HI Guys any help on this please.
i can receive calls from Genesys but with no voice.
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To: "90552215570" <sip:90552215570@192.168.171.219>
From: "poc" <sip:065982528@DCHQZCCV01.edi.ae>;tag=sqftsry
Via: SIP/2.0/TCP 172.17.2.101:5065;branch=z9hG4bKpU542hqpvkO3vgfv4zm6
Call-ID: 7fb036493c381689842f5d9af7801e3c@172.17.2.101
CSeq: 1 INVITE
Contact: <sip:065982528@172.17.2.101:5065;transport=tcp>
Max-Forwards: 70
x-inin-crn: 3001929130;loc=%3cRegionDefaultLocation%3e
Supported: join, replaces
User-Agent: ININ-TsServer/18.1.1.12
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, SUBSCRIBE, UPDATE
Accept: application/sdp
Accept-Encoding: identity
Content-Type: application/sdp
Content-Length: 201
v=0
o=ININ 2776777345 2776777346 IN IP4 172.17.2.101
s=Interaction
c=IN IP4 172.17.3.9
t=0 0
m=audio 30142 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
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To: "90552215570" <sip:90552215570@192.168.171.219>
From: "poc" <sip:065982528@DCHQZCCV01.edi.ae>;tag=sqftsry
Call-ID: 7fb036493c381689842f5d9af7801e3c@172.17.2.101
CSeq: 1 INVITE
Via: SIP/2.0/TCP 172.17.2.101:5065;branch=z9hG4bKpU542hqpvkO3vgfv4zm6
Content-Length: 0
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-¦»‹—SIP/2.0 180 Ringing
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, PRACK, UPDATE
Contact: <sip:0552215570@192.168.171.219:5065;transport=TCP;user=phone>
Supported: from-change
User-Agent: OXO103/029.001 GW_103/069.001
P-Asserted-Identity: <sip:0552215570@192.168.171.219;user=phone>
Content-Type: application/sdp
To: "90552215570" <sip:90552215570@192.168.171.219>;tag=8c7964f0a5a7ffdfd1171af4f167193f
From: "poc" <sip:065982528@DCHQZCCV01.edi.ae>;tag=sqftsry
Call-ID: 7fb036493c381689842f5d9af7801e3c@172.17.2.101
CSeq: 1 INVITE
Via: SIP/2.0/TCP 172.17.2.101:5065;branch=z9hG4bKpU542hqpvkO3vgfv4zm6
Content-Length: 279
v=0
o=OxO 1544707276 1544707276 IN IP4 192.168.171.219
s=Alcatel-Lucent OXO103/069.001
c=IN IP4 192.168.171.219
t=0 0
m=audio 32000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtcp:32001
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=maxptime:90
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:-»‹SIP/2.0 200 OK
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, PRACK, UPDATE
Contact: <sip:0552215570@192.168.171.219:5065;transport=TCP;user=phone>
Supported: 100rel,timer,from-change
User-Agent: OXO103/029.001 GW_103/069.001
Session-Expires: 43200;refresher=uas
P-Asserted-Identity: <sip:0552215570@192.168.171.219;user=phone>
To: "90552215570" <sip:90552215570@192.168.171.219>;tag=8c7964f0a5a7ffdfd1171af4f167193f
From: "poc" <sip:065982528@DCHQZCCV01.edi.ae>;tag=sqftsry
Call-ID: 7fb036493c381689842f5d9af7801e3c@172.17.2.101
CSeq: 1 INVITE
Via: SIP/2.0/TCP 172.17.2.101:5065;branch=z9hG4bKpU542hqpvkO3vgfv4zm6
Content-Length: 279
v=0
o=OxO 1544707276 1544707276 IN IP4 192.168.171.219
s=Alcatel-Lucent OXO103/069.001
c=IN IP4 192.168.171.219
t=0 0
m=audio 32000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtcp:32001
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=maxptime:90
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¿% @ @ €Ÿ¾aÀêä÷F E 2jr@ tY¬eÀ¨«ÛÄtÉÔ™li:Â@P¯Œ ACK sip:0552215570@192.168.171.219:5065;user=phone;transport=tcp SIP/2.0
To: "90552215570" <sip:90552215570@192.168.171.219>;tag=8c7964f0a5a7ffdfd1171af4f167193f
