PJSIP pickupexten

Hi all,

i have same difficult to pickup call when i using *8 (our another number)
pickupexten .

i´m using dumf_mode=rfc4733 on pjsip endpoints.

Regards,

What exactly happens? What does the console say? What is your configuration?

Hi

In Yealink ip phone appears forbidden

when i tru pickupexten *8 our another number example 40

In cli asterisk appears these
– Executing [s@open:1] NoOp(“DAHDI/9-1”, “”) in new stack
– Executing [s@open:2] Answer(“DAHDI/9-1”, “”) in new stack
– Executing [s@open:3] WaitExten(“DAHDI/9-1”, “0.5”) in new stack
– Timeout on DAHDI/9-1, continuing…
– Executing [s@open:4] BackGround(“DAHDI/9-1”, “bem”) in new stack
– <DAHDI/9-1> Playing ‘bem.slin’ (language ‘pt’)
– Executing [8@open:1] Goto(“DAHDI/9-1”, “default,20027,1”) in new stack
– Goto (default,20027,1)
– Executing [20027@default:1] NoOp(“DAHDI/9-1”, “”) in new stack
– Executing [20027@default:2] Dial(“DAHDI/9-1”, “PJSIP/20027,60”) in new stack
– Called PJSIP/20027
<— Transmitting SIP request (923 bytes) to UDP:192.168.0.22:5064 —>
INVITE sip:20027@192.168.0.22:5064 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;rport;branch=z9hG4bKPj7362ab2f-d5a2-475e-9d6e-e4891af68ef5
From: “93999999” sip:939999999@192.168.0.103;tag=68f44a6a-6219-4515-b895-d6fc3394bcd9
To: sip:20027@192.168.0.22
Contact: sip:6cec5696-9aa7-4bea-93ec-13a2d2473f99@192.168.0.103:5060
Call-ID: c9f76fb8-3b8a-49dd-a96d-04eb0069a8f6
CSeq: 7253 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 275

v=0
o=- 2095767463 2095767463 IN IP6 [fe80::225:90ff:feb5:b457]
s=Asterisk
c=IN IP6 192.168.0.103
t=0 0
m=audio 16056 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Received SIP response (366 bytes) from UDP:192.168.0.22:5064 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.103:5060;rport;branch=z9hG4bKPj7362ab2f-d5a2-475e-9d6e-e4891af68ef5
From: “939999999” sip:939999999@192.168.0.103;tag=68f44a6a-6219-4515-b895-d6fc3394bcd9
To: sip:20027@192.168.0.22
Call-ID: c9f76fb8-3b8a-49dd-a96d-04eb0069a8f6
CSeq: 7253 INVITE
User-Agent: Yealink SIP-T42G 29.73.193.50
Content-Length: 0

<— Received SIP response (592 bytes) from UDP:192.168.0.22:5064 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.103:5060;rport;branch=z9hG4bKPj7362ab2f-d5a2-475e-9d6e-e4891af68ef5
From: “939999999” sip:93999999@192.168.0.103;tag=68f44a6a-6219-4515-b895-d6fc3394bcd9
To: sip:20027@192.168.0.22;tag=1526488934
Call-ID: c9f76fb8-3b8a-49dd-a96d-04eb0069a8f6
CSeq: 7253 INVITE
Contact: sip:20027@192.168.0.22:5064
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T42G 29.73.193.50
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

-- PJSIP/20027-00000007 is ringing

<— Received SIP request (882 bytes) from UDP:192.168.0.23:5063 —>
INVITE sip:*8@192.168.0.29 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.23:5063;branch=z9hG4bK4131634689
From: “Boss_1” sip:20029@192.168.0.29;tag=2989263346
To: sip:*8@192.168.0.29
Call-ID: 2754712594@192.168.0.23
CSeq: 1 INVITE
Contact: sip:20029@192.168.0.23:5063
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T42G 29.73.193.50
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 306

v=0
o=- 20172 20172 IN IP4 192.168.0.23
s=SDP data
c=IN IP4 192.168.0.23
t=0 0
m=audio 11794 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv

<— Transmitting SIP response (444 bytes) to UDP:192.168.0.23:5063 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.23:5063;rport=5063;received=192.168.0.23;branch=z9hG4bK4131634689
Call-ID: 2754712594@192.168.0.23
From: “Boss_1” sip:20029@192.168.0.29;tag=2989263346
To: sip:*8@192.168.0.29;tag=z9hG4bK4131634689
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1467824571/3a335f6424f54aef68acaa4c2c8cdf5e”,opaque=“2f7d7e045a692c1e”,algorithm=md5,qop="auth"
Content-Length: 0

Regards,

Is that all the logging? We challenge for authentication so they authenticate themselves, but there’s no call attempt with that authentication afterwards.

hI

i using my mobile i make call to the office e choose the option in my ivr the phone ring but i´m not near i that extent i try pickupexten from another exten and information in my yealink appears forbidden.

