Asterisk call starts to fail few mins after calling starts with retransmission timeout error

Asterisk Asterisk SIP
1

Hello,

I have been experiencing a unique error since yesterday. A sample test was with a file of 1,500 mobile numbers. Call was initiated and was running well until after about 5 mins or so. Instead of the normal flow of calling a line and getting either an answered or unanswered response, it was retransmission timeout. Pls see excerpt from the verbose printout;

WARNING[11846]: chan_sip.c:4126 retrans_pkt: Retransmission timeout reached on transmission 0b4a39fc663a051816ecc41776c7a806@xxxx:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 12673ms with no response

[Apr 5 15:53:36] WARNING[11846]: chan_sip.c:4150 retrans_pkt: Hanging up call 0b4a39fc663a051816ecc41776c7a806@xxxx:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

At first, my call logic was making 10 calls every 5 secs and when this was experienced, i made some change. 2 calls every 5 secs gave the expected result until after an hour or so and subsequently, response was as mentioned earlier.

THis is not a case of trunk capacity as it supports 120 concurrent calls.

Please help.

2

This is likely some sort of network error – this is Asterisk’s way of saying “I sent a critical request to the other host, they didn’t respond, I kept re-transmitting the request, but now I’ve hit my limit of re-transmissions. They’re obviously not responding.”

I would start looking at packet captures on the network to try to find out why the other host isn’t responding to Asterisk’s requests.

3

Thanks Smith. We are now suspecting the SIP provider as network issue was eliminated. All through the calling period, we closely monitored connectivity by using PING and it was flawless. We would escalate to the SIP provider.

Thank You.

4

Hello Friends,

We did more troubleshooting and this was the result;

  1. Calls were made at this rate - 2 calls every 5 secs. It ran for over 30 mins and everything was fine.
  2. Call rate was increased to 20 calls every 5 secs. In less than 3 mins, all calls started to fail with the aforementioned error. Entry in the CDR shows “CONGESTION” and SIP provider did not see any traffic this period. We also monitored the server’s resources and it was okay.

I need help pls.

5

Could be that the provider is rate limiting you, before you get the part of the system that can return a proper 403 response.

6

I just solved it. Phew!

Adding wait() was the answer to this issue that gave me sleepless nights for a week. Asterisk error description did not help matters. I wish it is an insight into the exact issue.

7

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