[Asterisk-Users] HelpwithAsterisk@homeandBroadvoiceincomingcalls..

Howard Leadmon howard at leadmon.net
Mon Jul 25 07:59:02 MST 2005


 Well you were right, and I don't understand why that exact config worked for
my iax config, but doesn't work for my sip config.   Maybe they don't permit
as many variables in the sip file, who knows.

I ended up using:

[frombroadvoice]
exten => _X.,1,Noop(Incoming call for extension ${EXTEN}in context
frombroadvoice)
exten => 201,1,Macro(exten-vm,${BVEXT}@default,200)
exten => ${VM_PREFIX}${BVMBOX},1,Macro(vm,${BVMBOX})

Note the 201 and 200 entries, as if I tried to use a variable in either of of
the two locations, the it would not work, and I verified spelling, and even
tried a different variable name.

Anyway it's working, I can get incoming calls on the BV line which is super!

Now the only thing I am lost on, heck maybe you will have an idea, is Vmail,
guess I should probably post as a separate message.   Anyway if an incoming
call goes to an active/working extension, and the user doesn't answer it, it
never transfers to vMail.   If it's busy/in-use, or disconnected from the
network it's fine and vMail picks up.     

Well I am getting closer!



---
Howard Leadmon - howard at leadmon.net
http://www.leadmon.net 


> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of dbruce
> Sent: Sunday, July 24, 2005 10:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users]
> HelpwithAsterisk at homeandBroadvoiceincomingcalls..
> 
> Your BVRINGS variable is being evaluated to a null string... check your
> spelling in the macro call.
> 
> Regards,
> Derek
> 
> ----- Original Message -----
> From: "Howard Leadmon" <howard at leadmon.net>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Sent: Sunday, July 24, 2005 6:00 PM
> Subject: RE: [Asterisk-Users]
> HelpwithAsterisk at homeandBroadvoiceincomingcalls..
> 
> 
> >
> >  OK, I removed that line as you said below, and now when I call in on the
> BV
> > line, I see this as output:
> >
> >
> > Jul 24 19:55:41 DEBUG[1078]: Setting NAT on RTP to 0
> > Jul 24 19:55:41 DEBUG[1078]: Check for res for 2405243333
> > Jul 24 19:55:41 DEBUG[1078]: 2405243333 is not a local user
> > Jul 24 19:55:41 DEBUG[1078]: build_route: Contact hop:
> > <sip:4105156666 at 147.135.0.128:5060;ep=147.135.0.129;transport=udp>
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> Macro("SIP/2405243333-7457",
> > "exten-vm|@default|") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> SetVar("SIP/2405243333-7457",
> > "FROMCONTEXT=exten-vm") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> Macro("SIP/2405243333-7457",
> > "record-enable||IN") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> GotoIf("SIP/2405243333-7457",
> > "0 > 0?2:4") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Goto (macro-record-enable,s,4)
> > Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> GotoIf("SIP/2405243333-7457",
> > "0?5:8") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Goto (macro-record-enable,s,8)
> > Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> GotoIf("SIP/2405243333-7457",
> > "0?9:12") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Goto (macro-record-enable,s,12)
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> DBget("SIP/2405243333-7457",
> > "RecEnable=RECORD-IN/") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- DBget: varname=RecEnable,
> > family=RECORD-IN, key=
> > Jul 24 19:55:41 DEBUG[1078]: Unable to find key '' in family 'RECORD-IN'
> > Jul 24 19:55:41 VERBOSE[1078]:     -- DBget: Value not found in database.
