Asterisk 11.3 Avaya NRS

Asterisk Asterisk Support
1

Hi….
We are trying to get Asterisk 11.3 to work against Avaya NRS.
The problem is incoming calls against the Asterisk, outgoing calls from the Asterisk works.
When incoming calls we see the Invite  trying  200 OK from Asterisk but we don’t get an ack from the Avaya NRS…Asterisk retransmits and gives up after a while.
Here is the logoutput…
From: sip:3023@test.local;user=phone;tag=1eff750-6f280a0a-13c4-55013-540-1a69204a-540
To: sip:3131@test.local;user=phone
Call-ID: 29ff030-6f280a0a-13c4-55013-540-334868e4-540
CSeq: 1 INVITE
Contact: sip:3023@test.local:5060;maddr=10.10.40.111;transport=udp;user=phone
Max-forwards: 68
Supported: x-nortel-sipvc,replaces
User-agent: Nortel CS1000 SIP GW release_7.0 version_ssLinux-7.65.16.21
P-asserted-identity: sip:3023@test.local;user=phone
Privacy: none
History-info: sip:3131@test.local;user=phone;index=1
X-nt-corr-id: 000000520f18020309@0019e1e81798-c0a82865
Allow: INVITE, ACK, BYE, REGISTER, REFER, NOTIFY, CANCEL, OPTIONS, INFO, SUBSCRIBE, UPDATE
Content-Type: multipart/mixed ;boundary=unique-boundary-1
Alert-Info: cid:internal@test.local
Content-Length: 922

–unique-boundary-1
Content-Type: application/sdp

v=0
o=- 24 1 IN IP4 10.10.40.111
s=-
c=IN IP4 192.168.50.192
t=0 0
m=audio 50004 RTP/AVP 0 18 8 101 111
c=IN IP4 192.168.50.192
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:111 X-nt-inforeq/8000
a=fmtp:18 annexb=no

–unique-boundary-1
Content-Type: application/x-nt-mcdn-frag-hex;version=ssLinux-7.65.16.21;base=x2611
Content-Disposition: signal;handling=optional

0500a801
0107130081900000a200
09090f00e9a0830001002800
1315070011fa0f00a10d02010102020100cc049201538800
1e0403008183
460e01000a00010005000a0000000000
4a1c010018000100000000000000000000000000050000000000a3320000
–unique-boundary-1
Content-Type: application/x-nt-epid-frag-hex;version=ssLinux-7.65.16.21;base=x2611
Content-Disposition: signal;handling=optional

011201
00:19:e1:e8:17:98
–unique-boundary-1–
<------------->
[Sep 3 18:20:28] VERBOSE[6368] chan_sip.c: — (20 headers 35 lines) —
[Sep 3 18:20:28] VERBOSE[6368][C-00000014] chan_sip.c: Ignoring this INVITE request
[Sep 3 18:20:28] VERBOSE[6368][C-00000014] chan_sip.c:
<— Transmitting (NAT) to 10.10.40.101:40718 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.40.101:5060;branch=z9hG4bKccefb54e35198dd32d1b8dc0-7acec0eb.1;received=10.10.40.101;rport=40718
Via: SIP/2.0/UDP 10.10.40.111:5060;branch=z9hG4bK-540-1482ed-4b9bb20b;received=10.10.40.111
Record-Route: sip:10.10.40.101@10.10.40.101:5060;transport=udp;lr
From: sip:3023@test.local;user=phone;tag=1eff750-6f280a0a-13c4-55013-540-1a69204a-540
To: sip:3131@test.local;user=phone
Call-ID: 29ff030-6f280a0a-13c4-55013-540-334868e4-540
CSeq: 1 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3131@10.10.60.214:5060
Content-Length: 0

