I can not have a second call on my ip phone


#1

Hello @everyone,

On my Asterisk 13.23.0, I have this verbose information in the CLI Console

<--- Received SIP response (542 bytes) from UDP:192.168.2.48:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPjcbcabfca-2f98-47c2-b8b3-6c5d55529079
From: "Jane Doe" <sip:116@ip_address>;tag=b8061174-816c-4e3e-9876-d563c649919b
To: <sip:100@192.168.2.48>;tag=950722919
Call-ID: a8ec40ba-8049-4afb-ab18-26e2ef7dcf20
CSeq: 9181 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.9.108
Warning: 399 GS "All lines are in use"
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

I have five types of grandstream (GXP1625, GXP1630, GXP2130, GXP2140, GXP2170) and all made the same thing. I can not have a second call on it. Do you know where is the problem, is it on one of parameters in these Grandstreams, what is the matter ?


#2

You probably configured all lines as something else.
You need leave at least 2 lines as free (1 is account), 2 must be free or added normal account).
Lines determine how many calls you can have on phone.


#3

@marcin is correct here, which took us a while to figure out, coming from Cisco phones. On a Cisco phone each line button can actually handle two calls. So on a four button phone you could leave one as line, change one to park, and the other two to BLF monitor the first two parking lot slots.

On a Grandstream phone, you need one line button per call or else the phone will be considered busy by the system.


#4

Multi-Purpose Keys

Virtual Multi-Purpose Keys


#5

As mentioned, you only have a single “line” button. The rest are BLF buttons. You need to change one of your multi-purpose keys to be “Default” or “line” and specify the account to be the same as your primary (Account 1). This will let you make/take a second call.


#6

1 one is even worse, all BLF - that phone will not work.
2 lines are minimum if you plan use transfer.


#7

Now I change the Virtual Multi-Purpose Keys by this one above. And now I can have a second call on my ip phone. But I can not have a second call by the incoming calls. Is it my asterisk (dialplan) or the Grandstream ?


#8

Can be both.
Phone:
account x -> call settings
image

Setting -> call features
image

this will allow 2 call on phone.


#9

This is what I have as configuration on my ip phone but nothing change; This my CLI of my Asterisk if this is can help to fix that issue:

    <--- Received SIP request (1204 bytes) from UDP:192.168.40.50:5060 --->
INVITE sip:028992018@ip_address SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK616006517;rport
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
CSeq: 350 INVITE
Contact: "Jane Doe" <sip:116@192.168.40.50:5060>
Max-Forwards: 70
User-Agent: Grandstream GXP1625 1.0.4.128
Privacy: none
P-Preferred-Identity: "Jane Doe" <sip:116@ip_address>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=00-1D-AA-A8-19-30
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=00-0B-82-76-3F-97
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   403

v=0
o=116 8000 8000 IN IP4 192.168.40.50
s=SIP Call
c=IN IP4 192.168.40.50
t=0 0
m=audio 5076 RTP/AVP 0 8 18 4 2 9 97 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<--- Transmitting SIP response (492 bytes) to UDP:192.168.40.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.40.50:5060;rport=5060;received=192.168.40.50;branch=z9hG4bK616006517
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>;tag=z9hG4bK616006517
CSeq: 350 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1539180496/b30db263bc407c62f5fdbb74e284d316",opaque="777b70bd484d1d02",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.23.0
Content-Length:  0


<--- Received SIP request (297 bytes) from UDP:192.168.40.50:5060 --->
ACK sip:028992018@ip_address SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK616006517;rport
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>;tag=z9hG4bK616006517
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
CSeq: 350 ACK
Content-Length: 0


<--- Received SIP request (1477 bytes) from UDP:192.168.40.50:5060 --->
INVITE sip:028992018@ip_address SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK2143641284;rport
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
CSeq: 351 INVITE
Contact: "Jane Doe" <sip:116@192.168.40.50:5060>
Authorization: Digest username="116", realm="asterisk", nonce="1539180496/b30db263bc407c62f5fdbb74e284d316", uri="sip:028992018@ip_address", response="f666b74aaf71ca5a273b7c306bf5c85b", algorithm=md5, cnonce="02528634", opaque="777b70bd484d1d02", qop=auth, nc=00000001
Max-Forwards: 70
User-Agent: Grandstream GXP1625 1.0.4.128
Privacy: none
P-Preferred-Identity: "Jane Doe" <sip:116@ip_address>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=00-1D-AA-A8-19-30
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=00-0B-82-76-3F-97
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   403

v=0
o=116 8000 8000 IN IP4 192.168.40.50
s=SIP Call
c=IN IP4 192.168.40.50
t=0 0
m=audio 5076 RTP/AVP 0 8 18 4 2 9 97 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

