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21 posts

Geek
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Topic # 53407 10-Dec-2009 10:42
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Hey All,

I have been having an issue for the last couple of months when making outgoing calls from my asterisk box via VFX. Apon making the call, the call rings, other party picks up, voice passes in both directions, then after 5-ish seconds the call drops. This happens on all outgoing calls. Incoming calls work fine.

Asterisk/VFX have been working fine on this box in the past, and I have SIP trunks with other providers which work fine also so I am not sure where the issue lies. I have included a trace below. Any pointers would be appreciated!

I have tried connecting from different networks (Xnet/xtra/colo), different asterisk versions, and different IP handsets.

Box is on a public IP, hence no NAT settings below. Asterisk version is 1.2.37

mybox*CLI> sip show peer 7985xxxx

  * Name       : 7985xxxx
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : xxxxxxx-incoming
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  CallingPres  : Presentation Allowed, Not Screened
  FromUser     : 7985xxxx
  FromDomain   : pan.wxnz.net
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : No
  Callerid     : "" <>
  Expire       : -1
  Insecure     : port,invite
  Nat          : No
  ACL          : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID    : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : pan.wxnz.net
  Addr->IP     : 58.28.20.150 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Def. Username: xxxxxxxxxxxxxxxxxxxxxxxxx
  SIP Options  : 100rel
  Codecs       : 0x10c (ulaw|alaw|g729)
  Codec Order  : (alaw,ulaw,g729)
  Status       : Unmonitored
  Useragent    :
  Reg. Contact :


mybox*CLI> sip debug peer 7985xxxx                                           
SIP Debugging Enabled for IP: 58.28.20.150:5060                              
We're at 111.222.333.444 port 20012                                             
Adding codec 0x8 (alaw) to SDP                                                  
Adding codec 0x4 (ulaw) to SDP                                                  
Adding codec 0x100 (g729) to SDP                                                
Adding non-codec 0x1 (telephone-event) to SDP                                   
13 headers, 13 lines                                                            
Reliably Transmitting (no NAT) to 58.28.20.150:5060:                            
INVITE sip:07308xxxx@pan.wxnz.net SIP/2.0                                       
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK034e0c55                    
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb                        
To: <sip:07308xxxx@pan.wxnz.net>                                                
Contact: <sip:7985xxxx@111.222.333.444>                                         
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444                       
CSeq: 102 INVITE                                                                
User-Agent: Asterisk PBX                                                        
Max-Forwards: 70                                                                
Date: Wed, 09 Dec 2009 20:20:10 GMT                                             
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY              
Content-Type: application/sdp                                                   
Content-Length: 285                                                             

v=0
o=root 4964 4964 IN IP4 111.222.333.444
s=session                             
c=IN IP4 111.222.333.444              
t=0 0                                 
m=audio 20012 RTP/AVP 8 0 18 101      
a=rtpmap:8 PCMA/8000                  
a=rtpmap:0 PCMU/8000                  
a=rtpmap:18 G729/8000                 
a=fmtp:18 annexb=no                   
a=rtpmap:101 telephone-event/8000     
a=fmtp:101 0-16                       
a=silenceSupp:off - - - -                                                                                                                                                          
                                                                                                                                                                                   
---                                                                                                                                                                                
mybox*CLI>                                                                                                                                                                         
<-- SIP read from 58.28.20.150:5060:                                                                                                                                               
SIP/2.0 100 Trying                                                                                                                                                                 
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb                                                                                                                           
To: <sip:07308xxxx@pan.wxnz.net>                                                                                                                                                   
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444                                                                                                                          
CSeq: 102 INVITE                                                                                                                                                                   
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK034e0c55                                                                                                                       
Content-Length: 0                                                                                                                                                                  


--- (7 headers 0 lines) ---
mybox*CLI>                
<-- SIP read from 58.28.20.150:5060:
SIP/2.0 401 Unauthorized            
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
To: <sip:07308xxxx@pan.wxnz.net>;tag=3201400a-13c4-4b20067a-63fedcc-56cac022
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444                     
CSeq: 102 INVITE                                                              
WWW-Authenticate: DIGEST realm="xport.co.nz", nonce="BroadWorksXg30jo4chTcm196oBW", algorithm=MD5, qop="auth"
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK034e0c55                                                   
Content-Length: 0                                                                                              


--- (8 headers 0 lines) ---
Transmitting (no NAT) to 58.28.20.150:5060:
ACK sip:07308xxxx@pan.wxnz.net SIP/2.0    
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK034e0c55
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb   
To: <sip:07308xxxx@pan.wxnz.net>;tag=3201400a-13c4-4b20067a-63fedcc-56cac022
Contact: <sip:7985xxxx@111.222.333.444>                                       
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444                     
CSeq: 102 ACK                                                                 
User-Agent: Asterisk PBX                                                      
Max-Forwards: 70                                                              
Content-Length: 0                                                             


