Strange issue with incoming call from certain countrys

clopezg78 · ‎01-25-2012

Hello guys!

I have a cluster with CUCM 8.0.3.20000-2 8.0 . in Spain .

The topology is very simple :

TELCO "Is an ACME server" --sip------------------ 2800 router with (C2800NM-ADVIPSERVICESK9-M), Version 12.4(24)T2------------------- CUCM -Cluster

The problem is regarding incoming calls from Uruguay,Brazil. When the call enter the 2800 and forward to CUCM the G.729 codec under the

Content-Type: application/sdp is no send .When the phone hook-off still ringing and the final error on CUBE is :

Received:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 172.26.254.30:5060;branch=z9hG4bK24613FCC

From: <sip:50765500505@172.26.254.30>;tag=7755AAB4-2325

To: <sip:915786322@172.26.254.41>;tag=2c6ead86-57b1-4553-8bec-f27e30084b15-35003492

Date: Mon, 23 Jan 2012 16:01:20 GMT

Call-ID: 26BE30D-451411E1-8C32A158-B79B2307@172.26.254.30

CSeq: 101 INVITE

Allow-Events: presence

Reason: Q.850;cause=47

Content-Length: 0

If we change the codec to G711 works well.

Bellow the configuration and a bad and good calls "traces"

Any ideas guys= :d

The configuration of 2800 is :

Current configuration : 12725 bytes

!

! Last configuration change at 11:14:34 CET Thu Dec 22 2011 by EKT

! NVRAM config last updated at 12:29:40 CET Thu Dec 22 2011 by EKT

!

version 12.4

service tcp-keepalives-in

service tcp-keepalives-out

service timestamps debug datetime msec localtime

service timestamps log datetime msec localtime

service password-encryption

service sequence-numbers

!

hostname CUBE_BM_IBA

!

boot-start-marker

boot-end-marker

!

no ip bootp server

no ip domain lookup

--More-- ip domain name KUTXA.VITAL

no ipv6 cef

!

multilink bundle-name authenticated

!

voice service voip

address-hiding

allow-connections sip to sip

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw

sip

rel1xx disable

min-se 90

asserted-id pai

g729 annexb-all

sip-profiles 1

!

--More-- !

voice class codec 1

codec preference 1 g729r8

codec preference 2 g711alaw

!

voice class codec 2

codec preference 1 g711alaw

codec preference 2 g729r8

!

voice class sip-profiles 1

request ANY sip-header Allow-Header modify ", UPDATE" ""

response ANY sip-header Allow-Header modify ", UPDATE" ""

!

--More-- !

!

voice iec syslog

!

voice source-group DE_TRUNK_FAX_SIP

access-list 21

translation-profile incoming DE_TRUNK_FAX_SIP

!

voice source-group DE_TRUNK_VOZ_SIP

access-list 11

translation-profile incoming DE_TRUNK_VOZ_SIP

!

voice translation-rule 11

rule 1 /^/ /8/

!

voice translation-rule 21

rule 1 /^/ /5/

!

voice translation-rule 111

rule 1 /^0/ //

!

voice translation-rule 112

rule 1 /^8/ //

--More-- !

voice translation-rule 113

rule 1 /9.*/ /943504458/

!

voice translation-rule 114

rule 1 /9.*/ /943504463/

!

voice translation-rule 115

rule 1 /^[23]....$/ /943504458/

!

voice translation-rule 116

rule 1 /^2....$/ /943504463/

!

voice translation-rule 121

rule 1 /^4/ //

!

voice translation-rule 122

rule 1 /^5/ //

!

voice translation-profile A_CUCM_FAX

translate called 122

!

--More-- voice translation-profile A_CUCM_VOZ

translate called 112

!

voice translation-profile A_TRUNK_FAX_SIP

translate called 121

!

voice translation-profile A_TRUNK_FAX_SIP_PRUEBAS

translate calling 114

translate called 121

translate redirect-called 116

!

voice translation-profile A_TRUNK_VOZ_SIP

translate called 111

translate redirect-called 115

!

voice translation-profile A_TRUNK_VOZ_SIP_PRUEBAS

translate calling 113

translate called 111

translate redirect-called 115

!

voice translation-profile DE_TRUNK_FAX_SIP

translate called 21

!

--More-- voice translation-profile DE_TRUNK_VOZ_SIP

translate called 11

!

voice-card 0

!

archive

log config

logging enable

logging size 800

notify syslog contenttype plaintext

hidekeys

path flash:archived-config-

maximum 14

--More-- write-memory

!

ip tftp source-interface Loopback1

ip ssh maxstartups 10

ip ssh time-out 30

ip ssh source-interface Loopback1

!

interface Loopback1

description IP DE GESTION

ip address 10.1.2.64 255.255.255.255

ip flow ingress

ip flow egress

!

interface FastEthernet0/0

description CONEXION LAN VOZ (CUCM y demas)

ip address 172.26.254.30 255.255.255.0

--More-- ip access-group 169 in

ip access-group 167 out

ip flow ingress

ip flow egress

duplex auto

speed auto

!

interface FastEthernet0/1

description SIP TRUNK HACIA EUSKALTEL

no ip address

duplex auto

speed auto

!

interface FastEthernet0/1.3998

description VPLS SIP TRUNK VOZ

encapsulation dot1Q 3998

ip address 172.31.252.4 255.255.255.240

!

