SIP Trunk - SIP/2.0 484 Address Incomplete on incoming call

Answered Question
Aug 13th, 2014
User Badges:

Hi,

 

i am working on a customer site on cisco telephony. It is having SIP trunk with 100 DIDs. Outgoing call from site to the outside is working fine with no issue. But for incoming when i dial from my Cell phone to the company; cisco phone rings and when pick the phone no voice pass between and Cell phone play the continous ring even i pick the cisco phone for answer.

 

There are some debugs which are provided to me by ITSP having SIP/2.0 484 Address Incomplete issue.

 

================================================================================

[No.           ] 1
[TimeStamp     ] 2014-08-13 14:49:44
[Direction     ] RECV
[Msg Name      ] INVITE
[Module No     ] 194
[Local Address ] 10.200.0.8:5060
[Remote Address] 10.208.16.8:5063
[Hex Msg       ] 49 4E 56 49 54 45 20 73 69 70 3A 30 31 31 32 38 33 35 34 32 ...


INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.208.16.8:5063;branch=z9hG4bKehht5mgoypcpmm5hyyec1ysyp
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=m1k3mhck-CC-45
To: <sip:[email protected];user=phone>
CSeq: 1 INVITE
Contact: <sip:[email protected]:5063;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R010
Supported: 100rel
Max-Forwards: 70
Content-Length: 611
Content-Type: multipart/mixed;boundary=ssboundary

--ssboundary
Content-Length: 382
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 48206088 48206088 IN IP4 10.208.16.7
s=Sip Call
c=IN IP4 10.209.4.2
t=0 0
m=audio 32228 RTP/AVP 8 0 18 4 2 98 99 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

--ssboundary
Content-Length: 56
Content-Type: application/isup;version=itu-t92+

H
!8E2
TY)1X9x1:?:Cf
--ssboundary--

 

================================================================================

[No.           ] 2
[TimeStamp     ] 2014-08-13 14:49:43
[Direction     ] SEND
[Msg Name      ] INVITE
[Module No     ] 202
[Local Address ] 10.200.0.7:5069
[Remote Address] 10.200.20.235:5060
[Hex Msg       ] 49 4E 56 49 54 45 20 73 69 70 3A 30 31 31 32 38 33 35 34 32 ...


INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.200.0.7:5069;branch=z9hG4bKhk4u7ueo77tbep7of7opcck4s
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=hducfsbk-CC-36-TRC-703
To: <sip:[email protected];user=phone>
CSeq: 1 INVITE
Max-Forwards: 69
Contact: <sip:[email protected]:5069;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R010
Supported: 100rel
Content-Length: 381
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 27071045 27071045 IN IP4 10.200.0.7
s=Sip Call
c=IN IP4 10.209.4.2
t=0 0
m=audio 32228 RTP/AVP 8 0 18 4 2 98 99 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

 

================================================================================

[No.           ] 3
[TimeStamp     ] 2014-08-13 14:49:43
[Direction     ] RECV
[Msg Name      ] 100
[Module No     ] 202
[Local Address ] 10.200.0.7:5069
[Remote Address] 10.200.20.235:5060
[Hex Msg       ] 53 49 50 2F 32 2E 30 20 31 30 30 20 54 72 79 69 6E 67 0D 0A ...


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.200.0.7:5069;branch=z9hG4bKhk4u7ueo77tbep7of7opcck4s
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=hducfsbk-CC-36-TRC-703
To: <sip:[email protected];user=phone>
CSeq: 1 INVITE
Content-Length: 0

 


================================================================================

[No.           ] 4
[TimeStamp     ] 2014-08-13 14:49:44
[Direction     ] SEND
[Msg Name      ] 100
[Module No     ] 194
[Local Address ] 10.200.0.8:5060
[Remote Address] 10.208.16.8:5063
[Hex Msg       ] 53 49 50 2F 32 2E 30 20 31 30 30 20 54 72 79 69 6E 67 0D 0A ...


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.208.16.8:5063;branch=z9hG4bKehht5mgoypcpmm5hyyec1ysyp
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=m1k3mhck-CC-45
To: <sip:[email protected];user=phone>
CSeq: 1 INVITE
Content-Length: 0

 


================================================================================

[No.           ] 5
[TimeStamp     ] 2014-08-13 14:49:43
[Direction     ] RECV
[Msg Name      ] 484
[Module No     ] 202
[Local Address ] 10.200.0.7:5069
[Remote Address] 10.200.20.235:5060
[Hex Msg       ] 53 49 50 2F 32 2E 30 20 34 38 34 20 41 64 64 72 65 73 73 20 ...


SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.200.0.7:5069;branch=z9hG4bKhk4u7ueo77tbep7of7opcck4s
Record-Route: <sip:10.200.20.235:5060;transport=udp;lr>
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=hducfsbk-CC-36-TRC-703
To: <sip:[email protected];user=phone>;tag=sbc080787C44FE0-1AC9
CSeq: 1 INVITE
Date: Wed, 13 Aug 2014 11:49:44 GMT
Allow-Events: kpml,telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=28
Content-Length: 0

 


================================================================================

[No.           ] 6
[TimeStamp     ] 2014-08-13 14:49:43
[Direction     ] SEND
[Msg Name      ] ACK
[Module No     ] 202
[Local Address ] 10.200.0.7:5069
[Remote Address] 10.200.20.235:5060
[Hex Msg       ] 41 43 4B 20 73 69 70 3A 30 31 31 32 38 33 35 34 32 33 40 31 ...


ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.200.0.7:5069;branch=z9hG4bKhk4u7ueo77tbep7of7opcck4s
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=hducfsbk-CC-36-TRC-703
To: <sip:[email protected];user=phone>;tag=sbc080787C44FE0-1AC9
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

 


================================================================================

[No.           ] 7
[TimeStamp     ] 2014-08-13 14:49:44
[Direction     ] SEND
[Msg Name      ] 484
[Module No     ] 194
[Local Address ] 10.200.0.8:5060
[Remote Address] 10.208.16.8:5063
[Hex Msg       ] 53 49 50 2F 32 2E 30 20 34 38 34 20 41 64 64 72 65 73 73 20 ...


SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.208.16.8:5063;branch=z9hG4bKehht5mgoypcpmm5hyyec1ysyp
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=m1k3mhck-CC-45
To: <sip:[email protected];user=phone>;tag=hh2ketup
CSeq: 1 INVITE
Reason: Q.850;cause=28;text="address incomplete"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 6
Content-Type: application/isup;version=itu-t92+

 


================================================================================

[No.           ] 8
[TimeStamp     ] 2014-08-13 14:49:44
[Direction     ] RECV
[Msg Name      ] ACK
[Module No     ] 194
[Local Address ] 10.200.0.8:5060
[Remote Address] 10.208.16.8:5063
[Hex Msg       ] 41 43 4B 20 73 69 70 3A 30 31 31 32 38 33 35 34 32 33 40 31 ...


ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.208.16.8:5063;branch=z9hG4bKehht5mgoypcpmm5hyyec1ysyp
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=m1k3mhck-CC-45
To: <sip:[email protected];user=phone>;tag=hh2ketup
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

 

 

 

 

=======================================

Here is my configuration of the cisco Gateway

 


voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 no supplementary-service h450.2
 no supplementary-service h450.3
 no supplementary-service h225-notify cid-update
 redirect ip2ip
 fax protocol pass-through g711alaw
 h323
 sip
  rel1xx disable
  header-passing
  registrar server expires max 3600 min 3500
  transport switch udp tcp
  redirect contact order best-match
  midcall-signaling passthru
!


voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  registrar server

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

!
voice translation-rule 1
 rule 1 /^11/ /0/
 rule 2 /^2/ /02/
 rule 3 /^3/ /03/
 rule 4 /^4/ /04/
 rule 5 /^5/ /005/
 rule 6 /^6/ /06/
 rule 7 /^7/ /07/
 rule 8 /^1/ /001/
!
voice translation-rule 2
 rule 1 /^0\(\)/ /\1/
!
voice translation-rule 10
 rule 1 /^11/ /0/
 rule 2 /^12/ /0012/
 rule 3 /^13/ /0013/
 rule 4 /^14/ /0014/
 rule 5 /^5/ /005/
 rule 6 /^16/ /0016/
 rule 7 /^17/ /0017/
 rule 8 /^18/ /0018/
!
voice translation-rule 20
 rule 1 /^54..$/ /283\0/
!
voice translation-rule 120
 rule 1 /^89..$/ /243\0/
!
!
voice translation-profile INCO_SIP
 translate calling 10
!
voice translation-profile PSTN-IN
 translate calling 1
!
voice translation-profile PSTN-OUT
 translate called 2
!
voice translation-profile SIP
 translate calling 20
 translate called 10
!
voice translation-profile SIP2
 translate calling 120
 translate called 10

