SIP INVITE TIMEOUT - 100 trying Sooner

Asterisk Asterisk Support
1

Hello,
I have some issues and hope someone can help me out. I am not expert and my knowledge of asterisk is limited.
I have major issues and lots of calls are being rejected. When calls are delivered to my asterisk server it is failing.
According to the Number provider my server is taking to long to accept invite and by the time invite is accepted the call has timed-out. It looks like i have to increase the response time to accept invite quicker. How and where do I change the response time in asterisk to accept and respond to INVITE faster?

Below is reply from the DID number provider.

We trapped a test call to the number and it appears your switch is not responding to our INVITEs. By the time we get a 100 Trying from you, the call has timed out due to no response from the INVITE. Your switch then sends several 200 OKs followed by a 500 Error. Please see attached PCAP for more information.

Trace:
Via: SIP/2.0/UDP 64.136.174.30:5060;branch=z9hG4bK1sansay94457203rdb1784
Record-Route: sip:sansay94457203rdb1784@64.136.174.30:5060;lr;transport=udp
To: sip:6478477345@174.36.9.2
From: "WIRELESS CALLER " sip:14123104450@64.136.174.30;tag=sansay94457203rdb1784
Call-ID: 680710121-0-248166830@64.136.174.226
CSeq: 1 INVITE
Contact: sip:14123104450@64.136.174.30:5060
Supported: timer
Session-Expires: 1800;refresher=uac
Min-SE: 90
Max-Forwards: 18
Content-Type: application/sdp
Content-Length: 301

v=0
o=Sansay-VSXi 188 1 IN IP4 64.136.174.30
s=Session Controller
c=IN IP4 208.72.121.72
t=0 0
m=audio 32848 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
êæaOÈú   0–´ ¬K° E  @ @‘¡@ˆ®®$ ÄÄküUINVITE sip:6478477345@174.36.9.2:5060 SIP/2.0
Via: SIP/2.0/UDP 64.136.174.30:5060;branch=z9hG4bK1sansay94457203rdb1784
Record-Route: sip:sansay94457203rdb1784@64.136.174.30:5060;lr;transport=udp
To: sip:6478477345@174.36.9.2
From: "WIRELESS CALLER " sip:14123104450@64.136.174.30;tag=sansay94457203rdb1784
Call-ID: 680710121-0-248166830@64.136.174.226
CSeq: 1 INVITE
Contact: sip:14123104450@64.136.174.30:5060
Supported: timer
Session-Expires: 1800;refresher=uac
Min-SE: 90
Max-Forwards: 18
Content-Type: application/sdp
Content-Length: 301

Thank you so much
Francis

2

You mean decrease, rather than increase.

Unfortunately the time delay is already zero, so it is physically impossible to reduce it.

Also, the calling side should simple re-transmit the INVITE, and it should take at least 10 seconds before they fault the call.

My guess is that Asterisk is sending the immediate response, but it is not reaching the provider. That is likely to be a network or NAT issue.

One extreme possibility is that are using realtime and the database is taking excessively long to respond to the lookups needed to fetch the sip.conf information, but I think it much more likely that the 100 Trying is getting lost.

Also, once the switch sends a 200, it cannot send a 500 for the same dialogue, as 200 is a final response. I supposed if they failed to send ACK to the 200, things might just get confused.

Your trace is incomplete and appears to include, binary, IP, UDP and possibly MAC, level headers.