<------------>
[Sep 3 18:20:28] VERBOSE[6368][C-00000014] chan_sip.c: Audio is at 14400
[Sep 3 18:20:28] VERBOSE[6368][C-00000014] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Sep 3 18:20:28] VERBOSE[6368][C-00000014] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Sep 3 18:20:28] VERBOSE[6368][C-00000014] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 3 18:20:28] VERBOSE[6368][C-00000014] chan_sip.c:
<— Transmitting (NAT) to 10.10.40.101:40718 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.40.101:5060;branch=z9hG4bKccefb54e35198dd32d1b8dc0-7acec0eb.1;received=10.10.40.101;rport=40718
Via: SIP/2.0/UDP 10.10.40.111:5060;branch=z9hG4bK-540-1482ed-4b9bb20b;received=10.10.40.111
Record-Route: sip:10.10.40.101@10.10.40.101:5060;transport=udp;lr
From: sip:3023@test.local;user=phone;tag=1eff750-6f280a0a-13c4-55013-540-1a69204a-540
To: sip:3131@test.local;user=phone;tag=as1ba52239
Call-ID: 29ff030-6f280a0a-13c4-55013-540-334868e4-540
CSeq: 1 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3131@10.10.60.214:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 831067332 831067335 IN IP4 10.10.60.214
s=Asterisk PBX 11.3.0
c=IN IP4 10.10.60.214
t=0 0
m=audio 14400 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Sep 3 18:20:28] VERBOSE[6368] chan_sip.c: Retransmitting #3 (NAT) to 10.10.40.101:40718:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.40.101:5060;branch=z9hG4bKccefb54e35198dd32d1b8dc0-7acec0eb.1;received=10.10.40.101;rport=40718
Via: SIP/2.0/UDP 10.10.40.111:5060;branch=z9hG4bK-540-1482ed-4b9bb20b;received=10.10.40.111
Record-Route: sip:10.10.40.101@10.10.40.101:5060;transport=udp;lr
From: sip:3023@test.local;user=phone;tag=1eff750-6f280a0a-13c4-55013-540-1a69204a-540
To: sip:3131@test.local;user=phone;tag=as1ba52239
Call-ID: 29ff030-6f280a0a-13c4-55013-540-334868e4-540
CSeq: 1 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3131@10.10.60.214:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 831067332 831067332 IN IP4 10.10.60.214
s=Asterisk PBX 11.3.0
c=IN IP4 10.10.60.214
t=0 0
m=audio 14400 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[Sep 3 18:20:32] VERBOSE[6368] chan_sip.c: Retransmitting #4 (NAT) to 10.10.40.101:40718:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.40.101:5060;branch=z9hG4bKccefb54e35198dd32d1b8dc0-7acec0eb.1;received=10.10.40.101;rport=40718
Via: SIP/2.0/UDP 10.10.40.111:5060;branch=z9hG4bK-540-1482ed-4b9bb20b;received=10.10.40.111
Record-Route: sip:10.10.40.101@10.10.40.101:5060;transport=udp;lr
From: sip:3023@test.local;user=phone;tag=1eff750-6f280a0a-13c4-55013-540-1a69204a-540
To: sip:3131@test.local;user=phone;tag=as1ba52239
Call-ID: 29ff030-6f280a0a-13c4-55013-540-334868e4-540
CSeq: 1 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3131@10.10.60.214:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 831067332 831067332 IN IP4 10.10.60.214
s=Asterisk PBX 11.3.0
c=IN IP4 10.10.60.214
t=0 0
m=audio 14400 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[Sep 3 18:20:34] VERBOSE[6629][C-00000014] res_agi.c: – Playing ‘/var/lib/asterisk/sounds/vb/cache/_play_teleq_tqrec/65f48b4829039154b2a922e69a2b04ce’ (escape_digits=1234567890*#) (sample_offset 0)
[Sep 3 18:20:36] VERBOSE[6368] chan_sip.c: Retransmitting #5 (NAT) to 10.10.40.101:40718:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.40.101:5060;branch=z9hG4bKccefb54e35198dd32d1b8dc0-7acec0eb.1;received=10.10.40.101;rport=40718
Via: SIP/2.0/UDP 10.10.40.111:5060;branch=z9hG4bK-540-1482ed-4b9bb20b;received=10.10.40.111
Record-Route: sip:10.10.40.101@10.10.40.101:5060;transport=udp;lr
From: sip:3023@test.local;user=phone;tag=1eff750-6f280a0a-13c4-55013-540-1a69204a-540
To: sip:3131@test.local;user=phone;tag=as1ba52239
Call-ID: 29ff030-6f280a0a-13c4-55013-540-334868e4-540
CSeq: 1 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3131@10.10.60.214:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 831067332 831067332 IN IP4 10.10.60.214
s=Asterisk PBX 11.3.0
c=IN IP4 10.10.60.214
t=0 0
m=audio 14400 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[Sep 3 18:20:40] VERBOSE[6368] chan_sip.c: Retransmitting #6 (NAT) to 10.10.40.101:40718:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.40.101:5060;branch=z9hG4bKccefb54e35198dd32d1b8dc0-7acec0eb.1;received=10.10.40.101;rport=40718
Via: SIP/2.0/UDP 10.10.40.111:5060;branch=z9hG4bK-540-1482ed-4b9bb20b;received=10.10.40.111
Record-Route: sip:10.10.40.101@10.10.40.101:5060;transport=udp;lr
From: sip:3023@test.local;user=phone;tag=1eff750-6f280a0a-13c4-55013-540-1a69204a-540
To: sip:3131@test.local;user=phone;tag=as1ba52239
