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91
Other Yate server issues / Re: Yate Redirect SIP 300 Multiple Choices
« Last post by artur.baghdasaryan on September 18, 2017, 06:20:16 AM »
Hi,

Dear Monica,

Thanks for your response.
Actually we do need to use extensions.We are using Yate to act as SBC for transit calls.
Please could You advise how to get some params from SIP message and use them as variables in different contexts as it is described above the call cant be routed as there is no route in default context.Like from 300 Multiple choices message can we extract the contact field,get the user part and host part separately and use them to initialize a new call.We want to set this params as variables in regexroute for to route calls according to contact field from 300 Multiple choices message.
Please can you share the regexroute example.
Great thanks in advance.

/Arthur
92
Other Yate server issues / Re: Yate Redirect SIP 300 Multiple Choices
« Last post by Monica Tepelus on September 18, 2017, 03:58:25 AM »
Hi,

Yate doesn't implement all SIP 300 variations.

Pbxassist was not thought to work with SIP 300 requests. If you wish to use the pbxassist functioality you should configure the phones to send the dmnfs to Yate and configure the combination of digits in yate's pbxassist module. You can't combine SIP redirects with Yate's pbxassist.

93
Linux / Re: abonent information
« Last post by Monica Tepelus on September 18, 2017, 03:48:08 AM »
If Yate is restarted then yes, list of registered users is deleted. Users continue to have service.
Do you mean something else by "online users"?
94
Other Yate server issues / Yate Redirect SIP 300 Multiple Choices
« Last post by artur.baghdasaryan on September 15, 2017, 02:24:02 AM »
Dear Yate Team,

Please help to figure out why Yate cant redirect calls to destination which is in Contact header of SIP 300 Multiple Choices message.
I have enabled the pbxassist module and set diversion to yes.
Please could you advise where is the problem
Trace is below
20170915081051.612263 <sip:INFO> 'udp:0.0.0.0:5060' received 869 bytes SIP message from 1.1.1.1:5060 [0x1cfcef0]
------
INVITE sip:11111111@2.2.2.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6a19fc4e
Max-Forwards: 70
From: "+2222222222" <sip:+2222222222@1.1.1.1>;tag=as6e0e860a
To: <sip:11111111@2.2.2.2;user=phone>
Contact: <sip:+2222222222@1.1.1.1:5060>
Call-ID: 40d6a8f200621e456cbf8c1431df234a@1.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Test
Date: Fri, 15 Sep 2017 08:10:51 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 497095320 497095320 IN IP4 1.1.1.1
s=Test
c=IN IP4 1.1.1.1
t=0 0
m=audio 14266 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
------
20170915081051.621309 <sip:INFO> 'udp:0.0.0.0:5060' sending code 100 0x7f8ef0002b90 to 1.1.1.1:5060 [0x1cfcef0]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6a19fc4e;received=1.1.1.1
From: "+2222222222" <sip:+2222222222@1.1.1.1>;tag=as6e0e860a
To: <sip:11111111@2.2.2.2;user=phone>
Call-ID: 40d6a8f200621e456cbf8c1431df234a@1.1.1.1:5060
CSeq: 102 INVITE
Server: Test
Content-Length: 0