  == Setting global variable 'SIPDOMAIN' to 'ip_address'
<--- Transmitting SIP response (319 bytes) to UDP:192.168.40.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.40.50:5060;rport=5060;received=192.168.40.50;branch=z9hG4bK2143641284
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>
CSeq: 351 INVITE
Server: Asterisk PBX 13.23.0
Content-Length:  0


    -- Executing [028992018@from-internal:1] NoOp("PJSIP/116-0000042b", "Outgoing Call from "Jane Doe" <116> to 028992018") in new stack
    -- Executing [028992018@from-internal:2] Verbose("PJSIP/116-0000042b", "Call start time: 2018-10-10 16:08:16") in new stack
Call start time: 2018-10-10 16:08:16
    -- Executing [028992018@from-internal:3] Set("PJSIP/116-0000042b", "CDR(calldate)=2018-10-10 16:08:16") in new stack
    -- Executing [028992018@from-internal:4] Set("PJSIP/116-0000042b", "CDR(useragent)=Jane Doe") in new stack
    -- Executing [028992018@from-internal:5] Set("PJSIP/116-0000042b", "POSTE=116") in new stack
    -- Executing [028992018@from-internal:6] NoOp("PJSIP/116-0000042b", "SendedCID = 116") in new stack
    -- Executing [028992018@from-internal:7] Set("PJSIP/116-0000042b", "CALLERID(num)=042770677") in new stack
    -- Executing [028992018@from-internal:8] NoOp("PJSIP/116-0000042b", "SendedCID = 042770677") in new stack
    -- Executing [028992018@from-internal:9] Set("PJSIP/116-0000042b", "NOW=2018_10_10_16_08_16") in new stack
    -- Executing [028992018@from-internal:10] System("PJSIP/116-0000042b", "echo "--appel_sortant --- callerid : 042770677 ---- 2018_10_10_16_08_16 ----" >> /var/spool/asterisk/log/debug.txt") in new stack
    -- Executing [028992018@from-internal:11] Set("PJSIP/116-0000042b", "REC_FILE_NAME=OUT_2018_10_10_16_08_16_028992018_116.wav") in new stack
    -- Executing [028992018@from-internal:12] Answer("PJSIP/116-0000042b", "") in new stack
<--- Transmitting SIP response (903 bytes) to UDP:192.168.40.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.40.50:5060;rport=5060;received=192.168.40.50;branch=z9hG4bK2143641284
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>;tag=dba3540b-532b-4289-ab92-fafb39d568bd
CSeq: 351 INVITE
Server: Asterisk PBX 13.23.0
Contact: <sip:ip_address:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   320

