[asterisk-users] How to retrieve a phone number fromcall forwarding?
Soonthorn Ativanichayaphong
soonthorn at yapinc.com
Wed Jan 28 19:20:28 CST 2009
Thank you guys. I tried a tcpdump and couldn't find the information in SIP
INVITE. I don't even see CC-Diversion header. I'll talk to my SIP provider
(VoicePulse). Not really sure if they can add the header for me.
I have two more questions.
1. Let's say if there is "CC-Diversion" header field, how do I query the
header's value in Asterisk?
2. Do you know of any SIP provider (inexpensive if possible) that does
generate the header?
On Wed, Jan 28, 2009 at 8:04 PM, Don Kelly <dk at donkelly.biz> wrote:
> With ISDN service, DNIS presents the "DID" number, 222-222-2222 in the
> example--not what Soonthorn is looking for.
>
> 111-111-1111 is the "redirecting" number. This is available in an ISDN
> information element.
>
> For SIP, you'd apparently look for a "CC-Diversion header field." This is
> from a Cisco blurb:
>
> If generated by the SIP gateway during call process, the CC-Diversion
> header
> field is based on the contents of the Redirecting Number Information
> Element
> (IE) in the ISDN Setup message. In addition, information such as the reason
> the call was redirected is included in the CC-Diversion header field.
>
> --Don
>
> Don Kelly
> PCF Corp
> People Come First
>
> 651 842-1000
> 888 Don Kell(y)
> 651 842-1001 fax
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jose P.
> Espinal
> Sent: Wednesday, January 28, 2009 5:50 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to retrieve a phone number fromcall
> forwarding?
>
> Hello,
>
> Maybe what you are looking for is called DNIS (Dialed Number Information
> Service). Some companies provide this service, which you can use to
> route incoming calls to different dialplan options/contexts/etc.
>
>
>
> Regards,
>
>
>
> --
> Jose P. Espinal
> http://www.eSlackware.com
>
>
>
> Soonthorn Ativanichayaphong wrote:
> > Hi,
> >
> > I'm very new to Asterisk and I have the following scenario.
> >
> > 1. Let's say I have a number of 1-222-222-2222 from my SIP service
> > provider (VoicePulse).
> > 2. I point my phone, Verizon wireless cellphone (1-111-111-1111),
> > voicemail to the number provided by SIP service provider
> > (1-222-222-2222).
> > 3. I use another phone (1-333-333-333) to call 1-111-111-1111 and
> > leave a voicemail message.
> >
> > Within my Asterisk console , I can see a caller id of 1-333-333-333
> > and the number provided by SIP service provider (1-222-222-2222).
> > However, I couldn't figure out how to get the number the caller dialed
> > ( 1-111-111-1111). Is there a way to retrieve the number the caller
> > dialed (i.e. 1-111-1111) in this scenario?
> >
> > Note that as far as I know the carrier (e.g Verizon wireless) should
> > pass on those information. I see many companies that provide voicemail
> > to email services. They seem to be able to retrieve those information.
> > Is there a way to confirm that my SIP service provide does actually
> > pass on those information?
> >
> > Here is what I have in extensions.conf to test this scenario
> >
> > exten => _XX.,1,NoOp(Call received from VoicePulse)
> > exten => _XX.,n,Log(INFO|Caller ID Number: ${CALLERID(num)})
> > exten => _XX.,n,Answer()
> > exten => _XX.,n,DumpChan()
> > exten => _XX.,n,VoiceMail(101 at default,u)
> >
> > Here is what I see on the console.
