[asterisk-users] Direct dial
I have clarification in which how we can enable direct dial when we press numbers in 3cx phone on hook. Now its like we have to use dial button to dial. Previously I am able to dial directly after entering number. Now its not working. Can someone help me on it. Is this a setup that we have to do in freepbx or In 3cx Phones? Regards Darin Egocentrix -- _ -- 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
[asterisk-users] voicemail customization
Is there any solution, I want to know at first sight from which division the voicemail came from like for example I need 3 more users to send when some one calls an inbound route named darin when it reaches darin voicemail then admin should send from da...@yahoo.com -- _ -- 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
[asterisk-users] 503 unable to load
Can any one suggest me what I have to do for this issue. There is no nat as
i have directly connected to internert without firewall.
Got SIP response 503 "Unable to load gateways" back from
xxx.xxx.xxx.xxx:5060
-- SIP/outbound-0994 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/1002-0993", "Dial
failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34")
in new stack
-- Executing [s@macro-dialout-trunk:24] Goto("SIP/1002-0993",
"s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/1002-0993",
"RC=34") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/1002-0993",
"34,1") in new stack
-- Goto (macro-dialout-trunk,34,1)
-- Executing [34@macro-dialout-trunk:1] Goto("SIP/1002-0993",
"continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/1002-0993",
"1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/1002-0993",
"TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to
other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/1002-0993",
"CALLERID(number)=1002") in new stack
-- Executing [8834404@from-internal:6] Macro("SIP/1002-0993",
"outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/1002-0993", "") in new
stack
-- Executing [s@macro-outisbusy:2] Playback("SIP/1002-0993",
"all-circuits-busy-now,noanswer") in new stack
-- Playing 'all-circuits-busy-now.ulaw' (language 'en')
-- Executing [s@macro-outisbusy:3] Playback("SIP/1002-0993",
"pls-try-call-later,noanswer") in new stack
-- Playing 'pls-try-call-later.ulaw' (language 'en')
[2012-11-05 11:24:37] WARNING[29848]: file.c:766 ast_readaudio_callback:
Failed to write frame
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on
'SIP/1002-0993' in macro 'outisbusy'
== Spawn extension (from-internal, 8834404, 6) exited non-zero on
'SIP/1002-0993'
-- Executing [h@from-internal:1] Hangup("SIP/1002-0993", "") in new
stack
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/1002-0993'
== Extension Changed auto_hint_1002[from-internal] new state Idle for
Notify User 1002
== Extension Changed auto_hint_1002[from-internal] new state Idle for
Notify User 1001
Darin
--
_
-- 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
[asterisk-users] multitenanat third party app
Is it possible to bul multitenant system using some third party opensouce application My design is like this. Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- 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
[asterisk-users] Multitenant opensouce application
is another way to build Multi Tenant system, have to design like Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- 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
[asterisk-users] Cant we have same extension in different context?
Cant we have same extension in different context? This is what we we want in same pbx server? Company A Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B Context Company_B IVR Company B Extensions: 101,102,103,104 etc. -- _ -- 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
[asterisk-users] multi tenant
Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- 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
[asterisk-users] Asterisk call forward to mobile numbers if ringroup is not picking up the call
Hi team, I had setup an asterisk with freepbx and I want to forward the calls to mobile when nocone is picking up calls in the ringroup. I have already added custom ext and given string as Local/mobno/from -internal. But now reciever is geting pilot number only I need to get the callers number in the forwarded mobile. Can you tell me how to do it. -- _ -- 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
Re: [asterisk-users] Calls from talkonaut to pstn Phone
Hi I need to call from talkonaut to PSTN number using our asterisk server.
VoIP service which allows utilizing GPRS, 3G or WIFI mobile data
connections to make free of cheap VoIP calls to phones through my SIP
network, to Google Talk, MSN, AIM or Yahoo through GTalk2VoIP soft-switch
in which I can set up on my asterisk server in INDIA servers to deliver
VoIP calls from branded-Talkonaut users to your SIP network, to VoIM
clients and vice versa.
I have configured like this
Dial plan (extension.conf) is
[gtalk_incoming]
exten => s,1,Verbose(2,Incoming Gtalk call from ${CALLERID(all)})
same => n,Answer()
same => n,Dial(SIP/1000,30)
same => n,Hangup()
[google_out]
exten => 1010,1,Verbose(2,Extension 1010 calling darinfaststr...@gmail.com)
same => n,Dial(Gtalk/asterisk/adari...@gmail.com,30)
same => n,Hangup()
[LocalSets]
exten => XX,1,Verbose(2,Placing call to ${EXTEN} via Google Voice)
same => n,Dial(Gtalk/asterisk/+${EXTEN}@voice.google.com)
same => n,Hangup()
Jabber.conf
[general]
debug=no
autoprune=no
autoregister=yes
[username]
type=client
serverhost=talk.google.com
username=darinfaststr...@gmail.com/Talk
secret=x
priority=1
port=5222
usetls=yes
usesasl=yes
status=Available
statusmessage="I am an Asterisk Server"
timeout=100
keepalive=yes
Gtalk.conf
[guest]
disallow=all
allow=ulaw
context=google-in
[buddy]
username=m...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=gtalk_account
But when I tried to make call from talkonaut after configure the own dial
plan in talkonaut voip settings I cant make calls through asterisk or vice
versa.
