SIP min-se value - Unable to complete a call

[Alex Rich]

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

[Manish Prasad · Accepted solution]

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

[Ayodeji Okanlawon]

+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...

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

[Alex Rich]

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

[Ayodeji Okanlawon]

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"

[Alex Rich]

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.

[Ayodeji Okanlawon]

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"

[asafayan]

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

[Alex Rich]

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-CB4

To: <9809495529>;tag=SDs8r0899-032179045464501344_c1b07.2.1.1386311794229.0_4209973_8343285

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.

422 Session Interval Too Small

The received request contains a Session-Expires header field with a duration below the minimum timer

[Manish Prasad]

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

[Ayodeji Okanlawon]

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"

[Jon Clark1]

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

[Alex Rich]

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.

[Ayodeji Okanlawon]

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"

[Alex Rich]

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.