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> SetVar("SIP/2405243333-7457",
> > "CALLFILENAME=20050724-195541-1122249341.21") in new stack
> > Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> GotoIf("SIP/2405243333-7457",
> > "0?15:99") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Goto (macro-record-enable,s,99)
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> NoOp("SIP/2405243333-7457",
> > "NO RECORDING NEEDED") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> Macro("SIP/2405243333-7457",
> > "dial|||") in new stack
> > Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> GotoIf("SIP/2405243333-7457",
> > "0?4:2") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Goto (macro-dial,s,2)
> > Jul 24 19:55:41 WARNING[1078]: ast_yyerror(): syntax error: syntax error;
> > Input:
> >  !=
> >
> >     ^
> > Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> GotoIf("SIP/2405243333-7457",
> > "0?4:3") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Goto (macro-dial,s,3)
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> > SetCIDName("SIP/2405243333-7457", "Fork MD") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing AGI("SIP/2405243333-7457",
> > "dialparties.agi") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Launched AGI Script
> > /var/lib/asterisk/agi-bin/dialparties.agi
> > Jul 24 19:55:41 VERBOSE[1078]:     --  dialparties.agi: request =
> > dialparties.agi
> > Jul 24 19:55:41 VERBOSE[1078]:     --  dialparties.agi: priority = 4
> > Jul 24 19:55:41 VERBOSE[1078]:     --  dialparties.agi: extension = s
> > Jul 24 19:55:41 VERBOSE[1078]:     --  dialparties.agi: language = en
> > Jul 24 19:55:41 VERBOSE[1078]:     --  dialparties.agi: accountcode =
> > Jul 24 19:55:41 VERBOSE[1078]:     --  dialparties.agi: uniqueid =
> > 1122249341.21
> > Jul 24 19:55:41 VERBOSE[1078]:     --  dialparties.agi: channel =
> > SIP/2405243333-7457
> > Jul 24 19:55:41 VERBOSE[1078]:   dialparties.agi: callerid = Fork
> > Jul 24 19:55:41 VERBOSE[1078]:     --  dialparties.agi: context =
> macro-dial
> > Jul 24 19:55:41 VERBOSE[1078]:     --  dialparties.agi: type = SIP
> > Jul 24 19:55:41 VERBOSE[1078]:     --  dialparties.agi: rdnis = unknown
> > Jul 24 19:55:41 VERBOSE[1078]:     --  dialparties.agi: enhanced = 0.0
> > Jul 24 19:55:41 VERBOSE[1078]:     --  dialparties.agi: dnid = 201
> > Jul 24 19:55:41 VERBOSE[1078]:   dialparties.agi: Caller ID name is 'Fork
> MD'
> > number is '4105156666'
> > Jul 24 19:55:41 DEBUG[1078]: Manager received command 'Login'
> > Jul 24 19:55:41 VERBOSE[1078]:   == Parsing '/etc/asterisk/manager.conf':
> Jul
> > 24 19:55:41 VERBOSE[1078]:   == Parsing '/etc/asterisk/manager.conf':
> Found
> > Jul 24 19:55:41 VERBOSE[1078]:   == Parsing
> > '/etc/asterisk/manager_custom.conf': Jul 24 19:55:41 VERBOSE[1078]:   ==
> > Parsing '/etc/asterisk/manager_custom.conf': Found
> > Jul 24 19:55:41 DEBUG[1078]: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for
> peer
> > Jul 24 19:55:41 DEBUG[1078]: 127.0.0.1/255.255.255.0/255.255.255.0
> appended to
> > acl for peer
> > Jul 24 19:55:41 DEBUG[1078]: ##### Testing 127.0.0.1 with 0.0.0.0
> > Jul 24 19:55:41 DEBUG[1078]: ##### Testing 127.0.0.1 with 127.0.0.0
> > Jul 24 19:55:41 VERBOSE[1078]:   == Manager 'admin' logged on from
> 127.0.0.1
> > Jul 24 19:55:41 DEBUG[1078]: Manager received command 'command'
> > Jul 24 19:55:41 DEBUG[1078]: Manager received command ''
> > Jul 24 19:55:41 DEBUG[1078]: Manager received command 'Logoff'
> > Jul 24 19:55:41 VERBOSE[1078]:   == Manager 'admin' logged off from