Regards,

Hi ,

See what´s happen when i try pickup the call .

the call going to exten 20051 but i try pickup 40 in exten 20029 .

-- Span 5: Incoming Call, Type = Real, CallingNum = '"939999999"', CallingName = ''
-- Accepting call from '93999999', span 4
-- Executing [s@from-pstn:1] Goto("DAHDI/11-1", "ivr,s,1") in new stack
-- Goto (ivr,s,1)
-- Executing [s@ivr:1] Answer("DAHDI/11-1", "") in new stack
-- Executing [s@ivr:2] Set("DAHDI/11-1", "CHANNEL(language)=pt") in new stack
-- Executing [s@ivr:3] Set("DAHDI/11-1", "TIMEOUT(digit=2)") in new stack
-- Executing [s@ivr:4] Set("DAHDI/11-1", "TIMEOUT(response=4)") in new stack
-- Executing [s@ivr:5] GotoIfTime("DAHDI/11-1", "*,*,31-1,Dec-jan?happy_2014,s,1") in new stack
-- Executing [s@ivr:6] GotoIfTime("DAHDI/11-1", "*,*,17,feb?carnaval_2015,s,1") in new stack
-- Executing [s@ivr:7] GotoIfTime("DAHDI/11-1", "*,*,3,apr?sextasanta,s,1") in new stack
-- Executing [s@ivr:8] GotoIfTime("DAHDI/11-1", "*,*,5,apr?pascoa_2015,s,1") in new stack
-- Executing [s@ivr:9] GotoIfTime("DAHDI/11-1", "*,*,25,apr?25_apr_2015,s,1") in new stack
-- Executing [s@ivr:10] GotoIfTime("DAHDI/11-1", "*,*,1,may?diatrab_2015,s,1") in new stack
-- Executing [s@ivr:11] GotoIfTime("DAHDI/11-1", "*,*,10,jun?closed,s,1") in new stack
-- Executing [s@ivr:12] GotoIfTime("DAHDI/11-1", "*,*,24,jun?closed,s,1") in new stack
-- Executing [s@ivr:13] GotoIfTime("DAHDI/11-1", "*,*,15,aug?closed,s,1") in new stack
-- Executing [s@ivr:14] GotoIfTime("DAHDI/11-1", "*,*,29,aug?closed,s,1") in new stack
-- Executing [s@ivr:16] GotoIfTime("DAHDI/11-1", "*,*,8,Dec?Iclosed,s,1") in new stack
-- Executing [s@ivr:17] GotoIfTime("DAHDI/11-1", "00:00-08:59,mon-fri,*,*?closed,s,1") in new stack
-- Executing [s@ivr:18] GotoIfTime("DAHDI/11-1", "09:00-12:59,mon-fri,*,*?open,s,1") in new stack
-- Goto (open,s,1)
-- Executing [s@open:1] NoOp("DAHDI/11-1", "") in new stack
-- Executing [s@open:2] Answer("DAHDI/11-1", "") in new stack
-- Executing [s@open:3] WaitExten("DAHDI/11-1", "0.5") in new stack
-- Timeout on DAHDI/11-1, continuing...
-- Executing [s@open:4] BackGround("DAHDI/11-1", "bem") in new stack
-- <DAHDI/11-1> Playing 'bem.slin' (language 'pt')

<— Received SIP request (453 bytes) from UDP:192.168.0.22:5064 —>
SUBSCRIBE sip:20027@192.168.0.29 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:5064;branch=z9hG4bK2811220734
From: “Asas” sip:20027@192.168.0.29;tag=3014355053
To: “Asas” sip:20027@192.168.0.29
Call-ID: 1316064961@192.168.0.22
CSeq: 1 SUBSCRIBE
Contact: sip:20027@192.168.0.22:5064
Accept: application/x-as-feature-event+xml
Max-Forwards: 70
User-Agent: Yealink SIP-T42G 29.73.193.50
Expires: 3630
Event: as-feature-event
Content-Length: 0

