DP720 impossible to send DTMF info with 183 Progress


#1

Hi Support Team.
We’re facing a problem with a couple of DP720 cordless phones. We tried to install the latest firmware (1.0.7.5) but with no changes.
The problem seems triggered by calls with INVITE not acknowledged but stil in 183 Progress. In this situation the caller could listen to an IVR but it’s not allowed to interact with it.
I’m reporting here the SIP transaction who is generating the issue. We’re using Asterisk PBX 13.22.0 with PJSIP enabled.

<— Received SIP request (1224 bytes) from UDP:192.168.151.21:5060 —>
INVITE sip:800800800@pbx SIP/2.0
Via: SIP/2.0/UDP 192.168.151.21:5060;branch=z9hG4bK867812010;rport
From: “Banco” sip:91@pbx;tag=1149817573
To: <sip: 800800800@pbx>
Call-ID: 287692257-5060-45@BJC.BGI.BFB.CB
CSeq: 440 INVITE
Contact: “Banco” sip:91@192.168.151.21:5060>
Max-Forwards: 70
User-Agent: Grandstream DP750 1.0.7.5
Privacy: none
P-Preferred-Identity: “Banco” sip:91@pbx>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 478

v=0
o=91 8000 8000 IN IP4 192.168.151.21
s=SIP Call
c=IN IP4 192.168.151.21
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 123 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:2 G726-32/8000
a=rtpmap:123 opus/48000/2
a=fmtp:123 maxplaybackrate=16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54

<— Transmitting SIP response (557 bytes) to UDP:192.168.151.21:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.151.21:5060;rport=5060;received=192.168.151.21;branch=z9hG4bK867812010
Call-ID: 287692257-5060-45@BJC.BGI.BFB.CB
From: “Banco” sip:91@pbx>;tag=1149817573
To: <sip: 800800800@pbx>;tag=z9hG4bK867812010
CSeq: 440 INVITE
WWW-Authenticate: Digest realm=“pbx”,nonce=“1552464357/c2da926a2b6f84ad8f25c5dba8c32c”,opaque=“0f06bc0a06e0a556”,algorithm=md5,qop=“auth”
Server: Asterisk PBX 13.22.0
Content-Length: 0

<— Received SIP request (357 bytes) from UDP:192.168.151.21:5060 —>
ACK sip: 800800800@pbx SIP/2.0
Via: SIP/2.0/UDP 192.168.151.21:5060;branch=z9hG4bK867812010;rport
From: “Banco” sip:91@pbx>;tag=1149817573
To: <sip: 800800800@pbx>;tag=z9hG4bK867812010
Call-ID: 287692257-5060-45@BJC.BGI.BFB.CB
CSeq: 440 ACK
Content-Length: 0

<— Received SIP request (1553 bytes) from UDP:192.168.151.21:5060 —>
INVITE sip: 800800800@pbx SIP/2.0
Via: SIP/2.0/UDP 192.168.151.21:5060;branch=z9hG4bK180243985;rport
From: “Banco” sip:91@pbx>;tag=1149817573
To: <sip: 800800800@pbx>
Call-ID: 287692257-5060-45@BJC.BGI.BFB.CB
CSeq: 441 INVITE
Contact: “Banco” sip:91@192.168.151.21:5060>
Authorization: Digest username=“000D628521”, realm=“pbx”, nonce=“1552464357/c2da926a2b6f84ad8caf25c5dba8c32c”, uri=“sip: 800800800@pbx”, response=“e56a2416a0a8f7e2283cdfb8b688b7c9”, algorithm=md5, cnonce=“15839598”, opaque=“0f06bc0a06e0ab76”, qop=auth, nc=00000001
Max-Forwards: 70
User-Agent: Grandstream DP750 1.0.7.5
Privacy: none
P-Preferred-Identity: “Banco” sip:91@pbx>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 478

v=0
o=91 8000 8000 IN IP4 192.168.151.21
s=SIP Call
c=IN IP4 192.168.151.21
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 123 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:2 G726-32/8000
a=rtpmap:123 opus/48000/2
a=fmtp:123 maxplaybackrate=16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54

<— Transmitting SIP response (358 bytes) to UDP:192.168.151.21:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.151.21:5060;rport=5060;received=192.168.151.21;branch=z9hG4bK180243985
Call-ID: 287692257-5060-45@BJC.BGI.BFB.CB
From: “Banco” sip:91@pbx>;tag=1149817573
To: <sip: 800800800@pbx>
CSeq: 441 INVITE
Server: Asterisk PBX 13.22.0
Content-Length: 0