 

 

 

dial-peer voice 14 voip
 description Outgoing
 translation-profile outgoing SIP
 destination-pattern .T
 rtp payload-type cisco-codec-fax-ack 111
 rtp payload-type nte 97
 session protocol sipv2
 session target ipv4:10.200.7.157:5060
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!
dial-peer voice 17 voip
 description incoming From STC Server to CUCM
 preference 1
 destination-pattern ^28354..$
 session target ipv4:172.16.200.13
 voice-class codec 1
 dtmf-relay h245-alphanumeric
 no vad
!
dial-peer voice 18 voip
 description incoming From STC Server to CUCM 
 destination-pattern ^28354..$
 session target ipv4:172.16.200.20
 voice-class codec 1
 dtmf-relay h245-alphanumeric 
 no vad


dial-peer voice 10 voip
 translation-profile incoming INCO_SIP
 session protocol sipv2
 session target sip-server
 incoming called-number ^28354..$
 dtmf-relay rtp-nte
 codec g711alaw
 no vad

 

 

 

 

 

 

 

 

Correct Answer by Ayodeji Okanlawon about 3 years 3 days ago

Hi,

I have looked at your logs and here is what is going on...

This is the first INVITE that ITSP sent to you...

Received:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKcb2o4f4u2soukudt7u4o2che7T37049
Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=sbc0804dpubefak-CC-41

+++Next the gateway sends an INVITE to CUCM+++

Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.200.245:5060;branch=z9hG4bK287A719DB
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=no;privacy=off
From: <sip:[email protected]>;tag=8D02D420-F66
To: <sip:[email protected]>Date: Thu, 14 Aug 2014 12:16:09 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2614503111-0585306596-2704712946-2244811493
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITETimestamp: 1408018569

+++Next we see CUCM sent the following+++

Received:
------------------------------------------------------------------------
SIP/2.0 503 Service UnavailableVia: SIP/2.0/UDP
-------------------------------------------------------------------------
172.16.200.245:5060;branch=z9hG4bK287A719DB
From: <sip:[email protected]>;tag=8D02D420-F66
To: <sip:[email protected]>;tag=371226223
Date: Thu, 14 Aug 2014 12:16:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITEAllow-Events: presence
Warning: 399 RYDHO-CMS01 "Unable to find a device handler for the request received on port 52458 from 172.16.200.245"Content-Length: 0
046071: Aug 14 12:16:09.558: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

+++Next thing is that CUCM sent another invite to 172.16.200.13 which is the secondary cucm.+++

Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.200.245:5060;branch=z9hG4bK287A811

ANd we get the same response from CUCM

Received:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.16.200.245:5060;branch=z9hG4bK287A8117
FFrom: <sip:[email protected]>;tag=8D02D42C-2687
To: <sip:[email protected]>;tag=1598895316Date: Thu, 14 Aug 2014 12:16:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITEAllow-Events: presence
Warning: 399 RYDHO-CMP01 "Unable to find a device handler for the request received on port 65245 from 172.16.200.245"Content-Length: 0
046075: Aug 14 12:16:09.566: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

The next thing we see is that the gateway sends an INVITE back out to your ITSP (this is happening because you have a .T in the dial-peer)

Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.66.7.126:5060;branch=z9hG4bK287A97E

And then your ITSP sends back the 484 address incomplete because this number shouldnt be going to them as they dont have it ont heir system.

The reason why CUCM is sending you a 503 service unavailable with a 399 warning is because the ip address that is been used for signalling 172.16.200.245 is not the one configured.on the sip trunk in cucm.

I looked at your config and I observed that you do not have any sip binding configured, hence the gateway is using the closest interface to the destination to route the call. To resolve this, do one of the ff:

1. Change the ip address on your sip trunk to 172.16.200.245

2. On your dial-peers to cucm configure bind commands to use the interface that has the ip address you have on cucm

voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0

chnage gig0/0 to the correct interface.

