Asterisk 1.8.32.1 -> Cisco AS5400 ACK/OK timeout

Asterisk Asterisk Support
1

Hi all, this is my first post and I apologize if it’s not appropriate for this forum. I’m looking for a workaround to this, or any advice that anyone can give.

In a nutshell, topography for outbound calls is: Asterisk 1.8.32.1 -SIP> Cisco AS5400 -SIP> Carrier (Broadsoft)

For some phone numbers, my AS5400 throws a 488 and then it and the Asterisk go into an ACK/OK battle until the call simply disconnects.

I really appreciate any insight that the forum may have for my problem. Below is the SIP trace for this call for the call leg between Asterisk and the AS5400. I can provide any other configuration details that are necessary.

Thanks everyone, and take care!

-Douglas

    10.1.1.1                    10.1.1.2
    |                               |
1 : |U------------INVITE----------->|
2 : |<------100 Trying/INVITE------U|
3 : |<------180 Ringing/INVITE-----U|
4 : |<--------200 OK/INVITE--------U|
5 : |U-------------ACK------------->|
6 : |<-488 Not Acceptable Media/IN-U|
7 : |U-------------ACK------------->|
8 : |<--------200 OK/INVITE--------U|
9 : |U-------------ACK------------->|
10: |<--------200 OK/INVITE--------U|
11: |U-------------ACK------------->|
12: |<--------200 OK/INVITE--------U|
13: |U-------------ACK------------->|
14: |<--------200 OK/INVITE--------U|
15: |U-------------ACK------------->|
16: |<--------200 OK/INVITE--------U|
17: |U-------------ACK------------->|

Generated SIP Workbench by BreakPoint Software, Inc. (http://www.sipworkbench.com)

<<<<<<<< Msg #1 / Packet #21: 10.1.1.1:5060 --> 10.1.1.2:5060 >>>>>>>>>
INVITE sip:913193788760@10.1.1.2 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK5f82229c
Max-Forwards: 70
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>
Contact: <sip:5154229885@10.1.1.1:5060>
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.32.1
Date: Tue, 29 Sep 2015 18:40:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
X-comms-PILOT: 15152767447
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 672330474 672330474 IN IP4 10.1.1.1
s=Asterisk PBX 1.8.32.1
c=IN IP4 10.1.1.1
t=0 0
m=audio 13684 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<<<<<<<< Msg #2 / Packet #22: 10.1.1.2:5060 --> 10.1.1.1:5060 >>>>>>>>>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK5f82229c
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Date: Tue, 29 Sep 2015 18:40:56 GMT
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


<<<<<<<< Msg #3 / Packet #27: 10.1.1.2:5060 --> 10.1.1.1:5060 >>>>>>>>>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK5f82229c
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Date: Tue, 29 Sep 2015 18:40:56 GMT
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:913193788760@10.1.1.2:5060>
Content-Length: 0


<<<<<<<< Msg #4 / Packet #53: 10.1.1.2:5060 --> 10.1.1.1:5060 >>>>>>>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK5f82229c
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Date: Tue, 29 Sep 2015 18:40:56 GMT
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: <sip:913193788760@10.1.1.2:5060>
Content-Type: application/sdp
Content-Length: 240

v=0
o=CiscoSystemsSIP-GW-UserAgent 1965 2222 IN IP4 10.1.1.2
s=SIP Call
c=IN IP4 10.1.1.2
t=0 0
m=audio 19118 RTP/AVP 0 101
c=IN IP4 10.1.1.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 
a=ptime:20

<<<<<<<< Msg #5 / Packet #54: 10.1.1.1:5060 --> 10.1.1.2:5060 >>>>>>>>>
ACK sip:913193788760@10.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK7dfa98d1
Max-Forwards: 70
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Contact: <sip:5154229885@10.1.1.1:5060>
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.32.1
Content-Length: 0


<<<<<<<< Msg #6 / Packet #57: 10.1.1.2:5060 --> 10.1.1.1:5060 >>>>>>>>>
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK5f82229c
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
Warning: 304 10.1.1.2 "Media Type(s) Unavailable"
CSeq: 102 INVITE
Content-Length: 0


<<<<<<<< Msg #7 / Packet #58: 10.1.1.1:5060 --> 10.1.1.2:5060 >>>>>>>>>
ACK sip:913193788760@10.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK7dfa98d1
Max-Forwards: 70
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Contact: <sip:5154229885@10.1.1.1:5060>
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.32.1
Content-Length: 0


<<<<<<<< Msg #8 / Packet #61: 10.1.1.2:5060 --> 10.1.1.1:5060 >>>>>>>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK5f82229c
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Date: Tue, 29 Sep 2015 18:41:03 GMT
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: <sip:913193788760@10.1.1.2:5060>
Content-Type: application/sdp
Content-Length: 240

v=0
o=CiscoSystemsSIP-GW-UserAgent 1965 2222 IN IP4 10.1.1.2
s=SIP Call
c=IN IP4 10.1.1.2
t=0 0
m=audio 19118 RTP/AVP 0 101
c=IN IP4 10.1.1.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 
a=ptime:20

