Asterisk and audicodes MP 108 sip FXS


#1

Hi,

i am trying to use asterisk with audicodes mp108 sip fxs. i configured the audicodes and created the sip extensions. but the sip extesnions work gret with xlite. it is not working with audiocodes.

Here is the debug.

I shall be thankfull , if any one can help me figure out the issues.

Rizwan

Reliably Transmitting (no NAT) to xx.xx.xx.xx:0:
OPTIONS sip:xx.xx.xx.xx:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.141:5060;branch=z9hG4bK21265b8b;rport
From: “asterisk” sip:asterisk@192.168.111.141;tag=as3ee125dc
To: sip:xx.xx.xx.xx:0
Contact: sip:asterisk@192.168.111.141
Call-ID: 4519869a7c65d410697eb78d58c9b93c@192.168.111.141
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 15 Mar 2006 20:04:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Mar 15 15:04:02 WARNING[4545]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x87e8e68 (len 485) to xx.xx.xx.xx:-1 returned 0: Invalid argument
Retransmitting #1 (no NAT) to xx.xx.xx.xx:0:
OPTIONS sip:xx.xx.xx.xx:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.141:5060;branch=z9hG4bK21265b8b;rport
From: “asterisk” sip:asterisk@192.168.111.141;tag=as3ee125dc
To: sip:xx.xx.xx.xx:0
Contact: sip:asterisk@192.168.111.141
Call-ID: 4519869a7c65d410697eb78d58c9b93c@192.168.111.141
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 15 Mar 2006 20:04:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Mar 15 15:04:03 WARNING[4545]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x87e8e68 (len 485) to xx.xx.xx.xx:-1 returned 0: Invalid argument
Retransmitting #2 (no NAT) to xx.xx.xx.xx:0:
OPTIONS sip:xx.xx.xx.xx:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.141:5060;branch=z9hG4bK21265b8b;rport
From: “asterisk” sip:asterisk@192.168.111.141;tag=as3ee125dc
To: sip:xx.xx.xx.xx:0
Contact: sip:asterisk@192.168.111.141
Call-ID: 4519869a7c65d410697eb78d58c9b93c@192.168.111.141
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 15 Mar 2006 20:04:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Mar 15 15:04:04 WARNING[4545]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x87e8e68 (len 485) to xx.xx.xx.xx:-1 returned 0: Invalid argument
Retransmitting #3 (no NAT) to xx.xx.xx.xx:0:
OPTIONS sip:xx.xx.xx.xx:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.141:5060;branch=z9hG4bK21265b8b;rport
From: “asterisk” sip:asterisk@192.168.111.141;tag=as3ee125dc
To: sip:xx.xx.xx.xx:0
Contact: sip:asterisk@192.168.111.141
Call-ID: 4519869a7c65d410697eb78d58c9b93c@192.168.111.141
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 15 Mar 2006 20:04:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Mar 15 15:04:05 WARNING[4545]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x87e8e68 (len 485) to xx.xx.xx.xx:-1 returned 0: Invalid argument
Retransmitting #4 (no NAT) to xx.xx.xx.xx:0:
OPTIONS sip:xx.xx.xx.xx:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.141:5060;branch=z9hG4bK21265b8b;rport
From: “asterisk” sip:asterisk@192.168.111.141;tag=as3ee125dc
To: sip:xx.xx.xx.xx:0
Contact: sip:asterisk@192.168.111.141
Call-ID: 4519869a7c65d410697eb78d58c9b93c@192.168.111.141
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 15 Mar 2006 20:04:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Mar 15 15:04:06 WARNING[4545]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x87e8e68 (len 485) to xx.xx.xx.xx:-1 returned 0: Invalid argument
Destroying call '4519869a7c65d410697eb78d58c9b93c@192.168.111.141’
12 headers, 0 lines
Reliably Transmitting (no NAT) to xx.xx.xx.xx:0:
OPTIONS sip:xx.xx.xx.xx:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.141:5060;branch=z9hG4bK7e23e362;rport
From: “asterisk” sip:asterisk@192.168.111.141;tag=as7471ddd3
To: sip:xx.xx.xx.xx:0
Contact: sip:asterisk@192.168.111.141
Call-ID: 6d26ab05000b4b1d31f3d19852f97fc1@192.168.111.141