Then on the dial-peer facing your ITSP do a similar thing..

voice-class sip bind control source-interface FastEthernet0/3/0
 voice-class sip bind media source-interface FastEthernet0/3/0
 

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Overall Rating: 5 (2 ratings)
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islam.kamal Wed, 08/13/2014 - 21:57
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Hello

Can you update you dial-peers for incoming to the below:-

dial-peer voice 18 voip
 description incoming From STC Server to CUCM 
 destination-pattern ^28354..$
session protocol sipv2
 session target ipv4:172.16.200.20
 voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 no vad

 

dial-peer voice 17 voip
 description incoming From STC Server to CUCM
 preference 1
 destination-pattern ^28354..$
 session target ipv4:172.16.200.13
session protocol sipv2
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 no vad

no dial-peer voice 10 voip

On cucm - Device -Trunk - SIP trunk "which already configured" under inbound calls - on Significant Digits " select the number of digits for your internal extensions " this will be more enough



Thanks

please rate all useful information
 

mohsin majeed Wed, 08/13/2014 - 22:31
User Badges:

Dear Salam thanks for your prompt reply

 

I already did this, but i think you missing incoming called number in the dial peers you provided or no need. If i delete dial-peer 10 nothing happens even bell doesn't come on the cisco phone. i also tried the below configuration but nothing happens even this was working for another customer. May b in the next post i will share some snapshots

 

voice translation-rule 51
 rule 1 /^9\(\)/ /\1/
!
voice translation-rule 52
 rule 1 /^2835/ /5/
 rule 2 /^12835/ /5/
 rule 2 /^112835/ /5/
 rule 3 /^0112835/ /5/
 
 
!
voice translation-rule 53
 rule 1 /^5/ /905/
 rule 2 /^1/ /901/
 rule 3 /^2/ /902/
 rule 4 /^3/ /903/
 rule 5 /^4/ /904/
 rule 6 /^6/ /906/
 rule 7 /^7/ /907/
 rule 8 /^8/ /908/
 rule 10 /^00/ /900/
 rule 11 /'+'/ /900/
!
!
voice translation-profile OUT
 translate called 51
!
voice translation-profile REDIAL
 translate calling 53
!
voice translation-profile SIP-NEW
 translate called 52
!

 


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

 

dial-peer voice 802 voip
 description ** SIP TO STC **
 translation-profile outgoing OUT
 destination-pattern .T
 session protocol sipv2
 session target ipv4:10.200.7.157:5060
 session transport udp
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay sip-notify rtp-nte sip-kpml
 codec g711alaw
 


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

dial-peer voice 201 voip
 description ** SIP INCOMING FROM STC **
 translation-profile incoming SIP-NEW
 translation-profile outgoing REDIAL
 destination-pattern 5...
 session protocol sipv2
 session target ipv4:172.16.200.20
 incoming called-number 2835...$
 dtmf-relay sip-notify rtp-nte sip-kpml
 codec g711alaw
!
dial-peer voice 203 voip
 description ** SIP INCOMING FROM STC **
 translation-profile incoming SIP-NEW
 translation-profile outgoing REDIAL
 destination-pattern 5...
 session protocol sipv2
 session target ipv4:172.16.200.20
 incoming called-number 12835...$
 dtmf-relay sip-notify rtp-nte sip-kpml
 codec g711alaw
!
dial-peer voice 203 voip
 description ** SIP INCOMING FROM STC **
 translation-profile incoming SIP-NEW
 translation-profile outgoing REDIAL
 destination-pattern 5...
 session protocol sipv2
 session target ipv4:172.16.200.20
 incoming called-number 112835...$
 dtmf-relay sip-notify rtp-nte sip-kpml
 codec g711alaw

dial-peer voice 204 voip
 description ** SIP INCOMING FROM STC **
 translation-profile incoming SIP-NEW
 translation-profile outgoing REDIAL
 destination-pattern 5...
 session protocol sipv2
 session target ipv4:172.16.200.20
 incoming called-number 0112835...$
 dtmf-relay sip-notify rtp-nte sip-kpml
 codec g711alaw
!-----------------------
dial-peer voice 301 voip
 description ** SIP INCOMING FROM STC **
 translation-profile incoming SIP-NEW
 translation-profile outgoing REDIAL
 destination-pattern 5...
 session protocol sipv2
 session target ipv4:172.16.200.20
 preference 1
 incoming called-number 2835...$
 dtmf-relay sip-notify rtp-nte sip-kpml
 codec g711alaw
!
dial-peer voice 302 voip
 description ** SIP INCOMING FROM STC **
 translation-profile incoming SIP-NEW
 translation-profile outgoing REDIAL
 destination-pattern 5...
 session protocol sipv2
 session target ipv4:172.16.200.20
 Preference 1
 incoming called-number 12835...$
 dtmf-relay sip-notify rtp-nte sip-kpml
 codec g711alaw
!
dial-peer voice 303 voip
 description ** SIP INCOMING FROM STC **
 translation-profile incoming SIP-NEW
 translation-profile outgoing REDIAL
 destination-pattern 5...
 session protocol sipv2
 session target ipv4:172.16.200.20
 incoming called-number 112835...$
 Preference 1
 dtmf-relay sip-notify rtp-nte sip-kpml
 codec g711alaw