v=0
o=- 8000 8002 IN IP4 ip_address
s=Asterisk
c=IN IP4 ip_address
t=0 0
m=audio 14994 RTP/AVP 0 8 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (537 bytes) from UDP:192.168.40.50:5060 --->
ACK sip:ip_address:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK152621513;rport
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>;tag=dba3540b-532b-4289-ab92-fafb39d568bd
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
CSeq: 351 ACK
Contact: <sip:116@192.168.40.50:5060>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.128
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- Executing [028992018@from-internal:13] NoOp("PJSIP/116-0000042b", "n° Poste = 116") in new stack
    -- Executing [028992018@from-internal:14] MixMonitor("PJSIP/116-0000042b", "OUT_2018_10_10_16_08_16_028992018_116.wav,b V(1)") in new stack
    -- Executing [028992018@from-internal:15] Goto("PJSIP/116-0000042b", "SetProv") in new stack
    -- Goto (from-internal,028992018,17)
    -- Executing [028992018@from-internal:17] Set("PJSIP/116-0000042b", "PROV2USE=BelgiumVoIP") in new stack
    -- Executing [028992018@from-internal:18] NoOp("PJSIP/116-0000042b", "Provider to use : BelgiumVoIP") in new stack
    -- Executing [028992018@from-internal:19] GotoIf("PJSIP/116-0000042b", "0?WideVoIP") in new stack
    -- Executing [028992018@from-internal:20] GotoIf("PJSIP/116-0000042b", "0?Selfone") in new stack
    -- Executing [028992018@from-internal:21] GotoIf("PJSIP/116-0000042b", "1?BelgiumVoIP") in new stack
    -- Goto (from-internal,028992018,23)
    -- Executing [028992018@from-internal:23] Set("PJSIP/116-0000042b", "NUM2DIAL=028992018") in new stack
    -- Executing [028992018@from-internal:24] System("PJSIP/116-0000042b", "echo "--BelgiumVoIP  --- callerid : 042770677 ---- 2018_10_10_16_08_17 ----" >> /var/spool/asterisk/log/debug.txt") in new stack
  == Begin MixMonitor Recording PJSIP/116-0000042b
    -- Executing [028992018@from-internal:25] NoOp("PJSIP/116-0000042b", "CD BelgiumVoIP") in new stack
    -- Executing [028992018@from-internal:26] Dial("PJSIP/116-0000042b", "PJSIP/028992018@belgium-voip,60") in new stack
    -- Called PJSIP/028992018@belgium-voip
<--- Transmitting SIP request (1057 bytes) to UDP:188.66.8.19:5060 --->
INVITE sip:028992018@voip.belgium-voip.com:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPjf84a1734-56b3-4493-8d06-f8c04d473d6e
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>
Contact: <sip:asterisk@ip_address:5060>
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22269 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Type: application/sdp
Content-Length:   353

v=0
o=- 351740866 351740866 IN IP4 ip_address
s=Asterisk
c=IN IP4 ip_address
t=0 0
m=audio 11458 RTP/AVP 0 8 3 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (563 bytes) from UDP:188.66.8.19:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP ip_address:5060;rport=39748;branch=z9hG4bKPjf84a1734-56b3-4493-8d06-f8c04d473d6e;received=ip_address
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>;tag=6a4cc70e519e85e8bc5c654eeaf70770.2786
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22269 INVITE
Proxy-Authenticate: Digest realm="voip.belgium-voip.com", nonce="W74I/Vu+B9GTBdq3AS2Zd7ivAKcgC/4w"
Server: Enswitch SIP proxy
Content-Length: 0


<--- Transmitting SIP request (469 bytes) to UDP:188.66.8.19:5060 --->
ACK sip:028992018@voip.belgium-voip.com:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPjf84a1734-56b3-4493-8d06-f8c04d473d6e
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>;tag=6a4cc70e519e85e8bc5c654eeaf70770.2786
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22269 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0


<--- Transmitting SIP request (1274 bytes) to UDP:188.66.8.19:5060 --->
INVITE sip:028992018@voip.belgium-voip.com:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPja839b4d7-d17a-4c3f-adf7-f2603a641d6d
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>
Contact: <sip:asterisk@ip_address:5060>
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22270 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Proxy-Authorization: Digest username="0427714121", realm="voip.belgium-voip.com", nonce="W74I/Vu+B9GTBdq3AS2Zd7ivAKcgC/4w", uri="sip:028992018@voip.belgium-voip.com:5060", response="f030afed2b84de6cfd9720541e4409be"
Content-Type: application/sdp
Content-Length:   353

v=0
o=- 351740866 351740866 IN IP4 ip_address
s=Asterisk
c=IN IP4 ip_address
t=0 0
m=audio 11458 RTP/AVP 0 8 3 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (430 bytes) from UDP:188.66.8.19:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ip_address:5060;rport=39748;branch=z9hG4bKPja839b4d7-d17a-4c3f-adf7-f2603a641d6d;received=ip_address
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22270 INVITE
Server: Enswitch SIP proxy
Content-Length: 0


<--- Received SIP response (655 bytes) from UDP:188.66.8.19:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ip_address:5060;received=ip_address;rport=39748;branch=z9hG4bKPja839b4d7-d17a-4c3f-adf7-f2603a641d6d
Record-Route: <sip:188.66.8.19;lr=on>
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>;tag=as1d37edb5
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22270 INVITE
Server: 3StarsNet VoipSwitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028992018@188.66.8.52:5060>
Content-Length: 0