Call-ID: 29ff030-6f280a0a-13c4-55013-540-334868e4-540
CSeq: 1 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3131@10.10.60.214:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 831067332 831067332 IN IP4 10.10.60.214
s=Asterisk PBX 11.3.0
c=IN IP4 10.10.60.214
t=0 0
m=audio 14400 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[Sep 3 18:20:44] VERBOSE[6368] chan_sip.c: Retransmitting #7 (NAT) to 10.10.40.101:40718:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.40.101:5060;branch=z9hG4bKccefb54e35198dd32d1b8dc0-7acec0eb.1;received=10.10.40.101;rport=40718
Via: SIP/2.0/UDP 10.10.40.111:5060;branch=z9hG4bK-540-1482ed-4b9bb20b;received=10.10.40.111
Record-Route: sip:10.10.40.101@10.10.40.101:5060;transport=udp;lr
From: sip:3023@test.local;user=phone;tag=1eff750-6f280a0a-13c4-55013-540-1a69204a-540
To: sip:3131@test.local;user=phone;tag=as1ba52239
Call-ID: 29ff030-6f280a0a-13c4-55013-540-334868e4-540
CSeq: 1 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3131@10.10.60.214:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 831067332 831067332 IN IP4 10.10.60.214
s=Asterisk PBX 11.3.0
c=IN IP4 10.10.60.214
t=0 0
m=audio 14400 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[Sep 3 18:20:47] VERBOSE[6629][C-00000014] res_agi.c: – Playing ‘/var/lib/asterisk/sounds/vb/cache/_play_teleq_tqrec/9cc20eaa4c1166d3d1a03ea98f348a58’ (escape_digits=0123456789*#) (sample_offset 0)
[Sep 3 18:20:48] VERBOSE[6368] chan_sip.c: Retransmitting #8 (NAT) to 10.10.40.101:40718:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.40.101:5060;branch=z9hG4bKccefb54e35198dd32d1b8dc0-7acec0eb.1;received=10.10.40.101;rport=40718
Via: SIP/2.0/UDP 10.10.40.111:5060;branch=z9hG4bK-540-1482ed-4b9bb20b;received=10.10.40.111
Record-Route: sip:10.10.40.101@10.10.40.101:5060;transport=udp;lr
From: sip:3023@test.local;user=phone;tag=1eff750-6f280a0a-13c4-55013-540-1a69204a-540
To: sip:3131@test.local;user=phone;tag=as1ba52239
Call-ID: 29ff030-6f280a0a-13c4-55013-540-334868e4-540
CSeq: 1 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3131@10.10.60.214:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 831067332 831067332 IN IP4 10.10.60.214
s=Asterisk PBX 11.3.0
c=IN IP4 10.10.60.214
t=0 0
m=audio 14400 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[Sep 3 18:20:50] VERBOSE[6629][C-00000014] pbx.c: == Spawn extension (default, 3131, 1) exited non-zero on ‘SIP/test.local-0000000e’
[Sep 3 18:20:50] VERBOSE[6629][C-00000014] pbx.c: – Executing [h@default:1] NoOp(“SIP/test.local-0000000e”, ““Hangup””) in new stack
[Sep 3 18:20:50] VERBOSE[6629][C-00000014] chan_sip.c: Scheduling destruction of SIP dialog ‘29ff030-6f280a0a-13c4-55013-540-334868e4-540’ in 32000 ms (Method: INVITE)
[Sep 3 18:20:52] VERBOSE[6368] chan_sip.c: Retransmitting #9 (NAT) to 10.10.40.101:40718:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.40.101:5060;branch=z9hG4bKccefb54e35198dd32d1b8dc0-7acec0eb.1;received=10.10.40.101;rport=40718
Via: SIP/2.0/UDP 10.10.40.111:5060;branch=z9hG4bK-540-1482ed-4b9bb20b;received=10.10.40.111
Record-Route: sip:10.10.40.101@10.10.40.101:5060;transport=udp;lr
From: sip:3023@test.local;user=phone;tag=1eff750-6f280a0a-13c4-55013-540-1a69204a-540
To: sip:3131@test.local;user=phone;tag=as1ba52239
Call-ID: 29ff030-6f280a0a-13c4-55013-540-334868e4-540
CSeq: 1 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3131@10.10.60.214:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 831067332 831067332 IN IP4 10.10.60.214
s=Asterisk PBX 11.3.0
c=IN IP4 10.10.60.214
t=0 0
m=audio 14400 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[Sep 3 18:20:54] VERBOSE[6368] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.40.101:5060:
OPTIONS sip:test.local SIP/2.0
Via: SIP/2.0/UDP 10.10.60.214:5060;branch=z9hG4bK3d03b108
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.10.60.214;tag=as79b003f8
To: sip:test.local
Contact: sip:asterisk@10.10.60.214:5060
Call-ID: 5d25ccd35f6128633cdc2acd65467893@10.10.60.214:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.3.0
Date: Tue, 03 Sep 2013 16:20:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[Sep 3 18:20:54] VERBOSE[6368] chan_sip.c:
<— SIP read from UDP:10.10.40.101:40718 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.60.214:5060;branch=z9hG4bK3d03b108;received=10.10.60.214
From: “asterisk” sip:asterisk@10.10.60.214;tag=as79b003f8
To: sip:test.local;tag=57
Call-ID: 5d25ccd35f6128633cdc2acd65467893@10.10.60.214:5060
CSeq: 102 OPTIONS
Server: Nortel Networks SIP Proxy Server linux-7.65.16 (Primary)
Content-Length: 0