<— Transmitting SIP response (866 bytes) to UDP:192.168.151.21:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.151.21:5060;rport=5060;received=192.168.151.21;branch=z9hG4bK180243985
Call-ID: 287692257-5060-45@BJC.BGI.BFB.CB
From: “Banco” sip:91@pbx>;tag=1149817573
To: <sip: 800800800@pbx>;tag=78a232ee-eaf2-4553-98c8-ca3c68106b6c
CSeq: 441 INVITE
Server: Asterisk PBX 13.22.0
Contact: sip:192.168.151.11:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Content-Type: application/sdp
Content-Length: 276

v=0
o=- 8000 8002 IN IP4 192.168.151.11
s=Asterisk
c=IN IP4 192.168.151.11
t=0 0
m=audio 6828 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
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=maxptime:150
a=sendrecv

<— Transmitting SIP response (866 bytes) to UDP:192.168.151.21:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.151.21:5060;rport=5060;received=192.168.151.21;branch=z9hG4bK180243985
Call-ID: 287692257-5060-45@BJC.BGI.BFB.CB
From: “Banco” sip:91@pbx>;tag=1149817573
To: <sip: 800800800@pbx>;tag=78a232ee-eaf2-4553-98c8-ca3c68106b6c
CSeq: 441 INVITE
Server: Asterisk PBX 13.22.0
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: sip:192.168.151.11:5060>
Content-Type: application/sdp
Content-Length: 276

v=0
o=- 8000 8002 IN IP4 192.168.151.11
s=Asterisk
c=IN IP4 192.168.151.11
t=0 0
m=audio 6828 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
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=maxptime:150
a=sendrecv

Seems the phone is waiting for a proper INVITE ACK before enabling DTMF.
We have a set of GXP2135 running fine with the same SIP transactions.

Thank you and best regards,
Marco Signorini.


#2

ciao Marco,
hai provato a settare DTMF in RFC2833?
Dovrebbe essere su account/audio settings.


#3

Ciao Damiano. Si. All’inizio pensavo fosse proprio per un problema di negoziazione, ma dopo aver fatto un po’ di debug e aver visto i dialoghi SIP, confrontati con altre chiamate che terminano con 200 OK all’INVITE, mi convinco sempre di piu’ che il problema sia relativo a un piccolo baco nella user interface del telefono. Mentre la chiamata e’ in progress, disabilitano funzionalita’ quali transfer e DTMF…

Grazie
Marco.


#4

prova settando come unico codec PCMA e DTM in RFC2833, io non ho mai avuto problemi


#5

Purtroppo il problema e’ piu subdolo.
Non e’ un problema di trasmissione tra DP720 e centralino ma, piuttosto, il tastierino numerico e’ completamente disabilitato e, quindi, anche schiacciando i numeri questi non vengono neanche visualizzati sul display del telefono.

Questo problema avviene solo ed esclusivamente chiamando uno specifico numero verde (abbiamo un troncone nativo VoIP) che parte con un IVR senza effettuare un vero e proprio INVITE ACK ma rimanendo in 180 Progress fino alla selezione di una opzione che, pero’, e’ impossibile dato che i tasti sono disabilitati sul cordless…

Per ora ho temporaneamente risolto facendosi richiamare dal centralino. In questa situazione Asterisk invia al telefono un INVITE ACK e, di conseguenza, il telefono sblocca la tastiera… quindi i DTMF vengono inviati tramite RFC2833 e tutto funziona… pero’ e’ un trucco che vorrei evitare.

Grazie.
Marco.


#6

avevo avuto un problema simile, verifica questo parametro -> Caller ID Display e Callee ID Display.
Proprio in caso di numeri verdi ricordo mi bloccava la tastiera (l’assurdo era che lo faceva solo con alcuni numeri verdi).

n.b.: il nome potrebbe differenziarsi leggermente da modello a modello


#7

Purtroppo non mi sembra di vedere questo parametro o parametri simili su questo modello :cry:

Tu sai se e’ possibile segnalare a GS come bug?


#8

certo, scrivi qui


#9

WOW. Grazie. Questo mi era sfuggito.
Ci provo.
Buona giornata.