    -- PJSIP/belgium-voip-0000042c is ringing
    -- PJSIP/belgium-voip-0000042c is ringing
<--- Received SIP request (1302 bytes) from UDP:188.66.8.19:5060 --->
INVITE sip:028992018@ip_address:39748 SIP/2.0
Record-Route: <sip:188.66.8.19;lr=on>
Via: SIP/2.0/UDP 188.66.8.19;branch=z9hG4bK6092.1c17d19bd75143ad831581f6251e186f.0
Via: SIP/2.0/UDP 188.66.8.52:5060;received=188.66.8.52;branch=z9hG4bK208f6491;rport=5060
Max-Forwards: 69
From: "Jane Doe" <sip:042770677@188.66.8.52>;tag=as53b1f9bd
To: <sip:028992018@ip_address:39748>
Contact: <sip:042770677@188.66.8.52:5060>
Call-ID: 327ced0e1f5b5bf9347129773ecf5b4c@188.66.8.52:5060
CSeq: 102 INVITE
User-Agent: 3StarsNet VoipSwitch
Date: Wed, 10 Oct 2018 14:08:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Enswitch-Uniqueid: 1539180497.140627
Diversion: <sip:028992018@ast29>
Content-Type: application/sdp
Content-Length: 370
X-Enswitch-RURI: sip:028992018@ip_address:39748
X-Enswitch-Source: 188.66.8.52:5060
X-Enswitch-External: yes

v=0
o=root 1420374773 1420374773 IN IP4 188.66.8.27
s=3StarsNet VoipSwitch
c=IN IP4 188.66.8.27
t=0 0
m=audio 14040 RTP/AVP 8 0 18 3 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=sdpmangled:yes

  == Setting global variable 'SIPDOMAIN' to 'ip_address'
<--- Transmitting SIP response (486 bytes) to UDP:188.66.8.19:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 188.66.8.19;rport=5060;received=188.66.8.19;branch=z9hG4bK6092.1c17d19bd75143ad831581f6251e186f.0
Via: SIP/2.0/UDP 188.66.8.52:5060;rport=5060;received=188.66.8.52;branch=z9hG4bK208f6491
Record-Route: <sip:188.66.8.19;lr>
Call-ID: 327ced0e1f5b5bf9347129773ecf5b4c@188.66.8.52:5060
From: "Jane Doe" <sip:042770677@188.66.8.52>;tag=as53b1f9bd
To: <sip:028992018@ip_address>
CSeq: 102 INVITE
Server: Asterisk PBX 13.23.0
Content-Length:  0


    -- Executing [028992018@from-external:1] NoOp("PJSIP/belgium-voip-0000042d", "## Incoming Call from "Jane Doe" <042770677> ##") in new stack
    -- Executing [028992018@from-external:2] Verbose("PJSIP/belgium-voip-0000042d", "Call start time: 2018-10-10 16:08:17") in new stack
Call start time: 2018-10-10 16:08:17
    -- Executing [028992018@from-external:3] Set("PJSIP/belgium-voip-0000042d", "CDR(calldate)=2018-10-10 16:08:17") in new stack
    -- Executing [028992018@from-external:4] Set("PJSIP/belgium-voip-0000042d", "CDR(useragent)=Jane Doe") in new stack
    -- Executing [028992018@from-external:5] Set("PJSIP/belgium-voip-0000042d", "POSTE_EXT=042770677") in new stack
    -- Executing [028992018@from-external:6] Ringing("PJSIP/belgium-voip-0000042d", "") in new stack
    -- Executing [028992018@from-external:7] System("PJSIP/belgium-voip-0000042d", "echo "--appel_sortant --- callerid : 042770677 ---- 2018/10/10 16:08:17 ----" >> /var/spool/asterisk/log/debug.txt") in new stack
<--- Transmitting SIP response (673 bytes) to UDP:188.66.8.19:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 188.66.8.19;rport=5060;received=188.66.8.19;branch=z9hG4bK6092.1c17d19bd75143ad831581f6251e186f.0
Via: SIP/2.0/UDP 188.66.8.52:5060;rport=5060;received=188.66.8.52;branch=z9hG4bK208f6491
Record-Route: <sip:188.66.8.19;lr>
Call-ID: 327ced0e1f5b5bf9347129773ecf5b4c@188.66.8.52:5060
From: "Jane Doe" <sip:042770677@188.66.8.52>;tag=as53b1f9bd
To: <sip:028992018@ip_address>;tag=faf14a39-eea3-4020-89db-ed76ae7324f0
CSeq: 102 INVITE
Server: Asterisk PBX 13.23.0
Contact: <sip:ip_address:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Content-Length:  0


