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btilson
New Contributor
‎12-27-2012 02:18 PM

SIP Proxy Altering SIP Call ID - Prepending ADTN-FAILOVER

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We have a situation were inbound calls through an Adtran NetVanta 6355 using SIP Proxy are failing to Polycom SoundPoint IP phones.  The NetVanta SIP Proxy is changing the SIP Call-ID by prepending "ADTN-FAILOVER", which is a new call id and not recognized by the SBC.

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Anonymous
Not applicable
‎01-10-2013 03:51 PM

Re: SIP Proxy Altering SIP Call ID - Prepending ADTN-FAILOVER

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Btilson,

I wanted to update this ticket with what was found after calling Technical Support on this issue.  This issue was caused by a software bug that was resolved in A4.10.00 and A5.03.00.  The issues stems from not having a valid default route in the unit.  Your options were to enter a default route that points out an active and "up" interface, or upgrade the firmware version on the unit.

I'm going to go ahead and mark this thread as "assumed answered", but feel free to post any additional questions you may have.

Thanks!

David

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btilson
New Contributor
‎12-27-2012 02:24 PM

Re: SIP Proxy Altering SIP Call ID - Prepending ADTN-FAILOVER

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Related firewall and SIP messaging from the NetVanta:

(IP addresses, etc removed)

2012.12.21 15:02:35 FIREWALL.SIP Intercept SipInterceptALG: [Phone Private IP]->[NetVanta Loop1 Public IP] len=645 data="SIP/2.0 100 Trying"

2012.12.21 15:02:35 FIREWALL applyFirewallPolicy: vlan 100 policy-class Private [Phone Private IP]:5060 -> ppp 1 policy-class Public [SBC Public IP]:5060 protocol 17: nat source [NetVanta Loop1 Public IP] policy matched

2012.12.21 15:02:35 FIREWALL applyFirewallPolicy: ppp 1 policy-class Public [SBC Public IP]:5060 -> ppp 1 policy-class Public [SBC Public IP]:5060 protocol 17: allowing reflexive traffic

2012.12.21 15:02:35 FIREWALL.SIP Intercept SipInterceptALG: [NetVanta Loop1 Public IP]->[SBC Public IP] len=556 data="SIP/2.0 100 Trying"

15:02:35.249 SIP.STACK MSG     Rx: UDP src=[Phone Private IP]:5060 dst=[NetVanta Loop1 Public IP]:5060

15:02:35.249 SIP.STACK MSG         SIP/2.0 180 Ringing

15:02:35.249 SIP.STACK MSG         Via: SIP/2.0/UDP [NetVanta Loop1 Public IP]:5060;branch=z9hG4bK-963-5549AE53-c0faab0a148a841dc420a33be0cac56b.AdtnSeq.e35jk6

15:02:35.249 SIP.STACK MSG         Via: SIP/2.0/UDP [SBC Public IP]:5060;branch=z9hG4bKoc39cm30a8bgn5h1e2d0.1

15:02:35.249 SIP.STACK MSG         From: "[Caller]" <sip:[Orig TN]@[NetVanta Loop1 Public IP]>;tag=[Softswitch Private IP]+1+39d60a+46d728be

15:02:35.250 SIP.STACK MSG         To: "[Term TN]" <sip:[Term TN]@[NetVanta Loop1 Public IP]>;tag=BB4CC764-CDAC68CB

15:02:35.250 SIP.STACK MSG         CSeq: 409559840 INVITE

15:02:35.250 SIP.STACK MSG         Call-ID: BC81E1A7@[Softswitch Private IP]

15:02:35.250 SIP.STACK MSG         Contact: <sip:[Term TN]@[Phone Private IP]>

15:02:35.250 SIP.STACK MSG         Record-Route: <sip:[NetVanta Loop1 Public IP];lr>

15:02:35.250 SIP.STACK MSG         User-Agent: PolycomSoundPointIP-SPIP_335-UA/4.0.2.11532_0004f242b9f6

15:02:35.251 SIP.STACK MSG         Allow-Events: conference,talk,hold

15:02:35.251 SIP.STACK MSG         Accept-Language: en-us,en;q=0.9

15:02:35.251 SIP.STACK MSG         Require: 100rel

15:02:35.251 SIP.STACK MSG         RSeq: 8193

15:02:35.251 SIP.STACK MSG         Content-Length: 0

15:02:35.251 SIP.STACK MSG    

15:02:35.253 SIP.PROXY DOM getKeyFromTrans - "BC81E1A7_409559840_00000002"

15:02:35.253 SIP.PROXY DOM getKeyFromTrans - "BC81E1A7_409559840_00000001"

15:02:35.254 SIP.PROXY ROUTING Transc 0x43f8d28: SIP Response Code = 180 to INVITE.

