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jleed at me.com
Guest

Posted: Tue Sep 02, 2014 1:48 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

Hello guys. Have 2 external numbers that required registration on provider server, trunk1: 73432260005@80.75.132.66 trunk2: 73432260050@80.75.132.66 Thing is I can’t figure out how to route them to different IVRs by default Asterisk can’t match endpoint Request from '<[url=sip:+]sip:+[/url] 73432260005@80.75.132.66>' failed for '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No matching endpoint found Can’t set identify by IP because they got the same ip. Is there way to configure asterisk so incoming calls from same IP but different ID will use different contexts?

admin at tootai.net
Guest

Posted: Tue Sep 02, 2014 2:35 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

Le 02/09/2014 08:47, Nick Awesome a écrit :

Quote:

Hello guys.

Quote:

Have 2 external numbers that required registration on provider server,

trunk1: 734322600*05*@80.75.132.66
trunk2: 734322600*50*@80.75.132.66

Thing is I can’t figure out how to route them to different IVRs

by default Asterisk can’t match endpoint

Request from '<sip:+ 734322600*05*@80.75.132.66>' failed for
'80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
matching endpoint found

Can’t set /identify /by IP because they got the same ip.

Is there way to configure asterisk so incoming calls from same IP but
different ID will use different contexts?

You have to register to the gateway with each account user and password like

sip.conf

register = 734322600*05*:password1@myProvider/734322600*05*
register = 734322600*50*:password2@myProvider/734322600*50*

[myProvider]
type=peer
host=80.75.132.66
context=from-myProvider
...

extensions.conf

[from-myProvider]
exten = 734322600*05*,1,NoOp(Incoming call to 734322600*05*)
...

exten = 734322600*50*,1,NoOp(Incoming call to 734322600*50*)
...

--
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jleed at me.com
Guest

Posted: Tue Sep 02, 2014 2:39 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

So there is no way to do that with pjsip?

On 02 Sep 2014, at 11:35, Administrator TOOTAI <admin@tootai.net> wrote:

Quote:

Le 02/09/2014 08:47, Nick Awesome a écrit :

Quote:

Hello guys.

Quote:

--
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admin at tootai.net
Guest

Posted: Tue Sep 02, 2014 2:47 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

Le 02/09/2014 09:38, Nick Awesome a écrit :

Quote:

So there is no way to do that with pjsip?

Sorry, I didn't read carefully the subject. I can't answer for pjsip. My
bad Sad

Quote:

On 02 Sep 2014, at 11:35, Administrator TOOTAI <admin@tootai.net> wrote:

Quote:

Le 02/09/2014 08:47, Nick Awesome a écrit :

Quote:

Hello guys.

Quote:

jcolp at digium.com
Guest

Posted: Tue Sep 02, 2014 6:01 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

Nick Awesome wrote:

Quote:

Hello guys.

Kia ora,

Quote:

If the From header contains the destination number (as it seems to based
on your above log message and config) you can create two different
endpoints and match based on the user portion of the From header.

[734322600*05*]
type=endpoint
context=did-1
disallow=all
allow=ulaw

[734322600*50*]
type=endpoint
context=did-2
disallow=all
allow=ulaw

If this is not correct then you can only match once based on the source
IP address currently.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.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:
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jleed at me.com
Guest

Posted: Tue Sep 02, 2014 6:06 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

Thats because I call from one to other

here’s logs where I call from mobile

<--- Received SIP request (469 bytes) from UDP:80.75.132.66:5060 --->
ACK [url=sip:s@pbx_ip_address:57408;transport=UDP]sip:s@pbx_ip_address:57408;transport=UDP[/url] SIP/2.0
Via: SIP/2.0/UDP 80.75.132.66:5060;branch=z9hG4bK-524287-1-NGJlZjMxNThhYjI2YjI3Y2EyODE0MThhMTVkNjY0ZTA.--5c6da819e6300b26;rport
Max-Forwards: 70
To: <[url=sip:73432260005@80.75.132.66]sip:73432260005@80.75.132.66[/url]>;tag=z9hG4bK-524287-1-NGJlZjMxNThhYjI2YjI3Y2EyODE0MThhMTVkNjY0ZTA.--5c6da819e6300b26
From: <[url=sip:+79999823064@80.75.132.66]sip:+79999823064@80.75.132.66[/url]>;tag=7ozmpvsvqs26kcor.o
Call-ID: 18e2786560719216837824k41099rmwp
CSeq: 586 ACK
Content-Length: 0

<--- Received SIP request (469 bytes) from UDP:80.75.132.66:5060 --->
ACK [url=sip:s@pbx_ip_address:57408;transport=UDP]sip:s@pbx_ip_address:57408;transport=UDP[/url] SIP/2.0
Via: SIP/2.0/UDP 80.75.132.66:5060;branch=z9hG4bK-524287-1-YzhmMDE2NzE1YjRhOTM4NjQ1MjMxMmMyMmM0MWFiZTE.--be7c48325cdef400;rport
Max-Forwards: 70
To: <[url=sip:73432260050@80.75.132.66]sip:73432260050@80.75.132.66[/url]>;tag=z9hG4bK-524287-1-YzhmMDE2NzE1YjRhOTM4NjQ1MjMxMmMyMmM0MWFiZTE.--be7c48325cdef400
From: <[url=sip:+79999823064@80.75.132.66]sip:+79999823064@80.75.132.66[/url]>;tag=yddmzvcoi3waw24e.o
Call-ID: 22e7064301970213226722k41100rmwp
CSeq: 588 ACK
Content-Length: 0

On 02 Sep 2014, at 15:01, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:

Quote:

Nick Awesome wrote:

Quote:

Hello guys.

