain; charset=us-ascii
>
> Another option is to pass the information in the extension. At times I have
> an extension like
>
> _[s][o][m][e]-[e][x][a][m][p][l][e].
>
> And call it like some-example:info1:info2 and use cut to extract the info1
> and info2 values. Not real pretty but as this is computer generated calls it
> gets the job done.
> --
> Jim Dickenson
> mailto:dickenson at cfmc.com
>
> CfMC
> http://www.cfmc.com/
>
>
>
> On Apr 8, 2011, at 8:57 AM, Naomi Rosenberg wrote:
>
>> Thanks. That's as I thought (feared). Dial is not an option in this case
>> but I have come up with a workaround involving using a reference number as
>> the extension and then doing a database call. Not pretty but it works!
>>
>> Naomi
>> ----- Original Message -----
>> From: "Sherwood McGowan" <sherwood.mcgowan at gmail.com>
>> To: asterisk-users at lists.digium.com
>> Sent: Friday, 8 April, 2011 4:35:43 PM
>> Subject: Re: [asterisk-users] Variable inheritance with dialplan command
>> Originate
>>
>> On 4/8/2011 4:57 AM, Naomi Rosenberg wrote:
>>> Hi,
>>>
>>> I would have thought that when spawning a channel using the
>>> Originate() dialplan command, variables prefixed with two underscores
>>> would be preserved.
>>>
>>> However this does not work in the following case.
>>>
>>> Dialplan code:
>>>
>>> [intern]
>>> exten => 200,1,Set(__myvar="foo")
>>> exten => 200,n,Originate(Local/123 at test_orig,exten,dummy)
>>>
>>> [test_orig]
>>> exten => 123,1,NoOp(${myvar})
>>> exten => 123,n,Hangup()
>>>
>>> [dummy]
>>>
>>> /end dialplan code.
>>>
>>> Console output:
>>>
>>> -- Executing [200 at intern:1] Set("SIP/200-00000018",
>>> "__myvar="foo"") in new stack
>>> -- Executing [200 at intern:2] Originate("SIP/200-00000018",
>>> "Local/123 at test_orig,exten,dummy") in new stack
>>> -- Executing [123 at test_orig:1] NoOp("Local/123 at test_orig-cbab;2",
>>> "") in new stack
>>> -- Executing [123 at test_orig:2]
>>> Hangup("Local/123 at test_orig-cbab;2", "") in new stack
>>>
>>>
>>> /end console output.
>>>
>>> This is in Asterisk 1.8.3.
>>>
>>> Is this expected behaviour or a bug, or am I just confused? I would
>>> appreciate your thoughts on the matter.
>>>
>>> Thank you,
>>>
>>> Naomi
>>
>> I believe that it's expected behavior because you're not creating a
>> "child" channel, you're originating a different set. Try using Dial
>> instead of Originate, and you'll get the inheritance behavior you
>> expected.
>>
>> -- Sherwood McGowan <sherwood.mcgowan at gmail.com>
>> Carrier, ITSP, Call Center, and PBX Solutions Consultant
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> ------------------------------
>
> Message: 5
> Date: Fri, 08 Apr 2011 11:05:38 -0500
> From: Sherwood McGowan <sherwood.mcgowan at gmail.com>
> Subject: Re: [asterisk-users] Variable inheritance with dialplan
> command Originate
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4D9F3252.4080303 at gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On 4/8/2011 10:57 AM, Naomi Rosenberg wrote:
>> Thanks. That's as I thought (feared). Dial is not an option in this case
>> but I have come up with a workaround involving using a reference number as
>> the extension and then doing a database call. Not pretty but it works!
>>
>> Naomi
>
> I'm not sure why Dial wouldn't work...I use Dial all the time for
> triggering Local channels that perform database calls all the time
>
> --
> Sherwood McGowan <sherwood.mcgowan at gmail.com>
> Carrier, ITSP, Call Center, and PBX Solutions Consultant
>
>
>
>
> ------------------------------
>
> Message: 6
> Date: Fri, 08 Apr 2011 11:10:55 -0500
> From: Sherwood McGowan <sherwood.mcgowan at gmail.com>
> Subject: Re: [asterisk-users] Variable inheritance with dialplan
> command Originate
> To: asterisk-users at lists.digium.com
> Message-ID: <4D9F338F.1040408 at gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On 4/8/2011 11:05 AM, Jim Dickenson wrote:
>> Another option is to pass the information in the extension. At times I
>> have an extension like
>>
>> _[s][o][m][e]-[e][x][a][m][p][l][e].