> 127.0.0.1
> > Jul 24 19:55:41 VERBOSE[1078]:     -- AGI Script Executing Application:
> (NoOp)
> > Options: ()
> > Jul 24 19:55:41 VERBOSE[1078]:     -- AGI Script dialparties.agi
> completed,
> > returning 0
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> NoOp("SIP/2405243333-7457",
> > "Returned from dialparties with no extensions to call") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> SetVar("SIP/2405243333-7457",
> > "DIALSTATUS=BUSY") in new stack
> > Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> GotoIf("SIP/2405243333-7457",
> > "0?s-BUSY|1") in new stack
> > Jul 24 19:55:41 DEBUG[1078]: Not taking any branch
> > Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> GotoIf("SIP/2405243333-7457",
> > "0?s-BUSY|1") in new stack
> > Jul 24 19:55:41 DEBUG[1078]: Not taking any branch
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> NoOp("SIP/2405243333-7457",
> > "Sending to Voicemail box @default") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> Macro("SIP/2405243333-7457",
> > "vm|@default|BUSY") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> Goto("SIP/2405243333-7457",
> > "s-BUSY|1") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Goto (macro-vm,s-BUSY,1)
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> > VoiceMail("SIP/2405243333-7457", "b at default") in new stack
> > Jul 24 19:55:41 WARNING[1078]: No entry in voicemail config file for ''
> > Jul 24 19:55:41 VERBOSE[1078]:     -- Executing
> Hangup("SIP/2405243333-7457",
> > "") in new stack
> > Jul 24 19:55:41 VERBOSE[1078]:   == Spawn extension (macro-vm, s-BUSY, 2)
> > exited non-zero on 'SIP/2405243333-7457' in macro 'vm'
> > Jul 24 19:55:41 VERBOSE[1078]:   == Spawn extension (macro-exten-vm, s, 7)
> > exited non-zero on 'SIP/2405243333-7457' in macro 'exten-vm'
> > Jul 24 19:55:41 VERBOSE[1078]:   == Spawn extension (frombroadvoice, 201,
> 1)
> > exited non-zero on 'SIP/2405243333-7457'
> > Jul 24 19:55:41 DEBUG[1078]: cdr_mysql: inserting a CDR record.
> > Jul 24 19:55:41 DEBUG[1078]: cdr_mysql: SQL command as follows:  INSERT
> INTO
> > cdr
> >
> (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration
> ,b
> > illsec,disposition,amaflags,accountcode) VALUES ('2005-07-24
> 19:55:41','\"Fork
> > MD\" <4105156666>','4105156666','201','frombroadvoice',
> > 'SIP/2405243333-7457','','Hangup','',0,0,'ANSWERED',3,'')
> > Jul 24 19:55:41 DEBUG[1078]: update_user_counter(2405243333) - decrement
> inUse
> > counter
> > Jul 24 19:55:41 DEBUG[1078]: 2405243333 is not a local user
> > Jul 24 19:55:42 DEBUG[1078]: Stopping retransmission on
> > 'SD2o6lf01-7785a6b5fa835afb5133a88d729436bb-js11002' of Response
> 631438686:
> > Found
> > Jul 24 19:55:42 DEBUG[1078]: Stopping retransmission on
> > 'SD2o6lf01-7785a6b5fa835afb5133a88d729436bb-js11002' of Request 102: Found
> > Jul 24 19:55:45 DEBUG[1078]: Setting NAT on RTP to 0
> > Jul 24 19:55:45 DEBUG[1078]: Stopping retransmission on
> > '42817fbf52816b65420e8bd45b9c8f14 at 207.114.24.26' of Request 102: Found
> > Jul 24 19:55:55 DEBUG[1078]: Auto destroying call
> > '1865333733BC43D092222E5B07982954 at 207.114.24.26'
> >
> >
> >
> > I still get what acts like it took the call, I hear a click followed by a
> few
> > minutes of silence, and then I get dialtone back.      Strange indeed,
> sure
> > has me confused.
> >
> >
> >
> >
> >
> > ---
> > Howard Leadmon - howard at leadmon.net
> > http://www.leadmon.net
> >
> >
> > > -----Original Message-----
> > > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> > > bounces at lists.digium.com] On Behalf Of dbruce