<--- Received SIP response (655 bytes) from UDP:188.66.8.19:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ip_address:5060;received=ip_address;rport=39748;branch=z9hG4bKPja839b4d7-d17a-4c3f-adf7-f2603a641d6d
Record-Route: <sip:188.66.8.19;lr=on>
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>;tag=as1d37edb5
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22270 INVITE
Server: 3StarsNet VoipSwitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028992018@188.66.8.52:5060>
Content-Length: 0


    -- PJSIP/belgium-voip-0000042c is ringing
    -- PJSIP/belgium-voip-0000042c is ringing
    -- Executing [028992018@from-external:8] Set("PJSIP/belgium-voip-0000042d", "REC_FILE_NAME=IN__028992018_042770677.wav") in new stack
    -- Executing [028992018@from-external:9] MixMonitor("PJSIP/belgium-voip-0000042d", "IN__028992018_042770677.wav,b V(1)") in new stack
    -- Executing [028992018@from-external:10] ChanIsAvail("PJSIP/belgium-voip-0000042d", "PJSIP/100,sa") in new stack
    -- Executing [028992018@from-external:11] Set("PJSIP/belgium-voip-0000042d", "PHONESTATUS=2") in new stack
    -- Executing [028992018@from-external:12] Set("PJSIP/belgium-voip-0000042d", "PHONEAVAIL=") in new stack
    -- Executing [028992018@from-external:13] NoOp("PJSIP/belgium-voip-0000042d", "## Status of device is 2 ##") in new stack
    -- Executing [028992018@from-external:14] GotoIf("PJSIP/belgium-voip-0000042d", "0?busy:call") in new stack
    -- Goto (from-external,028992018,18)
    -- Executing [028992018@from-external:18] Dial("PJSIP/belgium-voip-0000042d", ",20") in new stack
[Oct 10 16:08:17] WARNING[20678][C-00000255]: app_dial.c:2281 dial_exec_full: Dial requires an argument (technology/resource)
  == Spawn extension (from-external, 028992018, 18) exited non-zero on 'PJSIP/belgium-voip-0000042d'
<--- Transmitting SIP response (662 bytes) to UDP:188.66.8.19:5060 --->
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 188.66.8.19;rport=5060;received=188.66.8.19;branch=z9hG4bK6092.1c17d19bd75143ad831581f6251e186f.0
Via: SIP/2.0/UDP 188.66.8.52:5060;rport=5060;received=188.66.8.52;branch=z9hG4bK208f6491
Record-Route: <sip:188.66.8.19;lr>
Call-ID: 327ced0e1f5b5bf9347129773ecf5b4c@188.66.8.52:5060
From: "Jane Doe" <sip:042770677@188.66.8.52>;tag=as53b1f9bd
To: <sip:028992018@ip_address>;tag=faf14a39-eea3-4020-89db-ed76ae7324f0
CSeq: 102 INVITE
Server: Asterisk PBX 13.23.0
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Reason: Q.850;cause=0
Content-Length:  0


  == Begin MixMonitor Recording PJSIP/belgium-voip-0000042d
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording PJSIP/belgium-voip-0000042d
<--- Received SIP request (378 bytes) from UDP:188.66.8.19:5060 --->
ACK sip:028992018@ip_address:39748 SIP/2.0
Via: SIP/2.0/UDP 188.66.8.19;branch=z9hG4bK6092.1c17d19bd75143ad831581f6251e186f.0
Max-Forwards: 69
From: "Jane Doe" <sip:042770677@188.66.8.52>;tag=as53b1f9bd
To: <sip:028992018@ip_address>;tag=faf14a39-eea3-4020-89db-ed76ae7324f0
Call-ID: 327ced0e1f5b5bf9347129773ecf5b4c@188.66.8.52:5060
CSeq: 102 ACK
Content-Length: 0


