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jleed at me.com
Guest
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 Posted: Tue Sep 02, 2014 12:50 pm Post subject: [asterisk-users] PJSIP issues with handling incoming calls |
 |
|
Okay, contact_user seems like do the job. Thanksis contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi script
On Sep 2, 2014, at 7:04 PM, Rainer Piper <rainer.piper@soho-piper.de (rainer.piper@soho-piper.de)> wrote:
| Quote: | I use in pjsip.conf
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=[url=sip:sipgate.de]sip:sipgate.de[/url]
client_uri=[url=sip:555123456@sipgate.de]sip:555123456@sipgate.de[/url]
contact_user=sipgatefilter ; goto the filter in extensions.conf
retry_interval=60
forbidden_retry_interval=600
expiration=3600
extensions.conf ; i'm cutting the dialed number out of the invite Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** ${CALLERID(num)} ***)
same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same => n,NoOp(**** 49${gotoadr:-11} ****)
same => n,Goto(49${gotoadr:-11},1)
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE -> MENU
exten => 4922897167160,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r ([email]EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r[/email]))
; incoming VOIP 97167161 SIPGATE
exten => 4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten => 4922897167162,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN}@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167163 SIPGATE
exten => 4922897167163,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN}@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167164 SIPGATE
exten => 4922897167164,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN}@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167165 SIPGATE
exten => 4922897167165,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN}@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incncoming VOIP 97167166 Mailbox
exten => 4922897167166,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN}@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167167 CONF. 1
exten => 4922897167167,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN}@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167168 CONF. 2
;exten => 4922897167168,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN}@sip.soho-piper.de&PJSIP/7000,,r[/email]))
exten => 4922897167168,1,Answer
same => n,echo()
same => n,Hangup()
; incoming VOIP 97167169 FAX
;exten => 4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN}@sip.soho-piper.de&PJSIP/7000,,r[/email]))
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
| Quote: | Nick Awesome wrote:
| Quote: | register => 73432260005:pass@10001
register => 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
|
Can you provide a sip debug of calls to both of these? I'm confused how that... works...
|
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: [url=sip:7000@sip.soho-piper.de:5072]sip:7000@sip.soho-piper.de:5072[/url] (pjsip-test)
--
_____________________________________________________________________
-- 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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rainer.piper at soho-p...
Guest
|
| Posted: Tue Sep 02, 2014 1:11 pm Post subject: [asterisk-users] PJSIP issues with handling incoming calls |
|
|
contact_user can be anything and calling an agi is no problem
Am 02.09.2014 um 19:49 schrieb Nick Awesome:
| Quote: | Okay, contact_user seems like do the job. Thanks is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi script
On Sep 2, 2014, at 7:04 PM, Rainer Piper <rainer.piper@soho-piper.de (rainer.piper@soho-piper.de)> wrote:
| Quote: | I use in pjsip.conf
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=[url=sip:sipgate.de]sip:sipgate.de[/url]
client_uri=[url=sip:555123456@sipgate.de]sip:555123456@sipgate.de[/url]
contact_user=sipgatefilter ; goto the filter in extensions.conf
retry_interval=60
forbidden_retry_interval=600
expiration=3600
extensions.conf ; i'm cutting the dialed number out of the invite Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** ${CALLERID(num)} ***)
same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same => n,NoOp(**** 49${gotoadr:-11} ****)
same => n,Goto(49${gotoadr:-11},1)
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE -> MENU
exten => 4922897167160,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r[/email]))
; incoming VOIP 97167161 SIPGATE
exten => 4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten => 4922897167162,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167163 SIPGATE
exten => 4922897167163,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167164 SIPGATE
exten => 4922897167164,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167165 SIPGATE
exten => 4922897167165,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incncoming VOIP 97167166 Mailbox
exten => 4922897167166,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167167 CONF. 1
exten => 4922897167167,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167168 CONF. 2
;exten => 4922897167168,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
exten => 4922897167168,1,Answer
same => n,echo()
same => n,Hangup()
; incoming VOIP 97167169 FAX
;exten => 4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
| Quote: | Nick Awesome wrote:
| Quote: | register => 73432260005:pass@10001
register => 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
|
Can you provide a sip debug of calls to both of these? I'm confused how that... works...
|
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: [url=sip:7000@sip.soho-piper.de:5072]sip:7000@sip.soho-piper.de:5072[/url] (pjsip-test)
--
_____________________________________________________________________
-- 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
|
|
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: [url=sip:7000@sip.soho-piper.de:5072]sip:7000@sip.soho-piper.de:5072[/url] (pjsip-test) |
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rainer.piper at soho-p...
Guest
|
| Posted: Tue Sep 02, 2014 1:32 pm Post subject: [asterisk-users] PJSIP issues with handling incoming calls |
|
|
Am 02.09.2014 um 20:11 schrieb Rainer Piper:
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: [url=sip:7000@sip.soho-piper.de:5072]sip:7000@sip.soho-piper.de:5072[/url] (pjsip-test) |
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| Back to top |
|
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rainer.piper at soho-p...