>>
>> And call it like some-example:info1:info2 and use cut to extract the info1
>> and info2 values. Not real pretty but as this is computer generated calls
>> it gets the job done.
>
> Still not sure why you guys need this...Here's my example
>
> [firstleg]
> exten => 200,1,Set(__myvar=foo) ; Don't forget you don't want quotes!)
> exten => 200,n,Dial(Local/123 at test_orig)
> [test_orig]
> exten => 123,1,Noop(${myvar})
> same => n,Set(dbtest=${ODBC_TESTQUERY(myvar)})
>
> --
> Sherwood McGowan <sherwood.mcgowan at gmail.com>
> Carrier, ITSP, Call Center, and PBX Solutions Consultant
>
>
>
>
> ------------------------------
>
> Message: 7
> Date: Fri, 8 Apr 2011 16:24:23 +0000
> From: satish patel <satish_lx at hotmail.com>
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> To: asterisk-users <asterisk-users at lists.digium.com>
> Message-ID: <BLU159-w54051586E707B5A3DD9BF590A70 at phx.gbl>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
>
> Look at this sip debug its saying something related Retransmitting #1 (no
> NAT) to 0.0.29.200:5060:
>
> <------------>
> -- Executing [7624 at from-sip:1] Macro("SIP/7527-000000c2",
> "stdexten,7624,SIP/7624") in new stack
> -- Executing [s at macro-stdexten:1] Dial("SIP/7527-000000c2",
> "SIP/7624&IAX2/7624,20,t") in new stack
> == Using SIP RTP CoS mark 5
> [Apr 8 12:20:53] WARNING[15194]: acl.c:698 ast_ouraddrfor: Cannot connect
> Audio is at 5060
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 0.0.29.200:5060:
> INVITE sip:7624 SIP/2.0
> Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3af914e2
> Max-Forwards: 70
> From: "Cambridge Guest" <sip:7527 at 172.30.1.47>;tag=as6f6822ba
> To: <sip:7624>
> Contact: <sip:7527 at 172.30.1.47:5060>
> Call-ID: 0ca7784d38d29be168f8f85711c43e4f at 172.30.1.47:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.8.3.2
> Date: Fri, 08 Apr 2011 19:20:53 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 257
>
> v=0
> o=root 1407056235 1407056235 IN IP4 172.30.1.47
> s=Asterisk PBX 1.8.3.2
> c=IN IP4 172.30.1.47
> t=0 0
> m=audio 16720 RTP/AVP 0 3 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> ---
> [Apr 8 12:20:53] WARNING[15194]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> 0x2ef3f00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
> -- Called 7624
> -- Called 7624
> Retransmitting #1 (no NAT) to 0.0.29.200:5060:
> INVITE sip:7624 SIP/2.0
> Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3af914e2
> Max-Forwards: 70
> From: "Cambridge Guest" <sip:7527 at 172.30.1.47>;tag=as6f6822ba
> To: <sip:7624>
> Contact: <sip:7527 at 172.30.1.47:5060>
> Call-ID: 0ca7784d38d29be168f8f85711c43e4f at 172.30.1.47:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.8.3.2
> Date: Fri, 08 Apr 2011 19:20:53 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 257
>
>
>
>
>> Date: Fri, 8 Apr 2011 11:12:59 -0400
>> From: pabelanger at digium.com
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] IAX2/0.0.29.199
>>
>> On 11-04-08 10:48 AM, satish patel wrote:
>> >
>> > Where this revers IP comes from ?
>> >
>> > == Using SIP RTP CoS mark 5
>> > -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000006b",
>> > "stdexten,7623,SIP/7623") in new stack
>> > -- Executing [s at macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b",
>> > "SIP/7623&IAX2/7623,20,t") in new stack
>> > -- Hungup 'IAX2/0.0.29.199:4569-5255'
>> > -- Executing [s at macro-stdexten:2] NoOp("SIP/7527-0000006b",
>> > "IAX2/0.0.29.199:4569-5255") in new stack
>> > -- Executing [s at macro-stdexten:3] NoOp("SIP/7527-0000006b", "0&0")
>> > in new stack
>> > -- Auto fallthrough, channel 'SIP/7527-0000006b' status is
>> > 'UNKNOWN'
>> >
>> Asterisk 1.8? Are you using realtime? Looks to be an issue with
>> netsock2.c.