<<<<<<<< Msg #9 / Packet #62: 10.1.1.1:5060 --> 10.1.1.2:5060 >>>>>>>>>
ACK sip:913193788760@10.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK06710643
Max-Forwards: 70
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Contact: <sip:5154229885@10.1.1.1:5060>
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.32.1
Content-Length: 0


<<<<<<<< Msg #10 / Packet #63: 10.1.1.2:5060 --> 10.1.1.1:5060 >>>>>>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK5f82229c
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Date: Tue, 29 Sep 2015 18:41:04 GMT
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: <sip:913193788760@10.1.1.2:5060>
Content-Type: application/sdp
Content-Length: 240

v=0
o=CiscoSystemsSIP-GW-UserAgent 1965 2222 IN IP4 10.1.1.2
s=SIP Call
c=IN IP4 10.1.1.2
t=0 0
m=audio 19118 RTP/AVP 0 101
c=IN IP4 10.1.1.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 
a=ptime:20

<<<<<<<< Msg #11 / Packet #64: 10.1.1.1:5060 --> 10.1.1.2:5060 >>>>>>>>
ACK sip:913193788760@10.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK1e13960a
Max-Forwards: 70
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Contact: <sip:5154229885@10.1.1.1:5060>
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.32.1
Content-Length: 0


<<<<<<<< Msg #12 / Packet #77: 10.1.1.2:5060 --> 10.1.1.1:5060 >>>>>>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK5f82229c
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Date: Tue, 29 Sep 2015 18:41:05 GMT
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: <sip:913193788760@10.1.1.2:5060>
Content-Type: application/sdp
Content-Length: 240

v=0
o=CiscoSystemsSIP-GW-UserAgent 1965 2222 IN IP4 10.1.1.2
s=SIP Call
c=IN IP4 10.1.1.2
t=0 0
m=audio 19118 RTP/AVP 0 101
c=IN IP4 10.1.1.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 
a=ptime:20

<<<<<<<< Msg #13 / Packet #78: 10.1.1.1:5060 --> 10.1.1.2:5060 >>>>>>>>
ACK sip:913193788760@10.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK771a82af
Max-Forwards: 70
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Contact: <sip:5154229885@10.1.1.1:5060>
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.32.1
Content-Length: 0


<<<<<<<< Msg #14 / Packet #91: 10.1.1.2:5060 --> 10.1.1.1:5060 >>>>>>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK5f82229c
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Date: Tue, 29 Sep 2015 18:41:07 GMT
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: <sip:913193788760@10.1.1.2:5060>
Content-Type: application/sdp
Content-Length: 240

v=0
o=CiscoSystemsSIP-GW-UserAgent 1965 2222 IN IP4 10.1.1.2
s=SIP Call
c=IN IP4 10.1.1.2
t=0 0
m=audio 19118 RTP/AVP 0 101
c=IN IP4 10.1.1.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 
a=ptime:20

<<<<<<<< Msg #15 / Packet #92: 10.1.1.1:5060 --> 10.1.1.2:5060 >>>>>>>>
ACK sip:913193788760@10.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK0b73be5a
Max-Forwards: 70
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Contact: <sip:5154229885@10.1.1.1:5060>
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.32.1
Content-Length: 0


<<<<<<< Msg #16 / Packet #119: 10.1.1.2:5060 --> 10.1.1.1:5060 >>>>>>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK5f82229c
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Date: Tue, 29 Sep 2015 18:41:11 GMT
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: <sip:913193788760@10.1.1.2:5060>
Content-Type: application/sdp
Content-Length: 240

v=0
o=CiscoSystemsSIP-GW-UserAgent 1965 2222 IN IP4 10.1.1.2
s=SIP Call
c=IN IP4 10.1.1.2
t=0 0
m=audio 19118 RTP/AVP 0 101
c=IN IP4 10.1.1.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 
a=ptime:20

<<<<<<< Msg #17 / Packet #120: 10.1.1.1:5060 --> 10.1.1.2:5060 >>>>>>>>
ACK sip:913193788760@10.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK4f50accc
Max-Forwards: 70
From: "Dymin Systems" <sip:5154229885@10.1.1.1>;tag=as6622dea1
To: <sip:913193788760@10.1.1.2>;tag=F9934C0-75B
Contact: <sip:5154229885@10.1.1.1:5060>
Call-ID: 68b4fa09575f09ef4660d0ae6348748b@10.1.1.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.32.1
Content-Length: 0
2

The peer is broken. You can only have one response with a number greater than or equal to 200 for a single transaction. Once it has sent 200, it is too late for it to complain about media compatibility.

Could you please wrap the trace in a code block, to make scrolling through this easier.

(As Asterisk uses early offer SDP, there is no reason why 488 could not be sent instead of 200, if the media type really is incompatible, but, for late offer SDP, the only option that the server has if the SDP on the ACK is unacceptable, is to immediately close the call with a BYE.)

3

OK, edited for readability, and thanks very much for the reply.

And yes I found it very strange that the 5400 complains about media compatibility after already negotiating sdp. I’ve spent so much time tweaking and editing things that I’m starting to think that it’s a bug with the AS5400. I have tried multiple firmware versions to resolve, to no avail.