dial-peer voice 304 voip
 description ** SIP INCOMING FROM STC **
 translation-profile incoming SIP-NEW
 translation-profile outgoing REDIAL
 destination-pattern 5...
 session protocol sipv2
 session target ipv4:172.16.200.20
 incoming called-number 0112835...$
 Preference 1
 dtmf-relay sip-notify rtp-nte sip-kpml
 codec g711alaw

 

Your feedback

islam.kamal Wed, 08/13/2014 - 22:45
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Hello

 

Yes , i forget . Please add the below for my dial peers which i shared before , then test and tell me what did you hear?.

translation-profile incoming 
 incoming called-number .

I need to get a brief which dialed number from outside , and what is the range for your internal extensions .

Thanks

please rate all useful infromation

mohsin majeed Thu, 08/14/2014 - 01:36
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Dear,

 

I will visit the site and apply the configuration you provided.

Rest of the information is that 2835400 - 2835499 and i am dialing 0112835423 from my cell phone and phone rings. the inter range is 54.. to so on. But here External phone number mask is being used like 283XXXX.

 

I am afraid here SIP trunk and E1 trunks are terminating on the same gateway and E1 range is from 4949... 

is there any possibility that incoming called-number . is overlapping with SIP and E1

Appreciated if you can share you cell number and i can give you team viewer access.

islam.kamal Thu, 08/14/2014 - 02:53
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Hi

There is no overlaping for sure , just try and everyting will be ok for sure . If you failed to do incoming calls , i will be so appreciated if you catch all incoming calls logs if it fails.

My friend there is no confilct ,and the call is reach to your CUCM . Now there are two ways , apply my config if no work . You have to apply "MTP , transcode " configuration media resources which can hep , because if you see on above incoming calls use g729.

 

Thanks

please rate all useful infrormation

mohsin majeed Thu, 08/14/2014 - 07:33
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Dear,

 

I would like to discuss the full scenario now. There is Gateway having E1 and SIP trunk on it. SIP trunk has two ranges 28354.. and 24389..  the behaviour of both ranges is the same. if i able to solve for one the 2nd will automaticlly solved.

 

i am attaching the full notepad file with Running config and traces. Just consider the traces which having 582210218 number.

i highlighted the errors in lines in the attached file.

SIP/2.0 503 Service UnavailableVia: SIP/2.0/UDP 

SIP/2.0 484 Address IncompleteVia: SIP/2.0/UDP

Reason: Q.850;cause=28Content-Length: 0

 

. I know there is some extra old configuration in the router but ignore i will remove it later.

According to your suggestion i applied the dial peers there is no ring at all. Before it was there with old dial peers which were dial-peer 10 and 18.

Anyway, i set up a SIP trunk on cucm to ITSP and outgoing call no probelm. For incoming i selected significant digits 4 as there are 4 digit extensions. One thing is that External phone number mask is also being used 283XXXX in the line of the phone.

Behavior is that once i dial from my cell phone it shows number invalid on my cell phone.

I tried destination pattern with ^2835...$     2835...    28354..   5...  after applying translation rule SIP-New     and    54..    means every possibility that i could apply.

I deleted the old dial peers for incoming 17,18 and 10. because i don't want to go with old configuration.

Transcoder is configured already you can see in the file and registered with cucm.

 

please let me know if you need any other information. Situation is crutial :)

 

 

Your feedback plz

 

 

Attachment: 
Correct Answer
Ayodeji Okanlawon Thu, 08/14/2014 - 08:41
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    2017 IP Telephony

Hi,

I have looked at your logs and here is what is going on...

This is the first INVITE that ITSP sent to you...