>>
>> --
>> Paul Belanger
>> Digium, Inc. | Software Developer
>> twitter: pabelanger | IRC: pabelanger (Freenode)
>> Check us out at: http://digium.com & http://asterisk.org
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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> ------------------------------
>
> Message: 8
> Date: Fri, 8 Apr 2011 12:26:27 -0400
> From: vip killa <vipkilla at gmail.com>
> Subject: Re: [asterisk-users] asterisk login to voicemail
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <BANLkTikXudxp=W=K+B6TzyCDVC10bFD17Q at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> can you explain how this can be done simpler?
>
> On Fri, Apr 8, 2011 at 10:10 AM, Satish Patel <satish_lx at hotmail.com> wrote:
>
>> Why are you using agi for this ? They are inbuild features of asterisk.
>>
>> Or may be I am missing something
>>
>> --
>> Sent from my iPhone
>>
>> On Apr 8, 2011, at 8:26 AM, vip killa <vipkilla at gmail.com> wrote:
>>
>> Wow, thanks, that worked...
>> in case anyone is interested this is what i did....
>>
>> [voicemail]
>> exten => a,1,VoiceMailMain(${MAILBOXID}@${MAILBOXCONTEXT},p)
>>
>> in AGI...
>>
>> $AGI->set_variable("MAILBOXID", $options);
>> $AGI->set_variable("MAILBOXCONTEXT","4");
>> $AGI->set_context("voicemail");
>> $AGI->exec("VoiceMail", $options);
>>
>> now the question is how to I get the VoiceMailMain to not ask for
>> "Mailbox"
>> and already know which mailbox and just prompt for "Password"
>>
>>
>> On Thu, Apr 7, 2011 at 6:44 PM, Dan Journo <<dan at keshercommunications.com>
>> dan at keshercommunications.com> wrote:
>>
>>> > Unfortunately, that solution will not work for me... The user must be
>>> able to hit * during the greeting of any voicemail and be prompted for
>>> the
>>> "Password" to that particular mailbox.... looks like i got a lot of
>>> programming to do to create a work around for this... thanks for your
>>> help...
>>>
>>> Forgive me if i'm wrong, but you guys seem to be over complicating
>>> things.
>>>
>>> Taken from: <http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail>
>>> http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
>>>
>>> during the prompt if the caller presses:
>>> '*' - the call jumps to extension 'a' in the current voicemail context.
>>> *Example:*
>>> Exten => a, 1, VoicemailMain(@default)
>>> Exten => a, 2, Hangup
>>>
>>> When using the star '*' it's important to note that the context you
>>> placed
>>> the application voicemail in is irrelevant, it's the context for the
>>> voicemail box that we're looking for in the dialplan for the jump to the
>>> 'a'
>>> extension.
>>>
>>> So this is what i do...
>>>
>>> Before passing the call to the voicemail app, i set ${MAILBOXCONTEXT} to
>>> the correct context, and i set ${MAILBOXID} to the mailbox name.
>>>
>>> Then, in extensions.conf, I added this:-
>>>
>>> [voicemail]
>>> exten => a,1,Playback(astcc-please-enter-your)
>>> exten => a,n,VoicemailMain(${MAILBOXID}@${MAILBOXCONTEXT})
>>>
>>> When the user presses *, they are passed to the 'a' extension above and
>>> into VoicemailMain.
>>>
>>> I'm sure you can turn this into AGI easily enough if needed.
>>>
>>>
>>>
>>> Dan Journo
>>>
>>> Kesher Communications (UK)
>>>
>>> Business Phone Systems <http://www.keshercommunications.com/> | Hosted
>>> PBX <http://www.keshercommunications.com/hostedpbx.html>
>>>
>>>
>>>
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by <http://www.api-digital.com>
>>> http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>> <http://www.asterisk.org/hello>
>>> http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
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>>> <http://lists.digium.com/mailman/listinfo/asterisk-users>
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> <http://www.asterisk.org/hello>
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>>
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>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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> ------------------------------
>
> Message: 9
> Date: Fri, 8 Apr 2011 16:33:12 +0000
> From: satish patel <satish_lx at hotmail.com>
> Subject: Re: [asterisk-users] asterisk login to voicemail
> To: asterisk-users <asterisk-users at lists.digium.com>
> Message-ID: <BLU159-w41A6239E58F8FB8D0F4D6C90A70 at phx.gbl>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
>
> I have this for same function.