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 15 Mar 2006 20:04:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Mar 15 15:04:16 WARNING[4545]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x87e8e68 (len 485) to xx.xx.xx.xx:-1 returned 0: Invalid argument
Retransmitting #1 (no NAT) to xx.xx.xx.xx:0:
OPTIONS sip:xx.xx.xx.xx:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.141:5060;branch=z9hG4bK7e23e362;rport
From: “asterisk” sip:asterisk@192.168.111.141;tag=as7471ddd3
To: sip:xx.xx.xx.xx:0
Contact: sip:asterisk@192.168.111.141
Call-ID: 6d26ab05000b4b1d31f3d19852f97fc1@192.168.111.141
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 15 Mar 2006 20:04:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Mar 15 15:04:17 WARNING[4545]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x87e8e68 (len 485) to xx.xx.xx.xx:-1 returned 0: Invalid argument
Retransmitting #2 (no NAT) to xx.xx.xx.xx:0:
OPTIONS sip:xx.xx.xx.xx:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.141:5060;branch=z9hG4bK7e23e362;rport
From: “asterisk” sip:asterisk@192.168.111.141;tag=as7471ddd3
To: sip:xx.xx.xx.xx:0
Contact: sip:asterisk@192.168.111.141
Call-ID: 6d26ab05000b4b1d31f3d19852f97fc1@192.168.111.141
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 15 Mar 2006 20:04:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Mar 15 15:04:18 WARNING[4545]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x87e8e68 (len 485) to xx.xx.xx.xx:-1 returned 0: Invalid argument
Retransmitting #3 (no NAT) to xx.xx.xx.xx:0:
OPTIONS sip:xx.xx.xx.xx:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.141:5060;branch=z9hG4bK7e23e362;rport
From: “asterisk” sip:asterisk@192.168.111.141;tag=as7471ddd3
To: sip:xx.xx.xx.xx:0
Contact: sip:asterisk@192.168.111.141
Call-ID: 6d26ab05000b4b1d31f3d19852f97fc1@192.168.111.141
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 15 Mar 2006 20:04:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Mar 15 15:04:19 WARNING[4545]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x87e8e68 (len 485) to xx.xx.xx.xx:-1 returned 0: Invalid argument
Retransmitting #4 (no NAT) to xx.xx.xx.xx:0:
OPTIONS sip:xx.xx.xx.xx:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.141:5060;branch=z9hG4bK7e23e362;rport
From: “asterisk” sip:asterisk@192.168.111.141;tag=as7471ddd3
To: sip:xx.xx.xx.xx:0
Contact: sip:asterisk@192.168.111.141
Call-ID: 6d26ab05000b4b1d31f3d19852f97fc1@192.168.111.141
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 15 Mar 2006 20:04:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Mar 15 15:04:20 WARNING[4545]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x87e8e68 (len 485) to xx.xx.xx.xx:-1 returned 0: Invalid argument
Destroying call '6d26ab05000b4b1d31f3d19852f97fc1@192.168.111.141’
12 headers, 0 lines
Reliably Transmitting (no NAT) to xx.xx.xx.xx:0:
OPTIONS sip:xx.xx.xx.xx:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.141:5060;branch=z9hG4bK7a72a4aa;rport
From: “asterisk” sip:asterisk@192.168.111.141;tag=as231478f8
To: sip:xx.xx.xx.xx:0
Contact: sip:asterisk@192.168.111.141
Call-ID: 7c3532e63264330f506b2b2c14ddc749@192.168.111.141
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 15 Mar 2006 20:04:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Mar 15 15:04:30 WARNING[4545]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x87e8e68 (len 485) to xx.xx.xx.xx:-1 returned 0: Invalid argument
Retransmitting #1 (no NAT) to xx.xx.xx.xx:0:
OPTIONS sip:xx.xx.xx.xx:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.141:5060;branch=z9hG4bK7a72a4aa;rport
From: “asterisk” sip:asterisk@192.168.111.141;tag=as231478f8
To: sip:xx.xx.xx.xx:0
Contact: sip:asterisk@192.168.111.141
Call-ID: 7c3532e63264330f506b2b2c14ddc749@192.168.111.141
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 15 Mar 2006 20:04:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Mar 15 15:04:31 WARNING[4545]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x87e8e68 (len 485) to xx.xx.xx.xx:-1 returned 0: Invalid argument
localhost*CLI>