<------------->
[Sep 3 18:20:54] VERBOSE[6368] chan_sip.c: — (8 headers 0 lines) —
[Sep 3 18:20:54] VERBOSE[6368] chan_sip.c: Really destroying SIP dialog ‘5d25ccd35f6128633cdc2acd65467893@10.10.60.214:5060’ Method: OPTIONS
[Sep 3 18:20:56] VERBOSE[6368] chan_sip.c: Retransmitting #10 (NAT) to 10.10.40.101:40718:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.40.101:5060;branch=z9hG4bKccefb54e35198dd32d1b8dc0-7acec0eb.1;received=10.10.40.101;rport=40718
Via: SIP/2.0/UDP 10.10.40.111:5060;branch=z9hG4bK-540-1482ed-4b9bb20b;received=10.10.40.111
Record-Route: sip:10.10.40.101@10.10.40.101:5060;transport=udp;lr
From: sip:3023@test.local;user=phone;tag=1eff750-6f280a0a-13c4-55013-540-1a69204a-540
To: sip:3131@test.local;user=phone;tag=as1ba52239
Call-ID: 29ff030-6f280a0a-13c4-55013-540-334868e4-540
CSeq: 1 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3131@10.10.60.214:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 831067332 831067332 IN IP4 10.10.60.214
s=Asterisk PBX 11.3.0
c=IN IP4 10.10.60.214
t=0 0
m=audio 14400 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[Sep 3 18:20:56] WARNING[6368] chan_sip.c: Retransmission timeout reached on transmission 29ff030-6f280a0a-13c4-55013-540-334868e4-540 for seqno 1 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 31999ms with no response.
The thing is that we don’t see that the NRS sends the “ACK” when we dump the traffic on the NRS side. My question is can it be something with the configuration on the Asterisk side that is causing the NRS not to send the ACK?

2

You are not receiving any ACKs to your 200 OK on the INVITE.

3

Thanks for the answer…the question is why the NRS dont send the ACK?
The People managing the NRS had some idea that we send the 200 OK to fast…
They also wanted us to add session in progress during the setup of the call, did that,no difference.
We will continue troublshooting this issue, anyone got NRS and Asterisk (SIP) to Work?

4

Either it is broken, or you have a NAT or firwall problem.