Hi,
I deployed an asterisk server with a SIP Trunk.
Incoming and outgoing calls work correctly except for a single A number.
When I call this number A, from a GSM number or else
, it rings correctly. But when I call this same number A from my ASTERISK server there is no ringing and we get a SIP 487 error.
Can you help me fix this problem???
You can see the packet capture
I can see an outline of a SIP trace, but not a full SIP trace. In particular, I cannot see to which request the OK and the 487 relate. I assume that 172.16.60.2 is the Asterisk client and 10.203.5.137 is the user agent server handling A, and is what you mean by the SIP trunk.
I would guess the OK is for the ACK and the 487 is for the INVITE.
My guess is that there is a bug in …137 and it is cancelling its outgoing side, and forwarding the 487 from there, rather than sending some sort of time out message.
22.5 seconds seems rather excessive for getting to 183.
According to the RFC section 13.3.1, if the INVITE contains an Expires header, the call will be implicitly cancelled when it expires without having sent a final response.
Again, without the full logging, I can’t tell if there is an Expires header. It is possible that a router is adding one. Make sure any SIP ALG is disabled.
Bonjour, voici les logs.
172.16.50.2.sip > uas-ip-address.sip: [udp sum ok] SIP, length: 1389
INVITE sip:+21524585963@domain.com SIP/2.0
Via: SIP/2.0/UDP 172.16.50.2:5060;rport;branch=z9hG4bKPj6ef8cec3-f5d5-4341-8644-1149542dbf61
From: <sip:+21524585632@domain.com>;tag=706d5c58-f3ba-4f26-ac93-0d0b28a5d92d
To: <sip:+21524585963@domain.com>
Contact: <sip:+21524585632@172.16.50.2:5060>
Call-ID: 4525ea6b-78ba-4ad1-b1d9-cfe544fc29f6
CSeq: 20700 INVITE
Route: <sip:uas-ip-address;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, whiteboard
Session-Expires: 1800
Min-SE: 90
X-GS-Trunk: yes
Remote-Party-ID: <sip:+21524585632@domain.com>;party=calling;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Grandstream UCM6300AV1.1A 1.0.19.10
Content-Type: application/sdp
Content-Length: 554
v=0
o=- 557895134 557895134 IN IP4 172.16.50.2
s=Asterisk
c=IN IP4 172.16.50.2
t=0 0
m=audio 18640 RTP/AVP 9 3 0 8 2 97 101
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=ssrc:939111654 cname:a340f114-9325-4f5a-87a7-429c5aef026b
a=ssrc:939111654 msid:trunk_1_24-audio audio
a=rtcp:18641 IN IP4 172.16.50.2
a=record:off
a=msid:trunk_1_24-audio audio
a=mid:0
uas-ip-address.sip > 172.16.50.2.sip: [udp sum ok] SIP, length: 345
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.50.2:5060;branch=z9hG4bKPj6ef8cec3-f5d5-4341-8644-1149542dbf61;received=10.204.38.138;rport=5060
Call-ID: 4525ea6b-78ba-4ad1-b1d9-cfe544fc29f6
From: <sip:+22534701510@domain.com>;tag=706d5c58-f3ba-4f26-ac93-0d0b28a5d92d
To: <sip:+21524585963@domain.com>
CSeq: 20700 INVITE
Content-Length: 0
uas-ip-address.sip > 172.16.50.2.sip: [udp sum ok] SIP, length: 999
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.16.50.2:5060;branch=z9hG4bKPj6ef8cec3-f5d5-4341-8644-1149542dbf61;received=10.204.38.138;rport=5060
Record-Route: <sip:uas-ip-address:5060;lr;Hpt=nw_ca_65b4db7a_104f3793_ex_8ee2_116;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=3076>
Call-ID: 4525ea6b-78ba-4ad1-b1d9-cfe544fc29f6
From: <sip:+22534701510@domain.com>;tag=706d5c58-f3ba-4f26-ac93-0d0b28a5d92d
To: <sip:+21524585963@domain.com>;tag=eryt2hmp
CSeq: 20700 INVITE
Contact: <sip:uas-ip-address:5060;Hpt=nw_ca_65b4db7a_104f3793_ex_8ee2_16;CxtId=3;TRC=ffffffff-ffffffff>
Require: 100rel
RSeq: 1
Reason: Q.850;cause=102;text="Recovery on timer expiry",SIP;cause=504
P-Early-Media: sendrecv,gated
P-Asserted-Service-Info: vrbt=00
Content-Length: 200
Content-Type: application/sdp
v=0
o=- 15871719 15871719 IN IP4 172.40.4.105
s=SBC call
t=0 0
m=audio 56434 RTP/AVP 0 101
c=IN IP4 172.40.4.105
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=ptime:20
uas-ip-address.sip > 172.16.50.2.sip: [udp sum ok] SIP, length: 558
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.16.50.2:5060;branch=z9hG4bKPj6ef8cec3-f5d5-4341-8644-1149542dbf61;received=10.204.38.138;rport=5060
Call-ID: 4525ea6b-78ba-4ad1-b1d9-cfe544fc29f6
From: <sip:+21524585632@domain.com>;tag=706d5c58-f3ba-4f26-ac93-0d0b28a5d92d
To: <sip:+21524585963@domain.com>;tag=eryt2hmp
CSeq: 20700 INVITE
Warning: 399 uas-ip-address "SS280000F114L701[00000] No Response From Network"
Reason: Q.850;cause=31;text="Normal, unspecified",SIP;cause=487
P-Asserted-Identity: <tel:+225+21524585963>
Content-Length: 0
172.16.50.2.sip > uas-ip-address.sip: [udp sum ok] SIP, length: 448
ACK sip:+21524585963@domain.com SIP/2.0
Via: SIP/2.0/UDP 172.16.50.2:5060;rport;branch=z9hG4bKPj6ef8cec3-f5d5-4341-8644-1149542dbf61
From: <sip:+21524585632@domain.com>;tag=706d5c58-f3ba-4f26-ac93-0d0b28a5d92d
To: <sip:+21524585963@domain.com>;tag=eryt2hmp
Call-ID: 4525ea6b-78ba-4ad1-b1d9-cfe544fc29f6
CSeq: 20700 ACK
Route: <sip:uas-ip-address;lr>
Max-Forwards: 70
User-Agent: Grandstream UCM6300AV1.1A 1.0.19.10
Content-Length: 0
Hi David551,
I have send the logs.
Regards,
The request that is being cancelled appears to be from Grandstream, not Asterisk.
There is no Expires header.
I think 487 is being misused, but you would need to find out what “SS280000F114L701[00000] No Response From Network” means, and to which network it is referring.
“Recovery on timer expiry”,SIP;cause=504 is also worrying. At a guess uas-ip-address.sip is not getting responses from its downstream side, and output 183 just to keep things alive, then sent 487, rather than a no response type error.
Your summary trace showed PRACK, but this one doesn’t.
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