---
We're at 111.222.333.444 port 20012
Adding codec 0x8 (alaw) to SDP    
Adding codec 0x4 (ulaw) to SDP    
Adding codec 0x100 (g729) to SDP  
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 58.28.20.150:5060:
INVITE sip:07308xxxx@pan.wxnz.net SIP/2.0          
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK6cc544e9
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb   
To: <sip:07308xxxx@pan.wxnz.net>                                   
Contact: <sip:7985xxxx@111.222.333.444>                               
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444             
CSeq: 103 INVITE                                                      
User-Agent: Asterisk PBX                                              
Max-Forwards: 70                                                      
Authorization: Digest username="1lHagsbl9uiliw7Hgz", realm="xport.co.nz", algorithm=MD5, uri="sip:07308xxxx@pan.wxnz.net", nonce="BroadWorksXg30jo4chTcm196oBW", response="2a20efb71593c008330bd56b3903ebbf", opaque="", qop=auth, cnonce="09474053", nc=00000001                                                                                                      
Date: Wed, 09 Dec 2009 20:20:10 GMT                                                                                                                                                
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY                                                                                                                 
Content-Type: application/sdp                                                                                                                                                      
Content-Length: 285                                                                                                                                                                

v=0
o=root 4964 4965 IN IP4 111.222.333.444
s=session                             
c=IN IP4 111.222.333.444              
t=0 0                                 
m=audio 20012 RTP/AVP 8 0 18 101      
a=rtpmap:8 PCMA/8000                  
a=rtpmap:0 PCMU/8000                  
a=rtpmap:18 G729/8000                 
a=fmtp:18 annexb=no                   
a=rtpmap:101 telephone-event/8000     
a=fmtp:101 0-16                       
a=silenceSupp:off - - - -             

---
mybox*CLI>
<-- SIP read from 58.28.20.150:5060:
SIP/2.0 100 Trying                  
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
To: <sip:07308xxxx@pan.wxnz.net>                                   
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444             
CSeq: 103 INVITE                                                      
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK6cc544e9          
Content-Length: 0                                                     


--- (7 headers 0 lines) ---
mybox*CLI>                
<-- SIP read from 58.28.20.150:5060:
SIP/2.0 183 Session Progress        
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
To: <sip:07308xxxx@pan.wxnz.net>;tag=3201400a-13c4-4b20067a-63fedcc-56cac022
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444                     
CSeq: 103 INVITE                                                              
Session: Media                                                                
Remote-Party-ID: <sip:07308xxxx@10.251.1.11;user=phone>;screen=yes;party=called;privacy=off;id-type=subscriber
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE                                          
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK6cc544e9                                                    
Contact: <sip:07308xxxx@pan.wxnz.net:5060;maddr=58.28.20.150;transport=udp>                                     
Content-Type: application/sdp                                                                                   
Content-Length: 214                                                                                             

v=0
o=BroadWorks 6434586 1 IN IP4 58.28.20.150
s=-                                      
c=IN IP4 58.28.20.150                    
t=0 0                                    
m=audio 35676 RTP/AVP 8 101              
a=rtpmap:8 PCMA/8000                     
a=rtpmap:101 telephone-event/8000        
a=fmtp:101 0-16                          
a=bsoft:1 image udptl t38

--- (12 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 58.28.20.150:35676
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Destroying call '0995ec382c90ae481394b5e25d64f715@111.222.333.444'
mybox*CLI>
<-- SIP read from 58.28.20.150:5060:
BYE sip:7985xxxx@111.222.333.444 SIP/2.0
From: <sip:07308xxxx@pan.wxnz.net>;tag=3201400a-13c4-4b20067a-63fedcc-56cac022
To: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444
CSeq: 1 BYE
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-3c390-4b200686-64019cc-152505d7
Max-Forwards: 69
Content-Length: 0


--- (8 headers 0 lines) ---
Sending to 58.28.20.150 : 5060 (non-NAT)
Transmitting (no NAT) to 58.28.20.150:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-3c390-4b200686-64019cc-152505d7;received=58.28.20.150
From: <sip:07308xxxx@pan.wxnz.net>;tag=3201400a-13c4-4b20067a-63fedcc-56cac022
To: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7985xxxx@111.222.333.444>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---

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205 posts

Master Geek
+1 received by user: 1

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WorldxChange

  Reply # 282106 14-Dec-2009 08:35
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PM me your number, it sounds like the Asterisk box is not generating an ACK to the 200 OK message when the call is answered. I won't know until I can check some of the call details. Usually this will have something to do with the modem having a SIP ALG or ports not being opened up.

If there is a SIP ALG on the router, I would recommend disabling it.

Cheers




294 posts

Ultimate Geek


  Reply # 283806 18-Dec-2009 15:21
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Actually, I have a similar problem too, on my SPA 3102...

The helpdesk guy did say it's probably my router, but as I use a "semi-public" broadband over here, it's hard to go configure the router that has probably hundreds of connections...

Is there any way that Xnet can do anything about this on their side?

Thanks. Ref: #232458

 
 
 
 


197 posts

Master Geek
+1 received by user: 2

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  Reply # 283927 19-Dec-2009 00:14
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I've found this can happen a bit on Cisco routers - for some reason their NAT table can end up with multiple translations for port 5060 e.g. I see instances where:

asterisk:5060 wanip:5060
asterisk:5060 wanip:1084
asterisk:5060 wanip:8923

Clearing the translation table resolves this. It's probably an IOS bug in recent versions but I haven't had time to create a TAC case about this. Also bear in mind IOS 12.4 has a SIP ALG enabled by default.

As a more permanent fix on any router, try adding a static NAT translation (AKA port forward) for port 5060 through to your Asterisk box. This causes all the traffic to be NAT'd to 5060 instead of the above behaviour.

Regarding the SPA3102 - WxC have a profile that moves SIP to a non-standard port to get around the SIP ALG and/or lots of SIP users issues. Call and ask the help desk to change your profile to the 8060 profile.



21 posts

Geek
+1 received by user: 5

Subscriber

  Reply # 283945 19-Dec-2009 08:10
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After some investigation (thanks to Josh) it would appear that my colo provider is messing with SIP traffic. The "200 - OK" on call answer was not getting through to my asterisk box.

I have just reconfigured Asterisk to register on port 8060 (thanks bender), and all seems to be working perfectly!

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