------
20170915081051.621410 <sip/23:ALL> YateSIPConnection::YateSIPConnection(0x7f8ef0010ad0,0x7f8ef402c1e0) [0x7f8ef0012020]
20170915081051.621550 <sip/23:ALL> NAT address is '(null)' [0x7f8ef0012020]
20170915081051.621793 <sip/23:ALL> Set media: audio=alaw,mulaw [0x7f8ef0012020]
20170915081051.622182 <INFO> Could not classify call from '+2222222222', wasted 214 usec
20170915081051.622255 <cdrbuild:INFO> Got message 'call.route' for untracked id 'sip/23'
20170915081051.622328 <RegexRoute:ALL> Jumping to context 'from_internal' by rule #1 '${module}sip'
20170915081051.622664 <RegexRoute:ALL> Jumping to context 'aster_test' by rule #9 '${address}^1\.1\.1\.1:'
20170915081051.622723 <pbxassist:CALL> Created guest assistant for 'sip/23'
20170915081051.622855 <INFO> Routing call to '11111111' in context 'default' via 'sip/sip:2222222222@3.3.3.3' in 558 usec
20170915081051.622903 <sip/23:ALL> NAT address is '(null)' [0x7f8ef0012020]
20170915081051.623092 <sip/24:ALL> YateSIPConnection::YateSIPConnection(0x7f8ef00036d0,'sip:2222222222@3.3.3.3') [0x7f8edc00ee80]
20170915081051.623171 <sip:ALL> YateSIPEndPoint::buildParty(0x7f8edc014400,'(null)',0,(nil))
20170915081051.623239 <sip/24:ALL> NAT address is '(null)' [0x7f8edc00ee80]
20170915081051.623381 <sip/24:ALL> Set media: audio=alaw,mulaw,g729 [0x7f8edc00ee80]
20170915081051.627078 <sip:INFO> 'udp:0.0.0.0:5060' sending code 183 0x7f8edc010830 to 1.1.1.1:5060 [0x1cfcef0]
------
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6a19fc4e;received=1.1.1.1
From: "+2222222222" <sip:+2222222222@1.1.1.1>;tag=as6e0e860a
To: <sip:11111111@2.2.2.2;user=phone>;tag=1459504950
Call-ID: 40d6a8f200621e456cbf8c1431df234a@1.1.1.1:5060
CSeq: 102 INVITE
Server: Test
Contact: <sip:11111111@2.2.2.2:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0

------
20170915081051.627200 <sip:INFO> 'udp:0.0.0.0:5060' sending 'INVITE sip:2222222222@3.3.3.3' 0x7f8edc014400 to 3.3.3.3:5060 [0x1cfcef0]
------
INVITE sip:2222222222@3.3.3.3 SIP/2.0
Max-Forwards: 69
Contact: <sip:2222222222@1.1.1.1:5060>
Via: SIP/2.0/UDP 2.2.2.2:5060;rport;branch=z9hG4bK962542606
From: <sip:2222222222@2.2.2.2>;tag=306964556
To: <sip:2222222222@3.3.3.3>
Call-ID: 564312445@2.2.2.2
CSeq: 44 INVITE
User-Agent: Test
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 256

v=0
o=yate 1505463051 1505463051 IN IP4 2.2.2.2
s=SIP Call
c=IN IP4 2.2.2.2
t=0 0
m=audio 26938 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
------
20170915081051.628027 <pbxassist:CALL> Created assistant for 'sip/24'
20170915081051.629268 <sip:INFO> 'udp:0.0.0.0:5060' received 278 bytes SIP message from 3.3.3.3:5060 [0x1cfcef0]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2:5060;rport;branch=z9hG4bK962542606
From: <sip:2222222222@2.2.2.2>;tag=306964556
To: <sip:2222222222@3.3.3.3>
Call-ID: 564312445@2.2.2.2
CSeq: 44 INVITE
Server: Invoice SIP-server
Content-Length: 0

------
20170915081051.639711 <sip:INFO> 'udp:0.0.0.0:5060' received 344 bytes SIP message from 3.3.3.3:5060 [0x1cfcef0]
------
SIP/2.0 300 Multiple choices
Via: SIP/2.0/UDP 2.2.2.2:5060;rport;branch=z9hG4bK962542606
From: <sip:2222222222@2.2.2.2>;tag=306964556
To: <sip:2222222222@3.3.3.3>
Call-ID: 564312445@2.2.2.2
CSeq: 44 INVITE
Contact: <sip:2222222222@188.115.195.177:5060>
                                               TG:  0
Server: Invoice SIP-server
Content-Length: 0