<--- Received SIP response (646 bytes) from UDP:188.66.8.19:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP ip_address:5060;received=ip_address;rport=39748;branch=z9hG4bKPja839b4d7-d17a-4c3f-adf7-f2603a641d6d
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>;tag=as1d37edb5
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22270 INVITE
Server: 3StarsNet VoipSwitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<--- Transmitting SIP request (442 bytes) to UDP:188.66.8.19:5060 --->
ACK sip:028992018@voip.belgium-voip.com:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPja839b4d7-d17a-4c3f-adf7-f2603a641d6d
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>;tag=as1d37edb5
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22270 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0


  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [028992018@from-internal:27] Playback("PJSIP/116-0000042b", "cannot-complete-temp-error") in new stack
    -- <PJSIP/116-0000042b> Playing 'cannot-complete-temp-error.ulaw' (language 'fr')
    -- Executing [028992018@from-internal:28] Goto("PJSIP/116-0000042b", "end") in new stack
    -- Goto (from-internal,028992018,29)
    -- Executing [028992018@from-internal:29] Hangup("PJSIP/116-0000042b", "") in new stack
  == Spawn extension (from-internal, 028992018, 29) exited non-zero on 'PJSIP/116-0000042b'
  == MixMonitor close filestream (mixed)
<--- Transmitting SIP request (423 bytes) to UDP:192.168.40.50:5060 --->
BYE sip:116@192.168.40.50:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPj8b81c975-72bc-42e1-b08d-94d40ca7661c
From: <sip:028992018@ip_address>;tag=dba3540b-532b-4289-ab92-fafb39d568bd
To: "Jane Doe" <sip:116@ip_address>;tag=1043865744
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
CSeq: 3805 BYE
Reason: Q.850;cause=17
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0


  == End MixMonitor Recording PJSIP/116-0000042b
[Oct 10 16:08:21] ERROR[15691]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle.  CDR failed.
[Oct 10 16:08:21] ERROR[15691]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle.  CDR failed.
<--- Received SIP response (534 bytes) from UDP:192.168.40.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPj8b81c975-72bc-42e1-b08d-94d40ca7661c
From: <sip:028992018@ip_address>;tag=dba3540b-532b-4289-ab92-fafb39d568bd
To: "Jane Doe" <sip:116@ip_address>;tag=1043865744
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
CSeq: 3805 BYE
Contact: <sip:116@192.168.40.50:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.128
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP request (575 bytes) from UDP:192.168.2.48:5060 --->
BYE sip:asterisk@ip_address:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.48:5060;branch=z9hG4bK953260347;rport
From: <sip:100@192.168.2.48>;tag=1726513566
To: "John Doe" <sip:042770677@ip_address>;tag=3d9031f7-765c-4152-9625-d9a09dd0a5fc
Call-ID: ce716248-370e-49cb-806f-5f4516f67d9b
CSeq: 28875 BYE
Contact: <sip:100@192.168.2.48:5060>
X-Grandstream-PBX: true
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.9.108
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (356 bytes) to UDP:192.168.2.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.48:5060;rport=5060;received=192.168.2.48;branch=z9hG4bK953260347
Call-ID: ce716248-370e-49cb-806f-5f4516f67d9b
From: <sip:100@192.168.2.48>;tag=1726513566
To: "John Doe" <sip:042770677@ip_address>;tag=3d9031f7-765c-4152-9625-d9a09dd0a5fc
CSeq: 28875 BYE
Server: Asterisk PBX 13.23.0
Content-Length:  0


    -- Channel PJSIP/100-0000042a left 'simple_bridge' basic-bridge <c109bca5-3c7d-4af7-9eea-e69f12027367>
    -- Channel PJSIP/belgium-voip-00000428 left 'simple_bridge' basic-bridge <c109bca5-3c7d-4af7-9eea-e69f12027367>
  == Spawn extension (from-external, 028992018, 18) exited non-zero on 'PJSIP/belgium-voip-00000428'
  == MixMonitor close filestream (mixed)
<--- Transmitting SIP request (478 bytes) to UDP:188.66.8.19:5060 --->
BYE sip:042770677@188.66.8.52:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPj642abaa4-4aaf-432a-a686-6adb4787c1d6
From: <sip:028992018@ip_address>;tag=34f021ef-f396-4a35-a69d-18d4e610f64b
To: "John Doe" <sip:042770677@188.66.8.52>;tag=as08e14896
Call-ID: 0a3369672c687c3725cd9e16769f3618@188.66.8.52:5060
CSeq: 29894 BYE
Route: <sip:188.66.8.19;lr>
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0

#10

Change your Line 2 button to be Account 1.