From: "poc" <sip:065982528@DCHQZCCV01.edi.ae>;tag=sqftsry
Call-ID: 7fb036493c381689842f5d9af7801e3c@172.17.2.101
CSeq: 1 ACK
Via: SIP/2.0/TCP 172.17.2.101:5065;branch=z9hG4bKjl7hUrvfdpEkEs3zOEx3
Max-Forwards: 70
x-inin-crn: 3001929130;loc=%3cRegionDefaultLocation%3e
Supported: join, replaces
User-Agent: ININ-TsServer/18.1.1.12
Content-Length: 0
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K ® ® €Ÿ¾aÀêä÷F E js@ rê¬eÀ¨«ÛÄtÉÔ™vi:Â@P}S INVITE sip:0552215570@192.168.171.219:5065;user=phone;transport=tcp SIP/2.0
To: "90552215570" <sip:90552215570@192.168.171.219>;tag=8c7964f0a5a7ffdfd1171af4f167193f
From: "poc" <sip:065982528@DCHQZCCV01.edi.ae>;tag=sqftsry
Call-ID: 7fb036493c381689842f5d9af7801e3c@172.17.2.101
CSeq: 2 INVITE
Via: SIP/2.0/TCP 172.17.2.101:5065;branch=z9hG4bKn5sUjhbkU7wvmv3cfmIf
Contact: <sip:065982528@172.17.2.101:5065;transport=tcp>
Max-Forwards: 70
x-inin-crn: 3001929130;loc=%3cRegionDefaultLocation%3e;ms=DCHQZCCM01
Supported: join, replaces
User-Agent: ININ-TsServer/18.1.1.12
Accept: application/sdp
Accept-Encoding: identity
Content-Type: application/sdp
Content-Length: 203
v=0
o=ININ 3434825557 3434825560 IN IP4 172.17.2.101
s=Interaction
c=IN IP4 172.17.2.110
t=0 0
m=audio 30676 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
\\Ø
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To: "90552215570" <sip:90552215570@192.168.171.219>;tag=8c7964f0a5a7ffdfd1171af4f167193f
From: "poc" <sip:065982528@DCHQZCCV01.edi.ae>;tag=sqftsry
Call-ID: 7fb036493c381689842f5d9af7801e3c@172.17.2.101
CSeq: 2 INVITE
Via: SIP/2.0/TCP 172.17.2.101:5065;branch=z9hG4bKn5sUjhbkU7wvmv3cfmIf
Content-Length: 0
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:™»ŒàSIP/2.0 200 OK
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, PRACK, UPDATE
Contact: <sip:0552215570@192.168.171.219:5065;transport=TCP;user=phone>
Supported: 100rel,timer,from-change
User-Agent: OXO103/029.001 GW_103/069.001
Session-Expires: 43200;refresher=uas
P-Asserted-Identity: <sip:0552215570@192.168.171.219;user=phone>
To: "90552215570" <sip:90552215570@192.168.171.219>;tag=8c7964f0a5a7ffdfd1171af4f167193f
From: "poc" <sip:065982528@DCHQZCCV01.edi.ae>;tag=sqftsry
Call-ID: 7fb036493c381689842f5d9af7801e3c@172.17.2.101
CSeq: 2 INVITE
Via: SIP/2.0/TCP 172.17.2.101:5065;branch=z9hG4bKn5sUjhbkU7wvmv3cfmIf
Content-Length: 306
v=0
o=OxO 1544707276 1544707277 IN IP4 192.168.171.219
s=Alcatel-Lucent OXO103/069.001
c=IN IP4 192.168.171.219
t=0 0
m=audio 32000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtcp:32001
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=sendrecv
a=ptime:20
a=maxptime:90
\\Ø åÁ B B €Ÿ¾aÀêä÷F E 4ju@ vT¬eÀ¨«ÛÄuÉs$ €Â Ê \\Ø æn B BÀêä÷F €Ÿ¾a E¸ 4 @ @À¨«Û¬eÉÄui\s%€Ð. ´\\Ø ¥ < < €Ÿ¾aÀêä÷F E (jv@ v_¬eÀ¨«ÛÄuÉs%i\PcÍ \\Ø B N N €Ÿ¾aÀêä÷F E @jw@ tF¬eÀ¨«ÛÄuÉs%i\P=Á ACK sip:0552215570@192.168.171.219:5065;user=phone;transport=tcp SIP/2.0
To: "90552215570" <sip:90552215570@192.168.171.219>;tag=8c7964f0a5a7ffdfd1171af4f167193f
From: "poc" <sip:065982528@DCHQZCCV01.edi.ae>;tag=sqftsry
Call-ID: 7fb036493c381689842f5d9af7801e3c@172.17.2.101
CSeq: 2 ACK
Via: SIP/2.0/TCP 172.17.2.101:5065;branch=z9hG4bKcp77nxq4lqwfh4yyIO5m
Max-Forwards: 70
x-inin-crn: 3001929130;loc=%3cRegionDefaultLocation%3e;ms=DCHQZCCM01
Supported: join, replaces
User-Agent: ININ-TsServer/18.1.1.12
Content-Length: 0