<— Transmitting SIP response (455 bytes) to UDP:192.168.0.22:5064 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.22:5064;rport=5064;received=192.168.0.22;branch=z9hG4bK2811220734
Call-ID: 1316064961@192.168.0.22
From: “Asas” sip:20027@192.168.0.29;tag=3014355053
To: “Asas” sip:20027@192.168.0.29;tag=z9hG4bK2811220734
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1467881870/ad68db56df3fce07a5c67be107717e64”,opaque=“2ea5b6a57b7f61ec”,algorithm=md5,qop="auth"
Content-Length: 0

<— Received SIP request (723 bytes) from UDP:192.168.0.22:5064 —>
SUBSCRIBE sip:20027@192.168.0.29 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:5064;branch=z9hG4bK1004600557
From: “Asas” sip:20027@192.168.0.29;tag=3014355053
To: “Asas” sip:20027@192.168.0.29
Call-ID: 1316064961@192.168.0.22
CSeq: 2 SUBSCRIBE
Contact: sip:20027@192.168.0.22:5064
Authorization: Digest username=“20027”, realm=“asterisk”, nonce=“1467881870/ad68db56df3fce07a5c67be107717e64”, uri="sip:20027@192.168.0.29", response=“2d18dbdf52499bc5d51fc5b960e2507f”, algorithm=MD5, cnonce=“0a4f113b”, opaque=“2ea5b6a57b7f61ec”, qop=auth, nc=00000001
Accept: application/x-as-feature-event+xml
Max-Forwards: 70
User-Agent: Yealink SIP-T42G 29.73.193.50
Expires: 3630
Event: as-feature-event
Content-Length: 0

<— Transmitting SIP response (305 bytes) to UDP:192.168.0.22:5064 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.22:5064;rport=5064;received=192.168.0.22;branch=z9hG4bK1004600557
Call-ID: 1316064961@192.168.0.22
From: “Asas” sip:20027@192.168.0.29;tag=3014355053
To: “Asas” sip:20027@192.168.0.29;tag=z9hG4bK1004600557
CSeq: 2 SUBSCRIBE
Content-Length: 0

<— Received SIP request (457 bytes) from UDP:192.168.0.23:5063 —>
SUBSCRIBE sip:20029@192.168.0.29 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.23:5063;branch=z9hG4bK1065234239
From: “Boss_1” sip:20029@192.168.0.29;tag=2232688849
To: “Boss_1” sip:20029@192.168.0.29
Call-ID: 3593458654@192.168.0.23
CSeq: 1 SUBSCRIBE
Contact: sip:20029@192.168.0.23:5063
Accept: application/x-as-feature-event+xml
Max-Forwards: 70
User-Agent: Yealink SIP-T42G 29.73.193.50
Expires: 3630
Event: as-feature-event
Content-Length: 0

<— Transmitting SIP response (459 bytes) to UDP:192.168.0.23:5063 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.23:5063;rport=5063;received=192.168.0.23;branch=z9hG4bK1065234239
Call-ID: 3593458654@192.168.0.23
From: “Boss_1” sip:20029@192.168.0.29;tag=2232688849
To: “Boss_1” sip:20029@192.168.0.29;tag=z9hG4bK1065234239
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1467881875/82a9c8b0fa9a9e3b8a836746e1fef77e”,opaque=“2d07a3f7464d0298”,algorithm=md5,qop="auth"
Content-Length: 0

<— Received SIP request (726 bytes) from UDP:192.168.0.23:5063 —>
SUBSCRIBE sip:20029@192.168.0.29 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.23:5063;branch=z9hG4bK568179133
From: “Boss_1” sip:20029@192.168.0.29;tag=2232688849
To: “Boss_1” sip:20029@192.168.0.29
Call-ID: 3593458654@192.168.0.23
CSeq: 2 SUBSCRIBE
Contact: sip:20029@192.168.0.23:5063
Authorization: Digest username=“20029”, realm=“asterisk”, nonce=“1467881875/82a9c8b0fa9a9e3b8a836746e1fef77e”, uri="sip:20029@192.168.0.29", response=“501e077f11bcce78dc7f28d6c8417338”, algorithm=MD5, cnonce=“0a4f113b”, opaque=“2d07a3f7464d0298”, qop=auth, nc=00000001
Accept: application/x-as-feature-event+xml
Max-Forwards: 70
User-Agent: Yealink SIP-T42G 29.73.193.50
Expires: 3630
Event: as-feature-event
Content-Length: 0