xx.xx.xx.xx- ip of audiocdes
192.168.111.141 is the ip of aasteisk

Both the boxes in the static ip.


#2

Any help may be appreciated.


#3

is there nat in this picture?


#4

Yes… asterisk behind the nat


#5

make sure asterisk has its external IP set and double check the redirects on your firewall/nat router

the AudioCodes MP100 series gateways send their responses back to the from in the sip msg and my guess would be those packets are going in to never never land


#6

hi ,

i have opened udp 5060 and udp 10k-20k for thsi asterisk ip and set externip in the sip.conf.

what else i need to check Mr.Swk?

i shall be thankfull , if you help me to fix this issues.

can you tell me yahoo im address…?

My im :rizu2k@yahoo.com


#7

you may have to do a redirect on your firewall so that those UDP ports are redirected over to asterisk


#8

hi,

i have took out the asterisk from firewall and it seems the audicodes registering with asterisk…

but i get follwing

Got SIP response 481 “Call/Transaction Does Not Exist” back from xx.xx.xx.xx
in cli prompt…

what is the error? and how do i resolve it?

Rizwan


#9

can you post a sip debug of whats happening here?

sip debug peer AUDIO_CODES_NAME

or

sip debug ip IP.OF.AC.GW


#10

Hi ,

i did it.

below is the sip debug

xx.xx.xx.xx is the audiocode ip
yy.yy.yy.yy. is asterisk ip.

i configured 218 as ext in the audiocodes end point number and authentication.

Looking for 218 in default (domain xx.xx.xx.xx)
— (12 headers 0 lines)—
Looking for 218 in default (domain xx.xx.xx.xx)
Transmitting (NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKacsGbmGav;received=xx.xx.xx.xx
From: sip:218@xx.xx.xx.xx;tag=1c812614970
To: sip:218@xx.xx.xx.xx;tag=as3dd70381
Call-ID: 1392621620ziCX@xx.xx.xx.xx
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:yy.yy.yy.yy
Accept: application/sdp
Content-Length: 0


Transmitting (NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKacsGbmGav;received=xx.xx.xx.xx
From: sip:218@xx.xx.xx.xx;tag=1c812614970
To: sip:218@xx.xx.xx.xx;tag=as3dd70381
Call-ID: 1392621620ziCX@xx.xx.xx.xx
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:yy.yy.yy.yy
Accept: application/sdp
Content-Length: 0


Destroying call '1392621620ziCX@xx.xx.xx.xx
Destroying call '1392621620ziCX@xx.xx.xx.xx
12 headers, 0 lines
Reliably Transmitting (NAT) to xx.xx.xx.xx:5060:
OPTIONS sip:218@xx.xx.xx.xx;user=phone SIP/2.0
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK4df64e75
From: “asterisk” sip:asterisk@yy.yy.yy.yy;tag=as0b9ec0a0
To: sip:218@xx.xx.xx.xx;user=phone
Contact: sip:asterisk@yy.yy.yy.yy
Call-ID: 5645a2123f36448d01461c7a235fceb6@yy.yy.yy.yy
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 16 Mar 2006 20:31:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


