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bascheew
New Contributor II

Having issues when transferring calls through an Adtran SBC

We have an Adtran 924e configured as an SBC.  The outside provider is AT&T and inside is running 3CX PBX.  All calls are working fine with one exception.  When we have a call from the outside and we attempt to transfer it to another call on the outside we get a "500 Server Internal Error" from the SBC.  Here are the relevant parts from the config.

voice feature-mode network

voice timeouts preconnected 0

voice timeouts preconnecting 0

voice transfer-mode local

voice forward-mode local

!

!

voice trunk T01 type sip

  description "AT&T - Primary"

  match dnis "NXX-XXXX" substitute "828-NXX-XXXX" name "Prepend Area Code"

  sip-server primary 1.1.1.1

  grammar from host local

!

voice trunk T02 type sip

  description "AT&T - Secondary"

  match dnis "NXX-XXXX" substitute "828-NXX-XXXX" name "Prepend Area Code"

  sip-server primary 2.2.2.2

  grammar from host local

!

voice trunk T04 type sip

  description "3CX"

  sip-server primary 10.16.0.6

  grammar from host local

  transfer-mode network

!

!

voice grouped-trunk AT&T

  trunk T01

  trunk T02

  accept NXX-XXXX cost 0

  accept 1-NXX-NXX-XXXX cost 0

  accept NXX-NXX-XXXX cost 0

!

!

voice grouped-trunk 3CX

  trunk T04

  accept ***OUR INTERNAL NUMBERS ARE HERE*** cost 0

!
ip rtp symmetric-filter

ip rtp media-anchoring

And here are the SIP logs from the SBC:

15:42:05.380 SIP.MSG INVITE RSP RX 8281111111 8283333333
SIP/2.0 180 Ringing
From: "Test User"<sip:8281111111@3.3.3.3:5060;transport=UDP>;tag=5355bf0-20fd1daa-13c4-2bf59-9b6bffaa-2bf59
To: <sip:8283333333@1.1.1.1:5060>;tag=192453536-1492112525261
Call-ID: 53b2720-20fd1daa-13c4-2bf59-c555a872-2bf59@3.3.3.3
CSeq: 1 INVITE
Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK-2bf59-abb76ec-335c53f7
Supported: timer
Call-Info: <sip:2.2.2.3>;appearance-index=1
Contact: <sip:1.1.1.1:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,NOTIFY,PRACK,OPTIONS
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 217

v=0
o=BroadWorks 36898327 1 IN IP4 2.2.2.4
s=-
c=IN IP4 2.2.2.4
t=0 0
m=audio 31790 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20

15:42:05.384 SIP.MSG INVITE RSP TX 8281111111 8283333333
SIP/2.0 183 Session Progress
From: "Test User"<sip:8281111111@172.16.0.7:5060>;tag=ee4ac675
To: <sip:8283333333@172.16.0.7:5060>;tag=5328ad8-20fd1daa-13c4-2bf5b-b22b0b07-2bf5b
Call-ID: YzJhMWMxOTc4NGE0YjdkZmQ0MzI0MTQwOGI1YTE4MDY.
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.16.0.6:5060;rport=5060;branch=z9hG4bK-d8754z-ca08cc64a178df5f-1---d8754z-
Contact: <sip:8283333333@172.16.0.7:5060;transport=UDP>
Supported: 100rel,replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_Total_Access_924e_2nd_Gen/R12.3.1.E
Content-Type: application/sdp
Content-Length: 211

v=0
o=BroadWorks 36898327 1 IN IP4 172.16.0.7
s=-
c=IN IP4 172.16.0.7
t=0 0
m=audio 12444 RTP/AVP 0 101
a=sendrecv
a=maxptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

15:42:06.771 SIP.MSG INVITE REQ RX 8281111111 8283333333
INVITE sip:8283333333@172.16.0.7:5060 SIP/2.0
From: "9001"<sip:8281111111@172.16.0.7:5060>;tag=73205c30
To: <sip:8283333333@172.16.0.7:5060>
Call-ID: OGNhM2Q1ZTVkNTA5NGVlYzc0ZWEzNWQ3ZGY0NzRiM2Q.
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.16.0.6:5060;rport;branch=z9hG4bK-d8754z-a106c01783148111-1---d8754z-
Max-Forwards: 70
Supported: replaces
User-Agent: 3CXPhoneSystem 14.0.49169.513 (48654)
Remote-Party-ID: "8281111111"<sip:8281111111@172.16.0.7:5060>;party=calling
Contact: <sip:8281111111@10.16.0.6:5060>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,INFO,MESSAGE
Replaces: YzJhMWMxOTc4NGE0YjdkZmQ0MzI0MTQwOGI1YTE4MDY.;to-tag=5328ad8-20fd1daa-13c4-2bf5b-b22b0b07-2bf5b;from-tag=ee4ac675
Content-Length: 0


15:42:06.775 SIP.MSG INVITE RSP TX 8281111111 8283333333
SIP/2.0 100 Trying
From: "9001"<sip:8281111111@172.16.0.7:5060>;tag=73205c30
To: <sip:8283333333@172.16.0.7:5060>
Call-ID: OGNhM2Q1ZTVkNTA5NGVlYzc0ZWEzNWQ3ZGY0NzRiM2Q.
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.16.0.6:5060;rport=5060;branch=z9hG4bK-d8754z-a106c01783148111-1---d8754z-
Contact: <sip:8283333333@172.16.0.7:5060;transport=UDP>
Supported: 100rel,replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_Total_Access_924e_2nd_Gen/R12.3.1.E
Content-Length: 0


15:42:09.881 SIP.MSG INVITE RSP TX 8281111111 8283333333
SIP/2.0 500 Server Internal Error
From: "9001"<sip:8281111111@172.16.0.7:5060>;tag=73205c30
To: <sip:8283333333@172.16.0.7:5060>;tag=535f448-20fd1daa-13c4-2bf60-9f6daffd-2bf60
Call-ID: OGNhM2Q1ZTVkNTA5NGVlYzc0ZWEzNWQ3ZGY0NzRiM2Q.
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.16.0.6:5060;rport=5060;branch=z9hG4bK-d8754z-a106c01783148111-1---d8754z-
Supported: 100rel,replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_Total_Access_924e_2nd_Gen/R12.3.1.E
Content-Length: 0

Labels (1)
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2 Replies
jay
Valued Contributor
Valued Contributor

Re: Having issues when transferring calls through an Adtran SBC

Brian, can you run that same debug with "debug voice verbose" running at the same time? Thanks

bascheew
New Contributor II

Re: Having issues when transferring calls through an Adtran SBC

We have found that Yealink phones are working fine, but that Cisco 7900 phones with SIP firmware are having the problem.  We will continue investigating this on our end to see what is different about them.