<— Transmitting SIP response (307 bytes) to UDP:192.168.0.23:5063 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.23:5063;rport=5063;received=192.168.0.23;branch=z9hG4bK568179133
Call-ID: 3593458654@192.168.0.23
From: “Boss_1” sip:20029@192.168.0.29;tag=2232688849
To: “Boss_1” sip:20029@192.168.0.29;tag=z9hG4bK568179133
CSeq: 2 SUBSCRIBE
Content-Length: 0

-- Executing [1@open:1] Goto("DAHDI/11-1", "default,20051,1") in new stack
-- Goto (default,20051,1)
-- Executing [20051@default:1] NoOp("DAHDI/11-1", "") in new stack
-- Executing [20051@default:2] Dial("DAHDI/11-1", "PJSIP/20051,60") in new stack
-- Called PJSIP/20051

<— Transmitting SIP request (937 bytes) to UDP:192.168.0.21:5064 —>
INVITE sip:20051@192.168.0.21:5064 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;rport;branch=z9hG4bKPj4da21e4e-ad44-4334-8c64-974e8618c324
From: “939999999” sip:939999999@192.168.0.103;tag=40f91ce7-8d19-4e6b-a684-17bd057de622
To: sip:20051@192.168.0.21
Contact: sip:64208d9f-7a3a-425c-94ca-903c0be673fc@192.168.0.103:5060
Call-ID: eb9a0aad-0835-44ea-87ba-319cd3f74dfa
CSeq: 20482 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 288

v=0
o=- 1287024468 1287024468 IN IP6 [fe80::225:90ff:feb5:b457]
s=Asterisk
c=IN IP6 [fe80::225:90ff:feb5:b457]
t=0 0
m=audio 14784 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Received SIP response (365 bytes) from UDP:192.168.0.21:5064 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.103:5060;rport;branch=z9hG4bKPj4da21e4e-ad44-4334-8c64-974e8618c324
From: “939999999” sip:939999999@192.168.0.103;tag=40f91ce7-8d19-4e6b-a684-17bd057de622
To: sip:20051@192.168.0.21
Call-ID: eb9a0aad-0835-44ea-87ba-319cd3f74dfa
CSeq: 20482 INVITE
User-Agent: Yealink SIP-T46G 28.73.0.28
Content-Length: 0

<— Received SIP response (637 bytes) from UDP:192.168.0.21:5064 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.103:5060;rport;branch=z9hG4bKPj4da21e4e-ad44-4334-8c64-974e8618c324
From: “939999999” sip:939999999@192.168.0.103;tag=40f91ce7-8d19-4e6b-a684-17bd057de622
To: sip:20051@192.168.0.21;tag=1544244752
Call-ID: eb9a0aad-0835-44ea-87ba-319cd3f74dfa
CSeq: 20482 INVITE
Contact: sip:20051@192.168.0.21:5064
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T46G 28.73.0.28
Supported: 100rel
Require: 100rel
Rseq: 50
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

<— Transmitting SIP request (385 bytes) to UDP:192.168.0.21:5064 —>
PRACK sip:20051@192.168.0.21:5064 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;rport;branch=z9hG4bKPjbbab7ee3-c743-4a26-98c8-1e618ebb4f7f
From: “939999999” sip:939999999@192.168.0.103;tag=40f91ce7-8d19-4e6b-a684-17bd057de622
To: sip:20051@192.168.0.21;tag=1544244752
Call-ID: eb9a0aad-0835-44ea-87ba-319cd3f74dfa
CSeq: 20483 PRACK
RAck: 50 20482 INVITE
Content-Length: 0

-- PJSIP/20051-00000002 is ringing

<— Received SIP response (415 bytes) from UDP:192.168.0.21:5064 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.103:5060;rport;branch=z9hG4bKPjbbab7ee3-c743-4a26-98c8-1e618ebb4f7f
From: “939999999” sip:9399999999@192.168.0.103;tag=40f91ce7-8d19-4e6b-a684-17bd057de622
To: sip:20051@192.168.0.21;tag=1544244752
Call-ID: eb9a0aad-0835-44ea-87ba-319cd3f74dfa
CSeq: 20483 PRACK
Contact: sip:20051@192.168.0.21:5064
User-Agent: Yealink SIP-T46G 28.73.0.28
Content-Length: 0