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»Œé :™\\á R €Ÿ¾aÀêä÷F E jy@ tr¬eÀ¨«ÛÄuÉs’=i\PÇÒ BYE sip:0552215570@192.168.171.219:5065;user=phone;transport=tcp SIP/2.0
To: "90552215570" <sip:90552215570@192.168.171.219>;tag=8c7964f0a5a7ffdfd1171af4f167193f
From: "poc" <sip:065982528@DCHQZCCV01.edi.ae>;tag=sqftsry
Call-ID: 7fb036493c381689842f5d9af7801e3c@172.17.2.101
CSeq: 3 BYE
Via: SIP/2.0/TCP 172.17.2.101:5065;branch=z9hG4bK6lU3fy2rrydxb5bw6Uxb
Max-Forwards: 70
x-inin-crn: 3001929130
Supported: join, replaces
User-Agent: ININ-TsServer/18.1.1.12
Content-Length: 0
\\á S 6 6Àêä÷F €Ÿ¾a E¸ (f®@ @¸oÀ¨«Û¬eÉÄui\s”'P ú`Ò \\á _ý û ûÀêä÷F €Ÿ¾a E¸íf¯@ @¶©À¨«Û¬eÉÄui\s”'P úÚ SIP/2.0 200 OK
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, PRACK, UPDATE
Supported: 100rel
User-Agent: OXO103/029.001 GW_103/069.001
To: "90552215570" <sip:90552215570@192.168.171.219>;tag=8c7964f0a5a7ffdfd1171af4f167193f
From: "poc" <sip:065982528@DCHQZCCV01.edi.ae>;tag=sqftsry
Call-ID: 7fb036493c381689842f5d9af7801e3c@172.17.2.101
CSeq: 3 BYE
Via: SIP/2.0/TCP 172.17.2.101:5065;branch=z9hG4bK6lU3fy2rrydxb5bw6Uxb
Content-Length: 0
\\á -¥ < < €Ÿ¾aÀêä÷F E (jz@ v[¬eÀ¨«ÛÄuÉs”'i]ãPÿ^ \] ¯£ < < €Ÿ¾aÀêä÷F E (j{@ vZ¬eÀ¨«ÛÄuÉs”'i]ãPÿ^ \] ²J 6 6Àêä÷F €Ÿ¾a E¸ (f°@ @¸mÀ¨«Û¬eÉÄui]ãs”(P ú_ \] Ú“ < < €Ÿ¾aÀêä÷F E (j|@ vY¬eÀ¨«ÛÄuÉs”(i]äPÿ^
NO voice from Genesys to OXO 10.x
Re: NO voice from Genesys to OXO 10.x
Hi,
Not really sure, but here are some tips:
1. Can you change/play with te settings of the P-asserted ID?
the destination and the p-asserted ID differ, in P-asserted a 9 is added:
P-Asserted-Identity: <sip:0552215570@192.168.171.219;user=phone>
To: "90552215570" <sip:90552215570@192.168.171.219>;
2. Can you change/play with the settings of the SIP Public numbering?
Most SIP connections aren't that sensitive about number formats, but the OXO really doesn't like it if it mismatches.
Problems like an open speech pad without speech can occur in those configurations.
3. Can you check network settings?
Since your Genesys (172.17.2.101) and OXO (192.168.171.219) are in different subnets... All routing between those IP addresses is OK?
I also noticed the use of port 5065 and not 5060 (since you are using a public numbering format), is this also set up like this in the OXO ?
Good luck.
Not really sure, but here are some tips:
1. Can you change/play with te settings of the P-asserted ID?
the destination and the p-asserted ID differ, in P-asserted a 9 is added:
P-Asserted-Identity: <sip:0552215570@192.168.171.219;user=phone>
To: "90552215570" <sip:90552215570@192.168.171.219>;
2. Can you change/play with the settings of the SIP Public numbering?
Most SIP connections aren't that sensitive about number formats, but the OXO really doesn't like it if it mismatches.
Problems like an open speech pad without speech can occur in those configurations.
3. Can you check network settings?
Since your Genesys (172.17.2.101) and OXO (192.168.171.219) are in different subnets... All routing between those IP addresses is OK?
I also noticed the use of port 5065 and not 5060 (since you are using a public numbering format), is this also set up like this in the OXO ?
Good luck.
If I'm ever on life support, unplug me. Then plug me back in, see if that works...
Re: NO voice from Genesys to OXO 10.x
HI Man, Thanks for the reply, issue resolved , was a network firewall configuration.
