Android imsdroid to PSTN call is not working

Question

i am using FreeSWITCH server and integrated with twilio SIP turnk. i am using android imsdroid application for making sip calls. imsdroid to imsdroid call is happening. imsdroid to PSTN no (i.e mobile number) call is not working. Gateway timeout error is shown in imsdroid. INVITE is sent to FreeSWITCH and freeswitch server is routing the call to Twilio. But not receiving any response. What could be the issue.

IP is allowed.From SIPML5 client to PSTN call is working. Below is the INVITE that is going from FreeSWITCH

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 220.227.38.107:5080;rport;branch=z9hG4bK5yt865UUKDF4H
Max-Forwards: 69
From: "+919845217138" <sip:[email protected]>;tag=eFNKtBae4HN1r
To: <sip:[email protected]>
Call-ID: 06f7df5f-1f81-1235-67b0-2e81eca04f81
CSeq: 98923179 INVITE
Contact: <sip:[email protected]:5080;transport=udp;gw=Twilio-outbound>
User-Agent: FreeSWITCH-mod_sofia/1.6.10+git~20160824T215404Z~726448d962~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 248
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
X-FS-Support: update_display,send_info
Remote-Party-ID: "+919845217138" <sip:[email protected]>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1478495490 1478495491 IN IP4 220.227.38.107
s=FreeSWITCH
c=IN IP4 220.227.38.107
t=0 0
m=audio 22996 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

SIPML5 INVITE

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 220.227.38.107:5080;rport;branch=z9hG4bKD5jv3tcpNtFDS
Max-Forwards: 69
From: "+919845217138" <sip:[email protected]>;tag=cvS8XKDp62Z5e
To: <sip:[email protected]>
Call-ID: 69171aa5-1d0d-1235-67b0-2e81eca04f81
CSeq: 98788399 INVITE
Contact: <sip:[email protected]:5080;transport=udp;gw=Twilio-outbound>
User-Agent: FreeSWITCH-mod_sofia/1.6.10+git~20160824T215404Z~726448d962~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 488
X-FS-Support: update_display,send_info
Remote-Party-ID: "+919845217138" <sip:[email protected]>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1478224523 1478224524 IN IP4 220.227.38.107
s=FreeSWITCH
c=IN IP4 220.227.38.107
t=0 0
m=audio 24404 RTP/AVP 102 9 0 8 103 101
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=0; cbr=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40; stereo=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 telephone-event/48000
a=fmtp:103 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
ACK sip:172.18.22.113:5060 SIP/2.0
Via: SIP/2.0/UDP 220.227.38.107:5080;rport;branch=z9hG4bKeecN5NXSj35Zm
Route: <sip:54.172.60.3:5060;lr;ftag=cvS8XKDp62Z5e>
Max-Forwards: 70
From: "+919845217138" <sip:[email protected]>;tag=cvS8XKDp62Z5e
To: <sip:[email protected]>;tag=70832428_6772d868_27e46324-e125-4895-a6ff-98031cdb43fc
Call-ID: 69171aa5-1d0d-1235-67b0-2e81eca04f81
CSeq: 98788399 ACK
Contact: <sip:[email protected]:5080;transport=udp;gw=Twilio-outbound>
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.3:5060;branch=z9hG4bK763a.52f422d4.0
Via: SIP/2.0/UDP 172.18.22.113:5060;rport=5060;received=54.172.61.235;branch=z9hG4bK27e46324-e125-4895-a6ff-98031cdb43fc_6772d868_294770755291215
From: <sip:[email protected]>;tag=70832428_6772d868_27e46324-e125-4895-a6ff-98031cdb43fc
To: "+919845217138" <sip:[email protected]>;tag=cvS8XKDp62Z5e
Call-ID: 69171aa5-1d0d-1235-67b0-2e81eca04f81
CSeq: 1 BYE
User-Agent: FreeSWITCH-mod_sofia/1.6.10+git~20160824T215404Z~726448d962~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Content-Length: 0

please post the siptrace here.you have to add your ip to twilio so it can allow call from it — suren
if you will post the siptrace then easily i can tell you the what the main reason behind it. — suren
IP is allowed.From SIPML5 client to PSTN call is working. Below is the INVITE that is going from FreeSWITCH — Hansika Penugonda
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000.please remove these two header before sending to twilio — suren
for removing header you to do this before bridge call to pstn twilio sip gateway. <action application="unset" data="sip_h_P-Preferred-Service"/> — suren

suren suren · Accepted Answer · 2016-11-08T11:48:11

Please remove these two headers before sending to Twilio:

P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel 

P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000

To do so you have to write these two line before sending call to Twilio sip gateway:

<action application="unset" data="sip_h_P-Preferred-Service"/> 

<action application="unset" data="sip_h_P-Access-Network-Info"/>

Android imsdroid to PSTN call is not working

1 Answers