<— Received SIP request (456 bytes) from UDP:192.168.0.23:5062 —>
SUBSCRIBE sip:20028@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.23:5062;branch=z9hG4bK1897508615
From: “Boss” sip:20028@192.168.0.103;tag=2557631483
To: “Boss” sip:20028@192.168.0.103
Call-ID: 2593745935@192.168.0.23
CSeq: 1 SUBSCRIBE
Contact: sip:20028@192.168.0.23:5062
Accept: application/x-as-feature-event+xml
Max-Forwards: 70
User-Agent: Yealink SIP-T42G 29.73.193.50
Expires: 3630
Event: as-feature-event
Content-Length: 0

<— Transmitting SIP response (457 bytes) to UDP:192.168.0.23:5062 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.23:5062;rport=5062;received=192.168.0.23;branch=z9hG4bK1897508615
Call-ID: 2593745935@192.168.0.23
From: “Boss” sip:20028@192.168.0.103;tag=2557631483
To: “Boss” sip:20028@192.168.0.103;tag=z9hG4bK1897508615
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1467881877/028ffefa475038ef2e44bf7c7935ae98”,opaque=“456aeac12c49260f”,algorithm=md5,qop="auth"
Content-Length: 0

<— Received SIP request (727 bytes) from UDP:192.168.0.23:5062 —>
SUBSCRIBE sip:20028@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.23:5062;branch=z9hG4bK1364184216
From: “Boss” sip:20028@192.168.0.103;tag=2557631483
To: “Boss” sip:20028@192.168.0.103
Call-ID: 2593745935@192.168.0.23
CSeq: 2 SUBSCRIBE
Contact: sip:20028@192.168.0.23:5062
Authorization: Digest username=“20028”, realm=“asterisk”, nonce=“1467881877/028ffefa475038ef2e44bf7c7935ae98”, uri="sip:20028@192.168.0.103", response=“5174563ee1e392f286c4d333a83135c2”, algorithm=MD5, cnonce=“0a4f113b”, opaque=“456aeac12c49260f”, qop=auth, nc=00000001
Accept: application/x-as-feature-event+xml
Max-Forwards: 70
User-Agent: Yealink SIP-T42G 29.73.193.50
Expires: 3630
Event: as-feature-event
Content-Length: 0

<— Transmitting SIP response (307 bytes) to UDP:192.168.0.23:5062 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.23:5062;rport=5062;received=192.168.0.23;branch=z9hG4bK1364184216
Call-ID: 2593745935@192.168.0.23
From: “Boss” sip:20028@192.168.0.103;tag=2557631483
To: “Boss” sip:20028@192.168.0.103;tag=z9hG4bK1364184216
CSeq: 2 SUBSCRIBE
Content-Length: 0

<— Received SIP request (885 bytes) from UDP:192.168.0.23:5060 —>
INVITE sip:40@192.168.0.29 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.23:5060;branch=z9hG4bK2547433866
From: “192.168.0.23” sip:192.168.0.23@192.168.0.23;tag=1160900264
To: sip:40@192.168.0.29
Call-ID: 2474139406@192.168.0.23
CSeq: 1 INVITE
Contact: sip:192.168.0.23@192.168.0.23
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T42G 29.73.193.50
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 294

v=0
o=- 20175 20175 IN IP4 192.168.0.23
s=SDP data
c=IN IP4 192.168.0.23
t=0 0
m=audio 11800 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<— Transmitting SIP response (457 bytes) to UDP:192.168.0.23:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.23:5060;rport=5060;received=192.168.0.23;branch=z9hG4bK2547433866
Call-ID: 2474139406@192.168.0.23
From: “192.168.0.23” sip:192.168.0.23@192.168.0.23;tag=1160900264
To: sip:40@192.168.0.29;tag=z9hG4bK2547433866
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1467881882/9ad69c06d18ebe1d4c9728ce927a7adb”,opaque=“390a31a41aee4785”,algorithm=md5,qop="auth"
Content-Length: 0

<— Received SIP request (280 bytes) from UDP:192.168.0.23:5060 —>
ACK sip:40@192.168.0.29 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.23:5060;branch=z9hG4bK2547433866
From: “192.168.0.23” sip:192.168.0.23@192.168.0.23;tag=1160900264
To: sip:40@192.168.0.29;tag=z9hG4bK2547433866
Call-ID: 2474139406@192.168.0.23
CSeq: 1 ACK
Content-Length: 0