12 headers, 0 lines
Reliably Transmitting (NAT) to xx.xx.xx.xx:5060:
OPTIONS sip:218@xx.xx.xx.xx;user=phone SIP/2.0
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK4df64e75
From: “asterisk” sip:asterisk@yy.yy.yy.yy;tag=as0b9ec0a0
To: sip:218@xx.xx.xx.xx;user=phone
Contact: sip:asterisk@yy.yy.yy.yy
Call-ID: 5645a2123f36448d01461c7a235fceb6@yy.yy.yy.yy
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 16 Mar 2006 20:31:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


yy-yy-yy-yy*CLI>
<-- SIP read from xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK4df64e75
From: “asterisk” sip:asterisk@yy.yy.yy.yy;tag=as0b9ec0a0
To: sip:218@xx.xx.xx.xx;user=phone;tag=1c522626182
Call-ID: 5645a2123f36448d01461c7a235fceb6@yy.yy.yy.yy
CSeq: 102 OPTIONS
Contact: sip:xx.xx.xx.xx;user=phone
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Content-Length: 0

— (10 headers 0 lines)—

<-- SIP read from xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK4df64e75
From: “asterisk” sip:asterisk@yy.yy.yy.yy;tag=as0b9ec0a0
To: sip:218@xx.xx.xx.xx;user=phone;tag=1c522626182
Call-ID: 5645a2123f36448d01461c7a235fceb6@yy.yy.yy.yy
CSeq: 102 OPTIONS
Contact: sip:xx.xx.xx.xx;user=phone
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Content-Length: 0

Destroying call '5645a2123f36448d01461c7a235fceb6@yy.yy.yy.yy
— (10 headers 0 lines)—
Destroying call '5645a2123f36448d01461c7a235fceb6@yy.yy.yy.yy
yy-yy-yy-yy*CLI>
<-- SIP read from xx.xx.xx.xx:5060:
OPTIONS sip:218@xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKacjybvQCW
Max-Forwards: 70
From: sip:218@xx.xx.xx.xx;tag=1c2486314868
To: sip:218@xx.xx.xx.xx
Call-ID: 138245588AkEZ@xx.xx.xx.xx
CSeq: 1 OPTIONS
Contact: sip:218@xx.xx.xx.xx;user=phone
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-108 FXS/v.4.40.200.371
Content-Length: 0

— (12 headers 0 lines)—

<-- SIP read from xx.xx.xx.xx:5060:
OPTIONS sip:218@xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKacjybvQCW
Max-Forwards: 70
From: sip:218@xx.xx.xx.xx;tag=1c2486314868
To: sip:218@xx.xx.xx.xx
Call-ID: 138245588AkEZ@xx.xx.xx.xx
CSeq: 1 OPTIONS
Contact: sip:218@xx.xx.xx.xx;user=phone
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Looking for 218 in default (domain xx.xx.xx.xx).40.200.3
71
Content-Length: 0

— (12 headers 0 lines)—
Transmitting (NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKacjybvQCW;received=xx.xx.xx.xx
From: sip:218@xx.xx.xx.xx;tag=1c2486314868
To: sip:218@xx.xx.xx.xx;tag=as2ddf5f8d
Call-ID: 138245588AkEZ@xx.xx.xx.xx
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:yy.yy.yy.yy
Accept: application/sdp
Content-Length: 0


Looking for 218 in default (domain xx.xx.xx.xx)
Destroying call '138245588AkEZ@xx.xx.xx.xx
Transmitting (NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKacjybvQCW;received=xx.xx.xx.xx
From: sip:218@xx.xx.xx.xx;tag=1c2486314868
To: sip:218@xx.xx.xx.xx;tag=as2ddf5f8d
Call-ID: 138245588AkEZ@xx.xx.xx.xx
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:yy.yy.yy.yy
Accept: application/sdp
Content-Length: 0


Destroying call '138245588AkEZ@xx.xx.xx.xx
12 headers, 0 lines
Reliably Transmitting (NAT) to xx.xx.xx.xx:5060:
OPTIONS sip:218@xx.xx.xx.xx;user=phone SIP/2.0
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK5af3c962
From: “asterisk” sip:asterisk@yy.yy.yy.yy;tag=as175882ec
To: sip:218@xx.xx.xx.xx;user=phone
Contact: sip:asterisk@yy.yy.yy.yy
Call-ID: 408fa7303bfafdf01a117212178a60cf@yy.yy.yy.yy
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 16 Mar 2006 20:32:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