> > > Sent: Sunday, July 24, 2005 7:38 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users]
> > > HelpwithAsterisk at homeandBroadvoiceincomingcalls..
> > >
> > > What is happening is that the _X. extension is catching the call... you
> need
> > > to take it out... it was only meant as a test to make sure which
> extension
> > > it was actually being sent to...
> > >
> > > It seems that it is then executing your vm macro without any
> parameters...
> > > don't know why that would happen...
> > >
> > > Try again after removing the catchall...
> > >
> > > Regards,
> > > Derek
> > >
> > > ----- Original Message -----
> > > From: "Howard Leadmon" <howard at leadmon.net>
> > > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> > > <asterisk-users at lists.digium.com>
> > > Sent: Sunday, July 24, 2005 4:35 PM
> > > Subject: RE: [Asterisk-Users] Help
> > > withAsterisk at homeandBroadvoiceincomingcalls..
> > >
> > >
> > > >
> > > >  OK, I added in the rule you gave me below, and yep it said 201, so
> your
> > > right
> > > > on with that one.
> > > >
> > > > So with the following extension rules:
> > > >
> > > > ;setup SIP extension for BroadVoice
> > > > [globals]
> > > > BVNUMBER=2405243333 ; your calling number
> > > > BVRINGS=201 ; the phone to ring
> > > > BVVMBOX=201 ; the VM box for this user
> > > >
> > > >
> > > > [outrt-003-BroadVoice]
> > > > include => outrt-003-BroadVoice-custom
> > > > exten => _8.,1,Dial(SIP/${EXTEN:1}@sip.broadvoice.com,30)
> > > > exten => _8.,2,Congestion()
> > > > exten => _8.,102,Busy()
> > > >
> > > > [frombroadvoice]
> > > > exten => _X.,1,Noop(Incoming call for extension ${EXTEN}in context
> > > > frombroadvoice)
> > > > exten => 201,1,Macro(exten-vm,${BVRINGS}@default,${BVRINGS})
> > > > exten => ${VM_PREFIX}${BVVMBOX},1,Macro(vm,${BVVMBOX})
> > > >
> > > >
> > > >
> > > > I now see:
> > > >
> > > >
> > > > Jul 24 18:25:37 DEBUG[1078]: Setting NAT on RTP to 0
> > > > Jul 24 18:25:37 DEBUG[1078]: Check for res for 2405243333
> > > > Jul 24 18:25:37 DEBUG[1078]: 2405243333 is not a local user
> > > > Jul 24 18:25:37 DEBUG[1078]: build_route: Contact hop:
> > > > <sip:4105156666 at 147.135.0.128:5060;ep=147.135.0.129;transport=udp>
> > > > Jul 24 18:25:37 VERBOSE[1078]:     -- Executing
> > > NoOp("SIP/2405243333-13be",
> > > > "Incoming call for extension 701in context frombroadvoice") in new
> stack
> > > > Jul 24 18:25:47 VERBOSE[1078]:   == CDR updated on SIP/2405243333-13be
> > > > Jul 24 18:25:47 VERBOSE[1078]:     -- Executing
> > > Macro("SIP/2405243333-13be",
> > > > "vm|") in new stack
> > > > Jul 24 18:25:47 VERBOSE[1078]:     -- Executing
> > > Goto("SIP/2405243333-13be",
> > > > "s-|1") in new stack
> > > > Jul 24 18:25:47 VERBOSE[1078]:     -- Goto (macro-vm,s-,1)
> > > > Jul 24 18:25:47 VERBOSE[1078]:     -- Executing
> > > > VoiceMail("SIP/2405243333-13be", "u") in new stack
> > > > Jul 24 18:25:47 WARNING[1078]: No entry in voicemail config file for
> ''
> > > > Jul 24 18:25:47 VERBOSE[1078]:     -- Executing
> > > Hangup("SIP/2405243333-13be",
> > > > "") in new stack
> > > > Jul 24 18:25:47 VERBOSE[1078]:   == Spawn extension (macro-vm, s-, 2)
> > > exited
> > > > non-zero on 'SIP/2405243333-13be' in macro 'vm'
> > > > Jul 24 18:25:47 VERBOSE[1078]:   == Spawn extension (frombroadvoice, ,
> 1)
> > > > exited non-zero on 'SIP/2405243333-13be'
> > > > Jul 24 18:25:47 DEBUG[1078]: cdr_mysql: inserting a CDR record.
> > > > Jul 24 18:25:47 DEBUG[1078]: cdr_mysql: SQL command as follows:
> INSERT
> > > INTO
> > > > cdr
> > > >
> > >