> >
> > zeus*CLI>
> > -- Executing [12222222 at voicepulse-in:1] NoOp("SIP/mrXXXX-08XXXX",
> > "Call received from VoicePulse") in new stack
> > -- Executing [12222222 at voicepulse-in:2] Log("SIP/mrXXXX-08XXXX",
> > "INFO|Caller ID Number: 3333333") in new stack
> > [Jan 28 18:20:24] ERROR[22123]: app_verbose.c:133 log_exec: Unknown
> > log level: 'INFO'
> > -- Executing [12222222 at voicepulse-in:3]
> > Answer("SIP/mrXXXX-08XXXX", "") in new stack
> > -- Executing [12222222 at voicepulse-in:4]
> > DumpChan("SIP/mrXXXX-08XXXX", "") in new stack
> > zeus*CLI>
> > Dumping Info For Channel: SIP/mrXXXX-08XXXX:
> >
>
> ============================================================================
> ====
> > Info:
> > Name= SIP/mrXXXX-08XXXX
> > Type= SIP
> > UniqueID= 12331856824.83
> > CallerID= 3333333
> > CallerIDName= ATIVA DAVID
> > DNIDDigits= 12222222
> > RDNIS= (N/A)
> > State= Up (6)
> > Rings= 0
> > NativeFormat= 0x4 (ulaw)
> > WriteFormat= 0x4 (ulaw)
> > ReadFormat= 0x4 (ulaw)
> > 1stFileDescriptor= 23
> > Framesin= 0
> > Framesout= 0
> > TimetoHangup= 0
> > ElapsedTime= 0h0m0s
> > Context= voicepulse-in
> > Extension= 12222222
> > Priority= 4
> > CallGroup=
> > PickupGroup=
> > Application= DumpChan
> > Data= (Empty)
> > Blocking_in= (Not Blocking)
> >
> > Variables:
> > SIPCALLID=282e93ca78805a039fdf01729af52c at 64.62.94.171
> > <mailto:282e93ca78805a039fdf01729af52c at 64.62.94.171>
> > SIPUSERAGENT=Asterisk PBX
> > SIPDOMAIN=66.195.225.160
> > SIPURI=sip:3333333 at 64.62.94.171 <sip%3A3333333 at 64.62.94.171> <mailto:
> sip%3A3333333 at 64.62.94.171 <sip%253A3333333 at 64.62.94.171>>
> >
>
> ============================================================================
> ====
> > -- Executing [12222222 at voicepulse-in:5]
> > VoiceMail("SIP/mrXXXX-08XXXX", "101 at default|u") in new stack
> > -- <SIP/mrXXXX-08XXXX> Playing 'vm-theperson' (language 'en')
> > -- <SIP/mrXXXX-08XXXX> Playing 'digits/1' (language 'en')
> > -- <SIP/mrXXXX-08XXXX> Playing 'digits/0' (language 'en')
> > -- <SIP/mrXXXX-08XXXX> Playing 'digits/1' (language 'en')
> > -- <SIP/mrXXXX-08XXXX> Playing 'vm-isunavail' (language 'en')
> > -- <SIP/mrXXXX-08XXXX> Playing 'vm-intro' (language 'en')
> > -- <SIP/mrXXXX-08XXXX> Playing 'beep' (language 'en')
> > -- Recording the message
> > -- x=0, open writing:
> > /var/spool/asterisk/voicemail/default/101/tmp/0oxv2s format: wav49,
> > 0x830d4a0
> > -- x=1, open writing:
> > /var/spool/asterisk/voicemail/default/101/tmp/0oxv2s format: gsm,
> > 0x83082c0
> > -- x=2, open writing:
> > /var/spool/asterisk/voicemail/default/101/tmp/0oxv2s format: wav,
> > 0x82f0888
> > -- User hung up
> > == Spawn extension (voicepulse-in, 12222222, 5) exited non-zero on
> > 'SIP/mrXXXX-08XXXX'
> > zeus*CLI>
> >
> >
> > Here is what I see in a text file in
> > /var/spool/asterisk/voicemail/default/101/INBOX
> >
> > ;
> > ; Message Information file
> > ;
> > [message]
> > origmailbox=101
> > context=voicepulse-in
> > macrocontext=
> > exten=12222222
> > priority=5
> > callerchan=SIP/mrXXXX-08XXXX
> > callerid="ATIVA DAVID " <3333333>
> > origdate=Wed Jan 28 06:20:34 PM EST 2009
> > origtime=1233184834
> > category=
> > duration=6
> >
> >
> > Thank you. I really appreciate any help.
> >
> >
> >
> > ------------------------------------------------------------------------
> >
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--
Thanks,
Soonthorn Ativanichayaphong
Software Engineer
Yap Inc.
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