Kia ora,

Quote:

Have 2 external numbers that required registration on provider server,

trunk1: 734322600*05*@80.75.132.66 ([email]734322600*05*@80.75.132.66[/email])
trunk2: 734322600*50*@80.75.132.66 ([email]734322600*50*@80.75.132.66[/email])

Thing is I can’t figure out how to route them to different IVRs

by default Asterisk can’t match endpoint

Request from '<734322600*05*@80.75.132.66 ([email]734322600*05*@80.75.132.66[/email])>' failed for
'80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
matching endpoint found

Can’t set /identify /by IP because they got the same ip.

Is there way to configure asterisk so incoming calls from same IP but
different ID will use different contexts?

If the From header contains the destination number (as it seems to based on your above log message and config) you can create two different endpoints and match based on the user portion of the From header.

[734322600*05*]
type=endpoint
context=did-1
disallow=all
allow=ulaw

[734322600*50*]
type=endpoint
context=did-2
disallow=all
allow=ulaw

If this is not correct then you can only match once based on the source IP address currently.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.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

jcolp at digium.com
Guest

Posted: Tue Sep 02, 2014 6:15 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

Nick Awesome wrote:

Quote:

Thats because I call from one to other

Then no, you can only match based on IP address. This also applies to
chan_sip. You have to send both to the same context and then within
there you can differentiate them based on the dialed number.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello

asterisk-users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users

asterisk_list at earth...
Guest

Posted: Tue Sep 02, 2014 6:32 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

On Tuesday 02 Sep 2014, Nick Awesome wrote:

Quote:

Hello guys.

Have 2 external numbers that required registration on provider server,

trunk1: 73432260005@80.75.132.66
trunk2: 73432260050@80.75.132.66

Thing is I canâ€™t figure out how to route them to different IVRs

by default Asterisk canâ€™t match endpoint

Request from '<sip:+ 73432260005@80.75.132.66>' failed for
'80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
matching endpoint found

Canâ€™t set identify by IP because they got the same ip.

Is there way to configure asterisk so incoming calls from same IP but
different ID will use different contexts?

Can't you send them both to the same context initially; but once you are
there, match the outside number (which can be found in ${EXTEN} if it is the
number that was dialled from their end, or ${CALLERID(num)} if it is the
number they are calling from) within that context and use a GoToIf() to send
calls from trunk 2 to the correct context?

--
AJS

Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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jleed at me.com
Guest

Posted: Tue Sep 02, 2014 6:39 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

Tried doing that, but

first: AGI->exten is ’s’ for some reason.
and second its not practical, for example if 80.75.132.66 wound like to register on my * server - it will not work because I already using that IP with different endpoint

well, its critical trouble for me, coming back to chat_sip Neutral

On 02 Sep 2014, at 15:32, A J Stiles <asterisk_list@earthshod.co.uk> wrote:

Quote:

On Tuesday 02 Sep 2014, Nick Awesome wrote:

Quote:

Hello guys.

Have 2 external numbers that required registration on provider server,

trunk1: 73432260005@80.75.132.66
trunk2: 73432260050@80.75.132.66

Thing is I can’t figure out how to route them to different IVRs

by default Asterisk can’t match endpoint

Request from '<sip:+ 73432260005@80.75.132.66>' failed for
'80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
matching endpoint found

Can’t set identify by IP because they got the same ip.

Is there way to configure asterisk so incoming calls from same IP but
different ID will use different contexts?

jcolp at digium.com
Guest

Posted: Tue Sep 02, 2014 6:50 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

Nick Awesome wrote:

Quote:

Tried doing that, but

first: AGI->exten is ’s’ for some reason. and second its not
practical, for example if 80.75.132.66 wound like to register on my *
server - it will not work because I already using that IP with
different endpoint

well, its critical trouble for me, coming back to chat_sip Neutral

How will you do this in chan_sip? The behavior between the two is the
same, despite the configuration being different.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.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

jleed at me.com
Guest

Posted: Tue Sep 02, 2014 7:14 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

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

so now in context dialmap (agi application) AGI->agi_channel is 'SIP/10001-00000005’
parsing 10001 and checking db for matches, in db I have table with all my trunks information

On 02 Sep 2014, at 15:49, Joshua Colp <jcolp@digium.com> wrote:

Quote:

Nick Awesome wrote:

Quote:

Tried doing that, but

first: AGI->exten is ’s’ for some reason. and second its not
practical, for example if 80.75.132.66 wound like to register on my *
server - it will not work because I already using that IP with
different endpoint

well, its critical trouble for me, coming back to chat_sip Neutral

How will you do this in chan_sip? The behavior between the two is the same, despite the configuration being different.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.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

jcolp at digium.com
Guest

Posted: Tue Sep 02, 2014 8:09 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

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...

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.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

rainer.piper at soho-p...
Guest

Posted: Tue Sep 02, 2014 10:05 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

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)

rainer.piper at soho-p...
Guest

Posted: Tue Sep 02, 2014 10:25 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

PS all incoming calls are directed to sipgatefilter in extentions.conf and then filtered.
You maid have to adjust the -11 in Goto(49${gotoadr:-11},1) ... just look at the cli output NoOp(**** 49${gotoadr:-11} ****)

Am 02.09.2014 um 17:04 schrieb Rainer Piper:

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)

rainer.piper at soho-p...
Guest

Posted: Tue Sep 02, 2014 10:42 am Post subject: [asterisk-users] PJSIP issues with handling incoming calls

upps .... and delete the 49 in Goto(49${gotoadr:-11},1) and NoOp(**** 49${gotoadr:-11} ****)
just look at the cli output

Am 02.09.2014 um 17:25 schrieb Rainer Piper:

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)

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