On Sat, Sep 8, 2012 at 10:30 PM, wrote:
> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>1. Automatic reconnect to ODBC sources? (Stefan at WPF)
>2. Rolm T1 not passing caller ID to asterisk (Kohler, Ed)
>3. how to load our own .wav sound files in the dial plans for
> playback (upendra)
>
>
> --
>
> Message: 1
> Date: Fri, 7 Sep 2012 19:17:58 +0200
> From: Stefan at WPF
> Subject: [asterisk-users] Automatic reconnect to ODBC sources?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID:
> pwaalhgkwnqjj_qksmi+z4qdgx...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I am using Asterisk 1.8.10.1 on Ubuntu Server 12.04. I use MySQL to store
> CDR records using cdr_adaptive_odbc.
> The problem: When the MySQL Server fails for whatever reason, Asterisk
> never reconnects automatically! So I loose all CDR informations even after
> the MySQL server works again.
> isql (used as ODBC testtool) tells me, that ODBC reconnects or does a new
> connection on request, so ODBC seems not to be the problem, but Asterisk.
> Is there any automatic reconnect option on Asterisk concercing ODBC /
> cdr_adaptive_odbc? My complete configuration including ODBC is as follows:
>
> *ODBC:
>
> /etc/odbcinst.ini
> *
> >
> > [MySQL]
> > Description = ODBC for MySQL
> > Driver = /usr/lib/i386-linux-gnu/odbc/libmyodbc.so
> > Setup = /usr/lib/i386-linux-gnu/odbc/libodbcmyS.so
> > FileUsage = 1
> > OPTION = 4194304
> >
> >
> */etc/odbc.ini *
>
> > [asterisk-connector]
> > Description = MySQL connection to 'asterisk' database
> > Driver= MySQL
> > Database = asterisk
> > Server= localhost
> > UserName = root
> > Password = password
> > Port = 3306
>
> *
>
> ASTERISK CDR*:
>
> */etc/asterisk/res_odbc.conf*
>
> > [asterisk]
> > enabled => yes
> > dsn => asterisk-connector
> > username => root
> > password => password
> > pre-connect => yes
> > pooling => no
> > limit => 1
> > connect_timeout => 1
> > idlecheck => 1
> >
>
> */etc/asterisk/cdr_adaptive_odbc.conf *
>
> > [mytable]
> > connection = asterisk
> > table = asterisk_cdr
> >
>
> */etc/asterisk/cdr.conf*
> >
> > [general]
> > enable = yes
> > unanswered = yes
> >
>
>
> Thanks for any hint!
>
> Best regards
> Stefan
> -- next part --
> An HTML attachment was scrubbed...
> URL: <
> http://lists.digium.com/pipermail/asterisk-users/attachments/20120907/ff635e3c/attachment-0001.htm
> >
>
> --
>
> Message: 2
> Date: Fri, 7 Sep 2012 14:49:21 -0400
> From: "Kohler, Ed"
> Subject: [asterisk-users] Rolm T1 not passing caller ID to asterisk
> To: "'asterisk-users@lists.digium.com'"
>
> Cc: "B
[asterisk-users] Getting error while dialing outside (PRI Connection card TE110P ASTERISK1.8 AND FREEPBX 2.10
Getting error while dialing outside (PRI Connection card TE110P ASTERISK1.8 AND FREEPBX 2.10 ; Copied from DAHDI Module of FreePBX [general] #include chan_dahdi_general.conf #include chan_dahdi_custom.conf [channels] ; include dahdi groups defined by DAHDI module of FreePBX #include chan_dahdi_groups.conf ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf system.conf span=1,0,0,ESF,B8ZS bchan=1-23 dchan=24 loadzone=in defaultzone=in -- _ -- 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
[asterisk-users] Fwd: changing port 5060 to 5061
-- Forwarded message -- From: darin iv Date: Mon, 11 Apr 2011 14:33:24 +0530 Subject: changing port 5060 to 5061 To: asterisk-users@lists.digium.com Dear Experts, please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.. plz send me step by step option to do this and from where i have to change this. -- _ -- 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
[asterisk-users] changing port 5060 to 5061
Dear Experts, please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.. -- _ -- 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
[asterisk-users] changing port 5060 to 5061