Guest
|
| Posted: Tue Sep 02, 2014 1:36 pm Post subject: [asterisk-users] PJSIP issues with handling incoming calls |
|
|
contact_user in pjsip.conf has to point to the filter or to an agi in the extentions.conf
like:
pjsip.conf
contact_user=blablabla
extensions.conf
exten => blablabla, 1,NoOp(*** ${PJSIP_HEADER(read,To)} *** ${CALLERID(num)} ***)
Am 02.09.2014 um 20:11 schrieb Rainer Piper:
| Quote: | contact_user can be anything and calling an agi is no problem
Am 02.09.2014 um 19:49 schrieb Nick Awesome:
| Quote: | Okay, contact_user seems like do the job. Thanks is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi script
On Sep 2, 2014, at 7:04 PM, Rainer Piper <rainer.piper@soho-piper.de (rainer.piper@soho-piper.de)> wrote:
| Quote: | I use in pjsip.conf
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=[url=sip:sipgate.de]sip:sipgate.de[/url]
client_uri=[url=sip:555123456@sipgate.de]sip:555123456@sipgate.de[/url]
contact_user=sipgatefilter ; goto the filter in extensions.conf
retry_interval=60
forbidden_retry_interval=600
expiration=3600
extensions.conf ; i'm cutting the dialed number out of the invite Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** ${CALLERID(num)} ***)
same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same => n,NoOp(**** 49${gotoadr:-11} ****)
same => n,Goto(49${gotoadr:-11},1)
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE -> MENU
exten => 4922897167160,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r[/email]))
; incoming VOIP 97167161 SIPGATE
exten => 4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten => 4922897167162,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167163 SIPGATE
exten => 4922897167163,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167164 SIPGATE
exten => 4922897167164,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167165 SIPGATE
exten => 4922897167165,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incncoming VOIP 97167166 Mailbox
exten => 4922897167166,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167167 CONF. 1
exten => 4922897167167,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167168 CONF. 2
;exten => 4922897167168,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
exten => 4922897167168,1,Answer
same => n,echo()
same => n,Hangup()
; incoming VOIP 97167169 FAX
;exten => 4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
| Quote: | Nick Awesome wrote:
| Quote: | register => 73432260005:pass@10001
register => 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
|
Can you provide a sip debug of calls to both of these? I'm confused how that... works...
|
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: [url=sip:7000@sip.soho-piper.de:5072]sip:7000@sip.soho-piper.de:5072[/url] (pjsip-test)
--
_____________________________________________________________________
-- 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
|
|
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: [url=sip:7000@sip.soho-piper.de:5072]sip:7000@sip.soho-piper.de:5072[/url] (pjsip-test)
|
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: [url=sip:7000@sip.soho-piper.de:5072]sip:7000@sip.soho-piper.de:5072[/url] (pjsip-test) |
|
| Back to top |
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|
jleed at me.com
Guest
|
| Posted: Tue Sep 02, 2014 11:50 pm Post subject: [asterisk-users] PJSIP issues with handling incoming calls |
|
|
Ok, thanks for an answer. That solution works.
On 02 Sep 2014, at 22:36, Rainer Piper <rainer.piper@soho-piper.de (rainer.piper@soho-piper.de)> wrote:
| Quote: | contact_user in pjsip.conf has to point to the filter or to an agi in the extentions.conf
like:
pjsip.conf
contact_user=blablabla
extensions.conf
exten => blablabla, 1,NoOp(*** ${PJSIP_HEADER(read,To)} *** ${CALLERID(num)} ***)
Am 02.09.2014 um 20:11 schrieb Rainer Piper:
| Quote: | contact_user can be anything and calling an agi is no problem
Am 02.09.2014 um 19:49 schrieb Nick Awesome:
| Quote: | Okay, contact_user seems like do the job. Thanks is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi script
On Sep 2, 2014, at 7:04 PM, Rainer Piper <rainer.piper@soho-piper.de (rainer.piper@soho-piper.de)> wrote:
| Quote: | I use in pjsip.conf
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=[url=sip:sipgate.de]sip:sipgate.de[/url]
client_uri=[url=sip:555123456@sipgate.de]sip:555123456@sipgate.de[/url]
contact_user=sipgatefilter ; goto the filter in extensions.conf
retry_interval=60
forbidden_retry_interval=600
expiration=3600
extensions.conf ; i'm cutting the dialed number out of the invite Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** ${CALLERID(num)} ***)
same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same => n,NoOp(**** 49${gotoadr:-11} ****)
same => n,Goto(49${gotoadr:-11},1)
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE -> MENU
exten => 4922897167160,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r[/email]))
; incoming VOIP 97167161 SIPGATE
exten => 4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten => 4922897167162,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167163 SIPGATE
exten => 4922897167163,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167164 SIPGATE
exten => 4922897167164,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167165 SIPGATE
exten => 4922897167165,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incncoming VOIP 97167166 Mailbox
exten => 4922897167166,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167167 CONF. 1
exten => 4922897167167,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
; incoming VOIP 97167168 CONF. 2
;exten => 4922897167168,1,Dial(PJSIP/EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
exten => 4922897167168,1,Answer
same => n,echo()
same => n,Hangup()
; incoming VOIP 97167169 FAX
;exten => 4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r ([email]EXTEN%7D@sip.soho-piper.de&PJSIP/7000,,r[/email]))
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
| Quote: | Nick Awesome wrote:
| Quote: | register => 73432260005:pass@10001
register => 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
|
Can you provide a sip debug of calls to both of these? I'm confused how that... works...
|
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: [url=sip:7000@sip.soho-piper.de:5072]sip:7000@sip.soho-piper.de:5072[/url] (pjsip-test)
--
_____________________________________________________________________
-- 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
|
|
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: [url=sip:7000@sip.soho-piper.de:5072]sip:7000@sip.soho-piper.de:5072[/url] (pjsip-test)
|
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: [url=sip:7000@sip.soho-piper.de:5072]sip:7000@sip.soho-piper.de:5072[/url] (pjsip-test)
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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