>
> [voice-mail]
>
> ;VM for external users calling from PSTN prompt for mailbox number and pin
> exten => 8000,1,Answer()
> exten => 8000,n,Wait(1)
> exten => 8000,n,VoicemailMain(@default)
> exten => 8000,n,Hangup()
>
> ;VM for internal users only pin
> exten => 8500,1,Answer()
> exten => 8500,n,Wait(1)
> exten => 8500,n,VoicemailMain(${CALLERID(num):-4}@default)
> exten => 8500,n,Hangup()
>
> exten => i,1,playback(invalid)
> exten => i,2,hangup
>
>
>
> Date: Fri, 8 Apr 2011 12:26:27 -0400
> From: vipkilla at gmail.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] asterisk login to voicemail
>
> can you explain how this can be done simpler?
>
> On Fri, Apr 8, 2011 at 10:10 AM, Satish Patel <satish_lx at hotmail.com> wrote:
>
> Why are you using agi for this ? They are inbuild features of asterisk.
>
> Or may be I am missing something
> --Sent from my iPhone
> On Apr 8, 2011, at 8:26 AM, vip killa <vipkilla at gmail.com> wrote:
>
>
> Wow, thanks, that worked...in case anyone is interested this is what i
> did....
> [voicemail]exten => a,1,VoiceMailMain(${MAILBOXID}@${MAILBOXCONTEXT},p)
>
>
> in AGI...
> $AGI->set_variable("MAILBOXID",
> $options);$AGI->set_variable("MAILBOXCONTEXT","4");$AGI->set_context("voicemail");
>
> $AGI->exec("VoiceMail", $options);
> now the question is how to I get the VoiceMailMain to not ask for "Mailbox"
> and already know which mailbox and just prompt for "Password"
>
>
>
> On Thu, Apr 7, 2011 at 6:44 PM, Dan Journo <dan at keshercommunications.com>
> wrote:
>
>
>> Unfortunately, that solution will not work for me... The user must be able
>> to hit * during the greeting of any voicemail and be prompted for the
>> "Password" to that particular mailbox.... looks like i got a lot of
>> programming to do to create a work around for this... thanks for your
>> help...
>
> Forgive me if i'm wrong, but you guys seem to be over complicating things.
>
> Taken from: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
>
> during the prompt if the caller presses:
> '*' - the call jumps to extension 'a' in the current voicemail context.
> Example:
>
>
> Exten => a, 1, VoicemailMain(@default)
> Exten => a, 2, Hangup
>
> When using the star '*' it's important to note that the context you placed
> the application voicemail in is irrelevant, it's the context for the
> voicemail box that we're looking for in the dialplan for the jump to the 'a'
> extension.
>
>
>
> So this is what i do...
>
> Before passing the call to the voicemail app, i set ${MAILBOXCONTEXT} to the
> correct context, and i set ${MAILBOXID} to the mailbox name.
>
> Then, in extensions.conf, I added this:-[voicemail]
> exten => a,1,Playback(astcc-please-enter-your)
>
>
> exten => a,n,VoicemailMain(${MAILBOXID}@${MAILBOXCONTEXT})When the user
> presses *, they are passed to the 'a' extension above and into
> VoicemailMain.
>
> I'm sure you can turn this into AGI easily enough if needed.
>
> Dan JournoKesher Communications (UK)Business Phone Systems | Hosted PBX
>
>
> --
>
> _____________________________________________________________________
>
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>
> http://www.asterisk.org/hello
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>
>
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>
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
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> --
>
> _____________________________________________________________________
>
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>
> http://www.asterisk.org/hello
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>
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>
>
> --
> _____________________________________________________________________
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> ------------------------------
>
> Message: 10
> Date: Fri, 08 Apr 2011 12:56:15 -0400
> From: Paul Belanger <pabelanger at digium.com>
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> To: asterisk-users at lists.digium.com
> Message-ID: <4D9F3E2F.3020709 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> On 11-04-08 11:55 AM, satish patel wrote:
>>
>> @Paul - many time i am gettting following SIP error when channel isn't
>> available. I want to get rid on this revers thing. I tried all version
>> 1.8.1,1.8.2,1.8.3 but not fix :(
>>
> Best you can do is collect a full debug[1] log and see when the issue is
> introduced.
>
> [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
>
> --
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
>
>
>
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>
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