12 headers, 0 lines
Reliably Transmitting (NAT) to xx.xx.xx.xx:5060:
OPTIONS sip:218@xx.xx.xx.xx;user=phone SIP/2.0
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK5af3c962
From: “asterisk” sip:asterisk@yy.yy.yy.yy;tag=as175882ec
To: sip:218@xx.xx.xx.xx;user=phone
Contact: sip:asterisk@yy.yy.yy.yy
Call-ID: 408fa7303bfafdf01a117212178a60cf@yy.yy.yy.yy
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 16 Mar 2006 20:32:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


yy-yy-yy-yy*CLI>
<-- SIP read from xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK5af3c962
From: “asterisk” sip:asterisk@yy.yy.yy.yy;tag=as175882ec
To: sip:218@xx.xx.xx.xx;user=phone;tag=1c36168643
Call-ID: 408fa7303bfafdf01a117212178a60cf@yy.yy.yy.yy
CSeq: 102 OPTIONS
Contact: sip:xx.xx.xx.xx;user=phone
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Content-Length: 0

— (10 headers 0 lines)—

<-- SIP read from xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK5af3c962
From: “asterisk” sip:asterisk@yy.yy.yy.yy;tag=as175882ec
To: sip:218@xx.xx.xx.xx;user=phone;tag=1c36168643
Call-ID: 408fa7303bfafdf01a117212178a60cf@yy.yy.yy.yy
CSeq: 102 OPTIONS
Contact: sip:xx.xx.xx.xx;user=phone
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Content-Length: 0

Destroying call '408fa7303bfafdf01a117212178a60cf@yy.yy.yy.yy
— (10 headers 0 lines)—
Destroying call '408fa7303bfafdf01a117212178a60cf@yy.yy.yy.yy
yy-yy-yy-yy*CLI>


#11

best thing to do at this time is try to duplicate that and match it up to a sip trace

I dont see any 481s in there

I would start placing calls and see how well its working at this time


#12

hi

here is the 481

Got SIP response 481 “Call/Transaction Does Not Exist” back from xx.xx.xx.xx in cli prompt.

please help


#13

how do trace the sip using sip trace?


#14

sip debug ip ip.of.peer.device is one way
sip debug peer name.of.peer.from.sip.conf is another

what you pasted about is a sip trace…

in that trace I dont see the 481

The 481 back from ip address can mean a number of things, but if the calls are working correctly its probably not a big deal.

on the other hand to figure out exactly what is happening we’re going to need to see that 481 message in the sip debug


#15

hi ,

i have noticed while doing this…

Transmitting (NAT) to xx.xx.xx.xx:5060:
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKacfuOjDYA;received=xx.xx.xx.xx
From: sip:282@xx.xx.xx.xx;tag=1c1441221255
To: sip:282@xx.xx.xx.xx;tag=as7db3f4d1
Call-ID: 428111067slFi@xx.xx.xx.xx
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:282@yy.yy.yy.yy
Content-Length: 0

Can you help debuggin the same?


#16

Also i keep on watching the debug while on call…

Please find the follwoing and help me to fix…

<-- SIP read fromxx.xx.xx,xx:5060:
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK0668a5ae;rport
From: “asterisk” sip:asterisk@yy.yy.yy.yy;tag=as300305c8
To: sip:218@xx.xx.xx.xx;user=phone;tag=1c985930815
Call-ID: 359c92ef4a3c0db7169ee7c25e5e7cbf@yy.yy.yy.yy
CSeq: 102 NOTIFY
Contact: sip:xx.xx.xx.xx;user=phone
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
– Got SIP response 481 “Call/Transaction Does Not Exist” back fromxx.xx.xx,xx
Content-Length: 0I>