ā02-04-2014 01:08 PM - edited ā03-16-2019 09:36 PM
We are transitioning over to SIP. We have run into an issue calling one of our customers MetLife 980-949-5529. When we try to place calls to them we are getting error 422 - session interval too small. MetLife has their min-se timer setting set to 6000 while we have ours set to the default of 1800. We are unable to increase ours due to calls coming from our clients are being rejected since they are using the default value. Is there anyway to correct this issue without causing an issue from other clients calling into us with the default?
show sip-ua min-se
SIP UA MIN-SE Value (seconds)
Min-SE: 1800
Thanks,
Alex
Solved! Go to Solution.
ā02-04-2014 11:57 PM
Hi Alex,
If you change session timer at global level then it will affect all sip calls which you don't want to do. Then one more option left is to change Min-Se through sip profile and then apply it to outgoing dial-peer. Note - this only applied when you make outbound calls to metlife not inbound call from metlife.
To achive this you need to create separate dial-peer for your customer Metlife and map the sip profile into it.
voice class sip-profiles 1
request INVITE sip-header Min-SE modify "1800" "6000"
dial-peer voice xxxxx voip
destination-pattern xxxxxxxxx
voice-class sip profiles 1
Rate all the helpful post.
Thanks
Manish
ā02-04-2014 11:57 PM
Hi Alex,
If you change session timer at global level then it will affect all sip calls which you don't want to do. Then one more option left is to change Min-Se through sip profile and then apply it to outgoing dial-peer. Note - this only applied when you make outbound calls to metlife not inbound call from metlife.
To achive this you need to create separate dial-peer for your customer Metlife and map the sip profile into it.
voice class sip-profiles 1
request INVITE sip-header Min-SE modify "1800" "6000"
dial-peer voice xxxxx voip
destination-pattern xxxxxxxxx
voice-class sip profiles 1
Rate all the helpful post.
Thanks
Manish
ā02-05-2014 12:14 AM
+5 Manish...This is what I was going to suggest and was going to send it out yesterday, but the app on my Android Tab wont work...
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
ā02-05-2014 08:48 AM
Thank you this is exactly what I was looking for. Unfortunately I am still getting a busy signal when calling. If I change the min-se for all calls it will ring through so I dont think there are any other issues causing this. Any other suggestions?
voice class sip-profiles 2
response ANY sip-header Allow-Header modify "UPDATE," ""
request INVITE sip-header Min-SE modify "1800" "6000"
request INVITE sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
response ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
request REINVITE sdp-header Attribute modify "a=T38FaxFillBitRemoval:0" ""
request INVITE sdp-header Audio-Attribute add "a=ptime:30"
dial-peer voice 110 voip
description Outgoing MetLife
destination-pattern 919809495529
translate-outgoing called 1
session protocol sipv2
session target sip-server
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
no voice-class sip outbound-proxy
voice-class sip early-offer forced
voice-class sip profiles 2
dtmf-relay rtp-nte
no dtmf-interworking
ā02-05-2014 08:50 AM
can you send us debug ccsip messages..please include calling and called number
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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
ā02-05-2014 09:07 AM
Adding this line has resolved my issue.
request INVITE sip-header Session-Expires modify "1800" "6000"
I wish ATT / TAC were as helpful as these forums. It took you all 10 min to resolve what has been a massive headache of mine for almost a month now.
ā02-05-2014 09:10 AM
Alex,
Thats what we are about here...I was up till 1am yesterday trying to send you the idea Manish put forward, but my Tab was down and had already shutdwon my laptop..Just to let you know we go the extra mile here and we love it!!!!
Glad we could help
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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
ā02-05-2014 09:14 AM
Another victory by the great engineers on this forum.
Alex - if you don't mind - would you please add the ccsip messages to this thread so I can see the actual messaging
resulting in the 422 - session interval too small error?
Thanks,
Amir
ā02-05-2014 10:35 AM
Amir please let me know if you would like anything else. I did change some of the IP addresses / phone number as I am not sure I am allowed to post that data.
Feb 5 10:55:06.327 cst: //1170969/41A840000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 422 Session Interval Too Small
Via: SIP/2.0/UDP 32.242.1.78:5060;branch=z9hG4bK9BD411DA
From: "Rich, Alex" <8124807384>;tag=6E93E5AE-CB48124807384>
To: <9809495529>;tag=SDs8r0899-032179045464501344_c1b07.2.1.1386311794229.0_4209973_83432859809495529>
Call-ID: 19142581-8DBD11E3-9CB780DD-E3500D13@32.242.1.78
CSeq: 101 INVITE
Timestamp: 1391619306
Min-SE: 6000
Server: Avaya CM/R015x.02.1.016.4 AVAYA-SM-6.2.3.0.623006
Content-Length: 0
Once I had that error code and it still wouldnt work with the correct min-se setting I got the below info and added the session-expires line to the sip-profile I created for the dial-peer.
ā02-05-2014 10:04 AM
Deji ..we learned a lot from your posts on this forum specially on SIP. I love the way you help people till the end on this forum. Really you are doing a great job. Hi 5 dear.
Sent from Cisco Technical Support iPhone App
ā02-05-2014 10:53 AM
Manish..Thank you for the nice words and keep up the great work you are doing too. Well done
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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
ā11-04-2017 05:24 AM
Deji,
I have a similar but more modern problem such as this.
Problem: 8841 user at home using MRA with physical phone... all calls drop after 15 mins.
After viewing CUCM traces we see the incoming invite from Expressway-C has min-se 500.
CUCM replies "422 Min-Se too short" and has a global min-se of 3600
There is a min-se of 500 in Expressway-C configuration->protocol->SIP
Phone firmware shows "TFTP Failed" trying to upgrade from 10-3-1 to 11-0-1
Do you think firmware upgrade would fix? No other home user has this issue but there are using 11-0-1
I'm afraid to change min-se on expressway or cucm.
Thanks
-JC
ā02-13-2014 07:00 AM
As we convert more locations to SIP we are starting to see more and more of this issue. Are there any other solutions then making a global change or dial-peers? We are hoping to avoid having to create a new dial peer every time this issue comes up. Thanks again for the assistance.
ā02-13-2014 07:44 AM
Alex,
Whats your architecture like? Does each of your remote locations have a dedicated gateway that have direct sip trunks to your ITSP or do remote sites go via the HQ router?
Why do you need to define multiple dial-peers?
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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
ā02-13-2014 07:49 AM
SIP trunks are at our HQ.
We are having the same issue calling multiple numbers with different organizations. We are created a dial-peer for each of these numbers. I was hoping to there was another solution.
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