15:02:35.255 SIP.PROXY DB PUD entry lookup: contact = original contact

15:02:35.255 SIP.PROXY ROUTING ProxyUserObject Key e35jk6  : Resp code 180 in phase CALLING

15:02:35.255 SIP.PROXY ROUTING ProxyUserObject e35jk6  : Set phase step -> 2

15:02:35.256 SIP.PROXY ROUTING ProxyUserObject Key 0 Org : Resp code 180 in phase CALLING

15:02:35.256 SIP.PROXY ROUTING ProxyUserObject 0 Org : Set phase step -> 2

15:02:35.256 SIP.PROXY ROUTING ProxyUserGroupObject BC81E1A7@[Softswitch Private IP] : Action FORWARD PRACK

15:02:35.256 SIP.PROXY ROUTING Using server transaction to forward 180 to [SBC Public IP]:5060 via UDP.

15:02:35.257 SIP.PROXY ROUTING Modify response for proxy

15:02:35.258 SIP.STACK MSG     Tx: UDP src=[NetVanta Loop1 Public IP]:5060 dst=[SBC Public IP]:5060

15:02:35.259 SIP.STACK MSG         SIP/2.0 180 Ringing

15:02:35.259 SIP.STACK MSG         From: "[Caller]"<sip:[Orig TN]@[SBC-Reg-Name]>;tag=[Softswitch Private IP]+1+39d60a+46d728be

15:02:35.259 SIP.STACK MSG         To: "[Term TN]"<sip:[Term TN]@[SBC-Reg-Name]>;tag=BB4CC764-CDAC68CB

15:02:35.259 SIP.STACK MSG         Call-ID: ADTN-FAILOVER-BC81E1A7@[Softswitch Private IP]

15:02:35.259 SIP.STACK MSG         CSeq: 409559840 INVITE

15:02:35.259 SIP.STACK MSG         Via: SIP/2.0/UDP [SBC Public IP]:5060;branch=z9hG4bKoc39cm30a8bgn5h1e2d0.1

15:02:35.259 SIP.STACK MSG         Contact: <sip:[Term TN]@[NetVanta Loop1 Public IP]>

15:02:35.260 SIP.STACK MSG         Record-Route: <sip:[NetVanta Loop1 Public IP];lr>

15:02:35.260 SIP.STACK MSG         User-Agent: PolycomSoundPointIP-SPIP_335-UA/4.0.2.11532_0004f242b9f6

15:02:35.260 SIP.STACK MSG         Accept-Language: en-us,en;q=0.9

15:02:35.260 SIP.STACK MSG         Require: 100rel

15:02:35.260 SIP.STACK MSG         Allow-Events: conference,talk,hold

15:02:35.260 SIP.STACK MSG         RSeq: 8193

15:02:35.261 SIP.STACK MSG         Content-Length: 0

15:02:35.261 SIP.STACK MSG     15:02:35: TX: src=[NetVanta Loop1 Public IP]:5060, dst=[SBC Public IP]:5060, 622 bytes,

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btilson
New Contributor
‎12-27-2012 02:27 PM

Re: SIP Proxy Altering SIP Call ID - Prepending ADTN-FAILOVER

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Note the SIP Call-ID is changed by the SIP Proxy:

BEFORE SIP Proxy

-------------------------------

15:02:35.250 SIP.STACK MSG         Call-ID: BC81E1A7@[Softswitch Private IP]

AFTER SIP Proxy

------------------------------

15:02:35.259 SIP.STACK MSG         Call-ID: ADTN-FAILOVER-BC81E1A7@[Softswitch Private IP]

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Anonymous
Not applicable
‎01-03-2013 10:50 AM

Re: SIP Proxy Altering SIP Call ID - Prepending ADTN-FAILOVER

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Hello and thank you for posting to our forum.

It may be easier to work this through a regular Technical Support ticket.  Please create a support ticket from the support link on our website or call us at 888-423-8726 to open a ticket.  We will need to see your complete configuration and the following debug commands, all run at the same time, for a given test call.

debug sip stack messages

debug sip proxy routing

debug sip proxy database

debug voice verbose

Once we have found the problem, we can post it on the forum.

Regards,

Geoff

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Anonymous
Not applicable
‎01-10-2013 03:51 PM

Re: SIP Proxy Altering SIP Call ID - Prepending ADTN-FAILOVER

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Btilson,

I wanted to update this ticket with what was found after calling Technical Support on this issue.  This issue was caused by a software bug that was resolved in A4.10.00 and A5.03.00.  The issues stems from not having a valid default route in the unit.  Your options were to enter a default route that points out an active and "up" interface, or upgrade the firmware version on the unit.

I'm going to go ahead and mark this thread as "assumed answered", but feel free to post any additional questions you may have.

Thanks!

David

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