<— Received SIP request (1156 bytes) from UDP:192.168.0.23:5060 —>
INVITE sip:40@192.168.0.29 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.23:5060;branch=z9hG4bK2321732086
From: “192.168.0.23” sip:192.168.0.23@192.168.0.23;tag=1160900264
To: sip:40@192.168.0.29
Call-ID: 2474139406@192.168.0.23
CSeq: 2 INVITE
Contact: sip:192.168.0.23@192.168.0.23
Authorization: Digest username=“SIP Phone”, realm=“asterisk”, nonce=“1467881882/9ad69c06d18ebe1d4c9728ce927a7adb”, uri="sip:40@192.168.0.29", response=“e6752cb4f4f8980e2f5b8d947ea6ec95”, algorithm=MD5, cnonce=“0a4f113b”, opaque=“390a31a41aee4785”, qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T42G 29.73.193.50
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 294

v=0
o=- 20175 20175 IN IP4 192.168.0.23
s=SDP data
c=IN IP4 192.168.0.23
t=0 0
m=audio 11800 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<— Transmitting SIP response (457 bytes) to UDP:192.168.0.23:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.23:5060;rport=5060;received=192.168.0.23;branch=z9hG4bK2321732086
Call-ID: 2474139406@192.168.0.23
From: “192.168.0.23” sip:192.168.0.23@192.168.0.23;tag=1160900264
To: sip:40@192.168.0.29;tag=z9hG4bK2321732086
CSeq: 2 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1467881882/9ad69c06d18ebe1d4c9728ce927a7adb”,opaque=“255db10e2fff6f32”,algorithm=md5,qop="auth"
Content-Length: 0

<— Received SIP request (280 bytes) from UDP:192.168.0.23:5060 —>
ACK sip:40@192.168.0.29 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.23:5060;branch=z9hG4bK2321732086
From: “192.168.0.23” sip:192.168.0.23@192.168.0.23;tag=1160900264
To: sip:40@192.168.0.29;tag=z9hG4bK2321732086
Call-ID: 2474139406@192.168.0.23
CSeq: 2 ACK
Content-Length: 0

<— Received SIP request (1156 bytes) from UDP:192.168.0.23:5060 —>
INVITE sip:40@192.168.0.29 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.23:5060;branch=z9hG4bK4222269279
From: “192.168.0.23” sip:192.168.0.23@192.168.0.23;tag=1160900264
To: sip:40@192.168.0.29
Call-ID: 2474139406@192.168.0.23
CSeq: 3 INVITE
Contact: sip:192.168.0.23@192.168.0.23
Authorization: Digest username=“SIP Phone”, realm=“asterisk”, nonce=“1467881882/9ad69c06d18ebe1d4c9728ce927a7adb”, uri="sip:40@192.168.0.29", response=“e6752cb4f4f8980e2f5b8d947ea6ec95”, algorithm=MD5, cnonce=“0a4f113b”, opaque=“255db10e2fff6f32”, qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T42G 29.73.193.50
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 294

v=0
o=- 20175 20175 IN IP4 192.168.0.23
s=SDP data
c=IN IP4 192.168.0.23
t=0 0
m=audio 11800 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<— Transmitting SIP response (457 bytes) to UDP:192.168.0.23:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.23:5060;rport=5060;received=192.168.0.23;branch=z9hG4bK4222269279
Call-ID: 2474139406@192.168.0.23
From: “192.168.0.23” sip:192.168.0.23@192.168.0.23;tag=1160900264
To: sip:40@192.168.0.29;tag=z9hG4bK4222269279
CSeq: 3 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1467881882/9ad69c06d18ebe1d4c9728ce927a7adb”,opaque=“0baa5dbb4daa02c0”,algorithm=md5,qop="auth"
Content-Length: 0

Regards,

Your new information does not show *8 being used at all or pickup being done. It shows a Yealink trying to call ‘40’. I don’t understand what you’re trying to do exactly.

Hi

I dont use *8 i change for 40 .

i try *8 doesn´t work if i change to 40 it´s the same doesn´t work

Regards,

Don’t change the conditions of what you’re trying to do. Provide the following:

  1. The complete call flow including technologies in use and what they’re doing
  2. What your configuration is for pickupexten, call group, and pickup group

Hi

in my chan_dahdi.conf , group 1 , calgroup=1 , pickupgroup=1
in featrures.conf , pickupgroup= *8 still like original
when i use pickupexten in pjsip endpoints the *8 not pickupexten
but in sip extensions work perfect .

Regards,

Have you configured in pjsip.conf a pickup group and call group for the endpoint?

Hi ,

I forget to post call_group an pickup_group in endpoints.

Regards,