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.


#18

It’s possible I’m misunderstanding the question here, but I think the issue is that you need the first key to be set to Default (or Line)

and the “Key Mode” to be set to “Account Mode” (which will allow you to make multiple calls without dedicating multiple keys).


#19

1 and 2 should not be used if you want two calls on the phone.


#20

So much information. Here’s the short version of what we’re all trying to tell you:

By default, you need to have one button assigned to your account for each call you want to have. This is called “Line Mode” in the “Virtual Multi-Purpose Keys Settings” screen.

If you change this to “Account Mode” then you can have one button assigned per extension and you can make as many calls as you have physical lines available on the phone (GXP21X0 have different numbers).

So you have a choice:

  • Make two buttons be “Default” mode and assign both to account 1 and assign buttons to “Line Mode” in the settings page
    or
  • Make one button be “Default” mode and assign it to account 1 and assign buttons to “Account Mode” in the settings page