Received:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKcb2o4f4u2soukudt7u4o2che7T37049
Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=sbc0804dpubefak-CC-41

+++Next the gateway sends an INVITE to CUCM+++

Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.200.245:5060;branch=z9hG4bK287A719DB
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=no;privacy=off
From: <sip:[email protected]>;tag=8D02D420-F66
To: <sip:[email protected]>Date: Thu, 14 Aug 2014 12:16:09 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2614503111-0585306596-2704712946-2244811493
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITETimestamp: 1408018569

+++Next we see CUCM sent the following+++

Received:
------------------------------------------------------------------------
SIP/2.0 503 Service UnavailableVia: SIP/2.0/UDP
-------------------------------------------------------------------------
172.16.200.245:5060;branch=z9hG4bK287A719DB
From: <sip:[email protected]>;tag=8D02D420-F66
To: <sip:[email protected]>;tag=371226223
Date: Thu, 14 Aug 2014 12:16:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITEAllow-Events: presence
Warning: 399 RYDHO-CMS01 "Unable to find a device handler for the request received on port 52458 from 172.16.200.245"Content-Length: 0
046071: Aug 14 12:16:09.558: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

+++Next thing is that CUCM sent another invite to 172.16.200.13 which is the secondary cucm.+++

Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.200.245:5060;branch=z9hG4bK287A811

ANd we get the same response from CUCM

Received:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.16.200.245:5060;branch=z9hG4bK287A8117
FFrom: <sip:[email protected]>;tag=8D02D42C-2687
To: <sip:[email protected]>;tag=1598895316Date: Thu, 14 Aug 2014 12:16:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITEAllow-Events: presence
Warning: 399 RYDHO-CMP01 "Unable to find a device handler for the request received on port 65245 from 172.16.200.245"Content-Length: 0
046075: Aug 14 12:16:09.566: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

The next thing we see is that the gateway sends an INVITE back out to your ITSP (this is happening because you have a .T in the dial-peer)

Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.66.7.126:5060;branch=z9hG4bK287A97E

And then your ITSP sends back the 484 address incomplete because this number shouldnt be going to them as they dont have it ont heir system.

The reason why CUCM is sending you a 503 service unavailable with a 399 warning is because the ip address that is been used for signalling 172.16.200.245 is not the one configured.on the sip trunk in cucm.

I looked at your config and I observed that you do not have any sip binding configured, hence the gateway is using the closest interface to the destination to route the call. To resolve this, do one of the ff:

1. Change the ip address on your sip trunk to 172.16.200.245

2. On your dial-peers to cucm configure bind commands to use the interface that has the ip address you have on cucm

voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0

chnage gig0/0 to the correct interface.

Then on the dial-peer facing your ITSP do a similar thing..

voice-class sip bind control source-interface FastEthernet0/3/0
 voice-class sip bind media source-interface FastEthernet0/3/0
 

mohsin majeed Fri, 08/15/2014 - 13:31
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Thank you both Ayodeji and Islam,

 

Mr Ayodeji,

The anwser seems to be very understandable. As there are 2 holidays. Once i will visit the customer site i will apply the configuration and will post the end result again with traces.

 

Thanks again for your efforts uptil now.

mohsin majeed Sun, 08/17/2014 - 06:03
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First of All Love you dear Ayodegi and you Awesome

 

What i all did changed the sip trunk ip to 172.16.200.245 in the cucm only and life is pretty good which was before 10.1.0.4.

Now i am thinking why i didn't think about it. After solving this issue life seems very interesting. Actually this was the only problem preventing us to close this project (not much expert ;)). I was working with ITSP asking them issue is from your side; opening tickets with them, searching blogs, using google; all i did.

Its my second time i discussed issue here. It is the best place to get the solution.

 

According to me you closed this project :)

 

Thanks again and stay happy

islam.kamal Thu, 08/14/2014 - 11:48
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Hello

+100000000000000 to Mr/Ayodeji.

Kindly add bind interfaces "which connected to STC back to back" under dial-peers as Mr/Ayodj .

Thanks

please rate all useful infromation

mohsin majeed Sun, 08/17/2014 - 06:14
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Thanks Mr. Islam for your efforts also. Issue is solved by just changing SIP trunk ip from 10.1.0.4 to 172.16.200.245 in the cucm as Ayodeji said nothing else.

 

Take care

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