please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.. On 4/10/11, asterisk-users-requ...@lists.digium.com wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-requ...@lists.digium.com > > You can reach the person managing the list at > asterisk-users-ow...@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > >1. Re: asterisk-users Digest, Vol 81, Issue 27 (Steve Edwards) >2. Re: Asterisk FOP (Doug Lytle) >3. Re: Asterisk FOP (Flavio Miranda) >4. Re: Asterisk FOP (Doug Lytle) >5. Re: IAX2/0.0.29.199 (Satish Patel) >6. Re: Call Recording using MixMonitor - close, but would like > some more words of wisdom. (Dan Journo) >7. Re: Call recording - methodology (Dan Journo) >8. Re: Asterisk FOP (Flavio Miranda) >9. Re: send voicemail to multiple emails (vip killa) > 10. Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199] > (Tzafrir Cohen) > 11. Re: IAX2/0.0.29.199 (Tzafrir Cohen) > 12. AsteriskNow updated to Centos 5.6 and DAHDI doesn't work > (Frank Tarczynski) > 13. Re: Call recording - methodology (Silver Thorne) > 14. Re: Call recording - methodology (Dan Journo) > > > -- > > Message: 1 > Date: Sat, 9 Apr 2011 10:38:00 -0700 (PDT) > From: Steve Edwards > Subject: Re: [asterisk-users] asterisk-users Digest, Vol 81, Issue 27 > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: > > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > > On Sat, 9 Apr 2011, darin iv wrote: > > 0) Don't re-post the entire digest back to the list it came from. Posting > 36k of cruft to ask 'How to change SIP port number?' seems somewhat > 'newbish.' > > 1) Try Google. Try 'How to change SIP port number in Asterisk?' > > 2) Re-post with a new, relevant Subject and you will get relevant > responses. > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > > > -- > > Message: 2 > Date: Sat, 09 Apr 2011 14:11:39 -0400 > From: Doug Lytle > Subject: Re: [asterisk-users] Asterisk FOP > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <4da0a15b.8020...@drdos.info> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Flavio Miranda wrote: >> >> >> I am truing to set up FOP but I getting the following log: >> > What version of FOP? 1 or 2, what OS? What version of Asterisk? > > Doug > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > > > > -- > > Message: 3 > Date: Sat, 9 Apr 2011 16:35:45 -0300 > From: Flavio Miranda > Subject: Re: [asterisk-users] Asterisk FOP > To: Asterisk Asterisk > Message-ID: > Content-Type: text/plain; charset="iso-8859-1" > > > Hi, > FOP 1 > OS Debian Lenny > Asterisk 1.6 > > Att, > > > > Flavio Roberto Miranda > > MSN:flaviormira...@hotmail.com > Skype: flaviormiranda > > > >> Date: Sat, 9 Apr 2011 14:11:39 -0400 >> From: supp...@drdos.info >> To: asterisk-users@lists.digium.com >> Subject: Re: [asterisk-users] Asterisk FOP >> >> Flavio Miranda wrote: >> > >> > >> > I am truing to set up FOP but I getting the following log: >> > >> What version of FOP? 1 or 2, what OS? What version of Asterisk? >> >> Doug >> >> >> -- >> Ben Franklin quote: >> >> "Those who would give up Essential Liberty to purchase a little Temporary >> Safety, deserve neither Liberty nor Safety." >> >> >> >> -- >> __
Re: [asterisk-users] asterisk-users Digest, Vol 81, Issue 27
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-requ...@lists.digium.com
wrote:
> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> 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 +
> From: satish patel
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> To: asterisk-users
> Message-ID:
> 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: pabelan...@digium.com
>> To: asterisk-users@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@from-sip:1] Macro("SIP/7527-006b",
>> > "stdexten,7623,SIP/7623") in new stack
>> > -- Executing [s@macro-stdexten:1] ChanIsAvail("SIP/7527-006b",
>> > "SIP/7623&IAX2/7623,20,t") in new stack
>> > -- Hungup 'IAX2/0.0.29.199:4569-5255'
>> > -- Executing [s@macro-stdexten:2] NoOp("SIP/7527-006b",
>> > "IAX2/0.0.29.199:4569-5255") in new stack
>> > -- Executing [s@macro-stdexten:3] NoOp("SIP/7527-006b", "0&0")
>> > in new stack
>> > -- Auto fallthrough, channel 'SIP/7527-006b' 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
>
> -- next part --
> An HTML attachment was scrubbed...
> URL:
>
>
> --
>
> Message: 2
> Date: Fri, 8 Apr 2011 16:57:27 +0100 (BST)
> From: Naomi Rosenberg
> Subject: Re: [asterisk-users] Variable inheritance with dialplan
> command Originate
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID: <28369896.3418.1302278246954.JavaMail.root@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"
> To: asterisk-users@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() d
