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Inbound FXO call will time out and disconnect

If I receive an inbound call on an FXO port on a 2901 router running CME, and it routes to a SIP connection, the call will timeout after 3 minutes if the call is not answered.   The router sends a SIP cancel request after the 3 minutes, and it doesn't appear that the router is trying to re-route the call to another destination.   Ideally, the call would ring until answered on the SIP side or the party calling into the FXO port hangs up.

I have tried the "timeouts ringing infinity" and "timeouts ringing 360" (I tried 360 in case the infinity setting was broken) in the voice-port section, and also tried the "timeouts ringing 6000" in the "telephony-service" section.   Regardless of any settings, it always seems to use the default of 180 seconds (3 minutes).

For some background on my application:   I have 8 FXO ports on the router and I register 8 SIP clients to the router.   Each SIP client id is 1111101, 11111102, through 1111108.    I want any call received on FXO port 1 to always route to SIP client 1111101, and any call placed from SIP client 1111101 to always be placed on FXO port 1.  Likewise, any call received on FXO port 2 to always route to SIP client 1111102, and any call placed from SIP client 1111102 to always be placed on FXO port 2.    This configuration is working fine, but the call times out as described above.   I am giving this background in case my configuration might not be optimal and is bypassing the needed settings....  I'm open to a different configuration if needed.

I have attached my router configuration and captures for "debug ccsip all", "debug voip ccapi inout", and "debug vpm signal".

Thanks,

Mike

1 ACCEPTED SOLUTION

Accepted Solutions
Bronze

Inbound FXO call will time out and disconnect

As per RFC 3261 if SIP Invite contains Expire header and no final response like 200 Ok is received for the Invite then session will be Canceled/disconnected as soon as timer expires. Below is the reference from RFC

13.3.1 Processing of the INVITE

1. If the request is an INVITE that contains an Expires header

field, the UAS core sets a timer for the number of seconds

indicated in the header field value. When the timer fires, the

invitation is considered to be expired.

16.10 CANCEL Processing

A stateful proxy MAY generate CANCEL requests for pending INVITE

client transactions based on the period specified in the INVITE’s

Expires header field elapsing. However, this is generally

unnecessary since the endpoints involved will take care of signaling

the end of the transaction.

As per SIP logs I can see in SIP Invite we are sending "Expire" timer as 180 sec (Default value) so call will be disconnected after 180 sec (3 min)

Expires: 180

To increase this timer configure below mentioed command. I don't see any option to set the "expire" value to Infinite but we can increase it to 5 hours (1800000 ms). As shown below

PSTN(config)#sip-ua

PSTN(config-sip-ua)#timers expires ?

  <60000-1800000>  expires timer value in milliseconds

PSTN(config-sip-ua)#timers expires 18000000

you can check the current expire timer value using below command

PSTN#sh sip-ua timers

SIP UA Timer Values (millisecs unless noted)

trying 500, expires 180000, connect 500, disconnect 500

prack 500, rel1xx 500, notify 500, update 500

refer 500, register 500, info 500, options 500, hold 2880 minutes

tcp/udp aging 5 minutes

tls aging 60 minutes

Regards,

Mohit Singh

3 REPLIES
Bronze

Inbound FXO call will time out and disconnect

As per RFC 3261 if SIP Invite contains Expire header and no final response like 200 Ok is received for the Invite then session will be Canceled/disconnected as soon as timer expires. Below is the reference from RFC

13.3.1 Processing of the INVITE

1. If the request is an INVITE that contains an Expires header

field, the UAS core sets a timer for the number of seconds

indicated in the header field value. When the timer fires, the

invitation is considered to be expired.

16.10 CANCEL Processing

A stateful proxy MAY generate CANCEL requests for pending INVITE

client transactions based on the period specified in the INVITE’s

Expires header field elapsing. However, this is generally

unnecessary since the endpoints involved will take care of signaling

the end of the transaction.

As per SIP logs I can see in SIP Invite we are sending "Expire" timer as 180 sec (Default value) so call will be disconnected after 180 sec (3 min)

Expires: 180

To increase this timer configure below mentioed command. I don't see any option to set the "expire" value to Infinite but we can increase it to 5 hours (1800000 ms). As shown below

PSTN(config)#sip-ua

PSTN(config-sip-ua)#timers expires ?

  <60000-1800000>  expires timer value in milliseconds

PSTN(config-sip-ua)#timers expires 18000000

you can check the current expire timer value using below command

PSTN#sh sip-ua timers

SIP UA Timer Values (millisecs unless noted)

trying 500, expires 180000, connect 500, disconnect 500

prack 500, rel1xx 500, notify 500, update 500

refer 500, register 500, info 500, options 500, hold 2880 minutes

tcp/udp aging 5 minutes

tls aging 60 minutes

Regards,

Mohit Singh

New Member

Inbound FXO call will time out and disconnect

Perfect!   I tried the change and it works great (I tested 6 minutes for the timeout).   5 hours should be sufficient for our needs.

Thank you for the detailed explanation and for taking time to look into this!

Mike

New Member

Inbound FXO call will time out and disconnect

As an update, I tested the limits on the 2901 router that we have, and the maximum that it can be set to is 1800000 (30 minutes).   I tested this limit and it works fine.

Still a workable solution for us.

Mike

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