#11

I made this change on my Grandstream, see the picture bellow

But nothing are changed. But the second internal call work well not the second incoming call.


#12

Same message in the CLI of Asterisk

<--- Received SIP response (646 bytes) from UDP:188.66.8.19:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP ip_address:5060;received=91.180.9.84;rport=39748;branch=z9hG4bKPja5ab9894-3ea0-4f6d-86d6-9bdf7f425a9a
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=a00ff406-1b04-4c76-ad8e-192353d37cba
To: <sip:028992018@voip.belgium-voip.com>;tag=as49385624
Call-ID: e436a057-d61b-41bb-b0b9-c5242c359834
CSeq: 14923 INVITE
Server: 3StarsNet VoipSwitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<--- Transmitting SIP request (442 bytes) to UDP:188.66.8.19:5060 --->
ACK sip:028992018@voip.belgium-voip.com:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPja5ab9894-3ea0-4f6d-86d6-9bdf7f425a9a
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=a00ff406-1b04-4c76-ad8e-192353d37cba
To: <sip:028992018@voip.belgium-voip.com>;tag=as49385624
Call-ID: e436a057-d61b-41bb-b0b9-c5242c359834
CSeq: 14923 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0


  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [028992018@from-internal:27] Playback("PJSIP/116-0000058f", "cannot-complete-temp-error") in new stack
    -- <PJSIP/116-0000058f> Playing 'cannot-complete-temp-error.ulaw' (language 'fr')
    -- Executing [028992018@from-internal:28] Goto("PJSIP/116-0000058f", "end") in new stack
    -- Goto (from-internal,028992018,29)
    -- Executing [028992018@from-internal:29] Hangup("PJSIP/116-0000058f", "") in new stack
  == Spawn extension (from-internal, 028992018, 29) exited non-zero on 'PJSIP/116-0000058f'
  == MixMonitor close filestream (mixed)
<--- Transmitting SIP request (422 bytes) to UDP:192.168.40.50:5060 --->
BYE sip:116@192.168.40.50:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPj75801a86-dac1-4e81-b6d9-8987fc2096b9
From: <sip:028992018@ip_address>;tag=5057bd73-a90d-4abe-b6ec-29f240a5eec7
To: "Jane Doe" <sip:116@ip_address>;tag=898187485
Call-ID: 1193951785-5060-40@BJC.BGI.EA.FA
CSeq: 5567 BYE
Reason: Q.850;cause=17
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0


  == End MixMonitor Recording PJSIP/116-0000058f
[Oct 10 18:12:57] ERROR[15691]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle.  CDR failed.
[Oct 10 18:12:57] ERROR[15691]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle.  CDR failed.
<--- Received SIP response (533 bytes) from UDP:192.168.40.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPj75801a86-dac1-4e81-b6d9-8987fc2096b9
From: <sip:028992018@ip_address>;tag=5057bd73-a90d-4abe-b6ec-29f240a5eec7
To: "Jane Doe" <sip:116@ip_address>;tag=898187485
Call-ID: 1193951785-5060-40@BJC.BGI.EA.FA
CSeq: 5567 BYE
Contact: <sip:116@192.168.40.50:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.128
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

#13

You need to figure out why Asterisk thinks your phone is busy. Can you make a call from the second line? If not, then you need to investigate that.


#14

Maybe call waiting is off on asterisk ?

You can try 2 IP call on this phone to check if cause is asterisk (99% it is)


#15

No, the SIP response from the phone indicates that the problem is at the phone side.


#16

No, check who sent reply.
Server: 3StarsNet VoipSwitch

This response is not FROM phone at all.
It is more like : GXP - asterisk - provider
And provider refuse 2 call ? Maybe there is limiter there ?


#17

Possibly. It depends on how many call paths they (provider) will allow on a single trunk as well as how many will your Asterisk instance allow on the trunk.