> (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration
> > > ,b
> > > > illsec,disposition,amaflags,accountcode) VALUES ('2005-07-24
> > > > 18:25:37','\"Leadmon H \"
> <4105156666>','4105156666','','frombroadvoice',
> > > > 'SIP/2405243333-13be','','Hangup','',10,0,'ANSWERED',3,'')
> > > > Jul 24 18:25:47 DEBUG[1078]: update_user_counter(2405243333) -
> decrement
> > > inUse
> > > > counter
> > > > Jul 24 18:25:47 DEBUG[1078]: 2405243333 is not a local user
> > > > Jul 24 18:25:47 DEBUG[1078]: Stopping retransmission on
> > > > 'SD2o9vc01-cd0c7ebdf9d4cc726c74f1e511173834-js11002' of Response
> > > 628736564:
> > > > Found
> > > > Jul 24 18:25:47 DEBUG[1078]: Stopping retransmission on
> > > > 'SD2o9vc01-cd0c7ebdf9d4cc726c74f1e511173834-js11002' of Request 102:
> Found
> > > >
> > > >
> > > >
> > > > On a side note, and I want to mention it as maybe relevant, something
> is
> > > now
> > > > screwy with my vMail, which is strange as I haven't done anything but
> the
> > > > tests here for you.   I can access the vmail, and if an extension is
> busy
> > > it
> > > > goes to the vmail.   What is messed up is that if I dial an extension
> and
> > > just
> > > > let it ring, it just keeps on ringing and doesn't transfer out.  For
> the
> > > life
> > > > of me I can't figure that one out right now, and don't see how just
> > > changing
> > > > this broadvoice context now has it so extension to extension (say 200
> > > calling
> > > > 202) is no longer getting a mailbox.   Never easy..  :(
> > > >
> > > >
> > > > Still when I call the broadvoice number, something changed, as now I
> get a
> > > > moment of silence, and then it hangs up.  No ringing, no vmail,
> actually I
> > > > think I got a fraction of a ring one time, then it hung up.
> > > >
> > > > I am for sure still confused on this one, so any help much
> appreciated...
> > > >
> > > >
> > > > ---
> > > > Howard Leadmon - howard at leadmon.net
> > > > http://www.leadmon.net
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-
> > > > > bounces at lists.digium.com] On Behalf Of dbruce
> > > > > Sent: Sunday, July 24, 2005 5:04 PM
> > > > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > > > Subject: Re: [Asterisk-Users] Help with
> > > > > Asterisk at homeandBroadvoiceincomingcalls..
> > > > >
> > > > > Ok.. I screwed up...
> > > > >
> > > > > You have a register statement:
> > > > >
> > >
> register=2405243333 at sip.broadvoice.com:123abc:2405243333 at sip.broadvoice.com/
> > > > > 201
> > > > >
> > > > > so, the incoming call from broadvoice will be sent to extension 201
> in
> > > the
> > > > > frombroadvoice context.
> > > > >
> > > > > To ensure what is going on, use this as your context.
> > > > > exten => _X.,1,Noop(Incoming call for extension ${EXTEN} in context
> > > > > frombroadvoice)
> > > > >
> > > > > That will tell you exactly what is being sent into the context.
> > > > >
> > > > > You are using the latest AAH, so the variable substitutions will
> work.
> > > > >
> > > > > I expect that you will end up using the following in frombroadvoice:
> > > > >
> > > > > exten => ${BVRINGS},1,Macro(exten-vm,${BVRINGS}@default,${BVRINGS})
> > > > >
> > > > > Sorry for the confusion....
> > > > >
> > > > > Regards,
> > > > > Derek
> > > >
> > > > (Old parts removed)
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users at lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
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> > >
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> >
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