------
20170915081051.642651 <sip/24:ALL> YateSIPConnection::hangup() state=1 trans=0x7f8edc015c40 error='(null)' code=300 reason='Multiple choices' [0x7f8edc00ee80]
20170915081051.642724 <ALL> Cleaning up RTP 0x7f8edc00a950 [0x7f8edc00a7a0]
20170915081051.642781 <sip/23:ALL> YateSIPConnection::disconnected() 'Multiple choices' [0x7f8ef0012020]
20170915081051.642817 <sip/24:ALL> YateSIPConnection::~YateSIPConnection() [0x7f8edc00ee80]
20170915081051.642884 <sip:INFO> 'udp:0.0.0.0:5060' sending 'ACK sip:2222222222@3.3.3.3' 0x7f8ef4028160 to 3.3.3.3:5060 [0x1cfcef0]
------
ACK sip:2222222222@3.3.3.3 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;rport;branch=z9hG4bK962542606
From: <sip:2222222222@2.2.2.2>;tag=306964556
To: <sip:2222222222@3.3.3.3>
Call-ID: 564312445@2.2.2.2
CSeq: 44 ACK
Max-Forwards: 69
Contact: <sip:2222222222@1.1.1.1:5060>
User-Agent: Test
Content-Length: 0

------
20170915081051.643742 <INFO> Could not classify call from '2222222222', wasted 104 usec
20170915081051.643830 <INFO> Could not route call to '11111111' in context 'default', wasted 22 usec
20170915081051.643857 <pbxassist:MILD> Chan 'sip/23' operation failed: no route
20170915081051.643884 <sip/23:ALL> YateSIPConnection::hangup() state=0 trans=0x7f8ef402c1e0 error='noanswer' code=487 reason='Multiple choices' [0x7f8ef0012020]
20170915081051.643966 <sip/23:ALL> YateSIPConnection::~YateSIPConnection() [0x7f8ef0012020]
20170915081051.648052 <sip:INFO> 'udp:0.0.0.0:5060' sending code 487 0x7f8eec0175a0 to 1.1.1.1:5060 [0x1cfcef0]
------
SIP/2.0 487 Multiple choices
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6a19fc4e;received=1.1.1.1
From: "+2222222222" <sip:+2222222222@1.1.1.1>;tag=as6e0e860a
To: <sip:11111111@2.2.2.2;user=phone>;tag=1459504950
Call-ID: 40d6a8f200621e456cbf8c1431df234a@1.1.1.1:5060
CSeq: 102 INVITE
Server: Test
Contact: <sip:11111111@2.2.2.2:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0

------
20170915081051.648564 <sip:INFO> 'udp:0.0.0.0:5060' received 418 bytes SIP message from 1.1.1.1:5060 [0x1cfcef0]
------
ACK sip:11111111@2.2.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6a19fc4e
Max-Forwards: 70
From: "+2222222222" <sip:+2222222222@1.1.1.1>;tag=as6e0e860a
To: <sip:11111111@2.2.2.2;user=phone>;tag=1459504950
Contact: <sip:+2222222222@1.1.1.1:5060>
Call-ID: 40d6a8f200621e456cbf8c1431df234a@1.1.1.1:5060
CSeq: 102 ACK
User-Agent: Test
Content-Length: 0
95
Linux / Re: abonent information
« Last post by Y2Kot on September 14, 2017, 01:18:25 AM »
I know about this but it doesn't show online users
96
Linux / Re: abonent information
« Last post by Monica Tepelus on September 13, 2017, 03:40:46 AM »
97
Yate users hangout place / Re: 4 digit dialing
« Last post by marian on September 13, 2017, 12:34:50 AM »
You must escape chars used by regexp:

^\([0-9]\{4\}=goto ....

Note that this regexp will match any target starting with 4 digits. If you to match exactly 4 digits you must add a $ char at the end
98
Yate users hangout place / 4 digit dialing
« Last post by epl692 on September 12, 2017, 11:09:41 PM »
I'm trying to setup a regex match for a 4 digit dialed number, and was trying to use
([0-9]{4})
entered into the config as
^([0-9]{4})=goto to_target
and am trying to figure out why it's not working. I'm new to yate, so It might just be a simple error.

Thanks!
epl692
99
Yate users hangout place / Re: yate www
« Last post by Luiscatpone on September 12, 2017, 11:01:23 PM »
Really? Have more details?
100
Linux / abonent information
« Last post by Y2Kot on September 12, 2017, 06:55:00 AM »
Hello everyone, is any command to get information about connected abonents? (IMSI, IMEI etc)
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