interface FastEthernet0/1.3999

description VPLS SIP TRUNK FAX

encapsulation dot1Q 3999

ip address 172.31.251.4 255.255.255.240

ip access-group 166 in

--More-- ip access-group 168 out

no cdp enable

!

router ospf 3

log-adjacency-changes

no auto-cost

network 10.1.2.0 0.0.0.255 area 0

network 172.26.254.0 0.0.0.255 area 0

!

ip forward-protocol nd

ip route 10.1.2.65 255.255.255.255 172.31.252.5 254

ip route 172.26.182.101 255.255.255.255 172.26.254.253

no ip http server

no ip http secure-server

!

ip flow-export source Loopback1

ip flow-export destination 172.26.101.160 9996

!

ip radius source-interface Loopback1

logging 172.26.101.160

access-list 11 remark DE_TRUNK_VOZ

access-list 11 permit 172.31.252.1

--More-- access-list 21 remark DE_TRUNK_FAX

access-list 21 permit 172.31.251.1

access-list 166 permit ip any host 10.27.66.2

access-list 166 permit ip any any

access-list 167 permit ip any host 10.27.66.2

access-list 167 permit ip any any

access-list 168 permit ip host 10.27.66.2 any

access-list 168 permit ip any any

access-list 169 permit ip host 10.27.66.2 any

access-list 169 permit ip any any

no cdp run

!

snmp-server community cgrc1999 RW

snmp-server community public RO

snmp-server community ServKutxa RO 53

snmp-server community kut89208 RW 54

snmp-server host 172.26.99.14 ServKutxa

radius-server attribute 44 include-in-access-req

--More-- radius-server domain-stripping

radius-server vsa send authentication

!

control-plane

!

call fallback active

!

no mgcp package-capability res-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

mgcp fax t38 ecm

!

sccp local FastEthernet0/0

sccp ccm 172.26.254.40 identifier 2 version 7.0

sccp ccm 172.26.254.41 identifier 1 version 7.0

sccp

!

sccp ccm group 1

bind interface FastEthernet0/0

associate ccm 1 priority 1

associate ccm 2 priority 2

--More-- !

!

dial-peer voice 1 voip

preference 1

service session

rtp payload-type nse 99

session protocol sipv2

incoming called-number 9T

dtmf-relay rtp-nte

fax rate 14400

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw

no vad

!

dial-peer voice 2 voip

preference 1

service session

rtp payload-type nse 99

rtp payload-type nte 100

session protocol sipv2

incoming called-number 0T

dtmf-relay rtp-nte

fax rate 14400

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw

--More-- no vad

!

dial-peer voice 3 voip

preference 1

service session

rtp payload-type nse 99

rtp payload-type nte 100

voice-class codec 2

session protocol sipv2

incoming called-number 4T

dtmf-relay rtp-nte

fax rate 14400

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw

no vad

!

dial-peer voice 11 voip

description WAN-VOZ-ACME

translation-profile outgoing A_TRUNK_VOZ_SIP

preference 1

service session

destination-pattern 0T

voice-class codec 1

session protocol sipv2

--More-- session target ipv4:172.31.252.1

session transport udp

dtmf-relay rtp-nte

no vad

!

dial-peer voice 21 voip

description WAN-FAX-ACME

translation-profile outgoing A_TRUNK_FAX_SIP

preference 1

service session

destination-pattern 4T

voice-class codec 2

session protocol sipv2

session target ipv4:172.31.251.1

session transport udp

dtmf-relay rtp-nte

fax rate 14400

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw

no vad

!

dial-peer voice 101 voip

description VOZ_CUCM_PPAL

translation-profile outgoing A_CUCM_VOZ

--More-- preference 2

service session

destination-pattern 8T

voice-class codec 1

session protocol sipv2

session target ipv4:172.26.254.41

session transport udp

dtmf-relay rtp-nte

no vad

!

dial-peer voice 102 voip

description VOZ_CUCM_BKP

translation-profile outgoing A_CUCM_VOZ

preference 3

service session

destination-pattern 8T

voice-class codec 1

session protocol sipv2

session target ipv4:172.26.254.40

session transport udp

dtmf-relay rtp-nte

no vad

!

--More-- dial-peer voice 121 voip

description FAX_CUCM_PPAL

translation-profile outgoing A_CUCM_FAX

preference 2

service session

destination-pattern 5T

session protocol sipv2

session target ipv4:172.26.254.41:6060

session transport udp

dtmf-relay rtp-nte

trunk-group-label source DE_FAX

codec g711alaw

fax rate 14400

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw

no vad

!

dial-peer voice 122 voip

description FAX_CUCM_BKP

translation-profile outgoing A_CUCM_FAX

preference 3

service session

destination-pattern 5T

voice-class codec 2

--More-- session protocol sipv2

session target ipv4:172.26.254.40:6060

session transport udp

dtmf-relay rtp-nte

trunk-group-label source DE_FAX

fax rate 14400

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw

no vad

!

dial-peer voice 12 voip

description WAN-VOZ-ACME

translation-profile outgoing A_TRUNK_VOZ_SIP_PRUEBAS

preference 1

service session

destination-pattern 0943008728

voice-class codec 1

session protocol sipv2

session target ipv4:172.31.252.1

session transport udp

dtmf-relay rtp-nte

no vad

!

dial-peer voice 22 voip

--More-- description WAN-FAX-ACME

translation-profile outgoing A_TRUNK_FAX_SIP_PRUEBAS

preference 1

service session

destination-pattern 4945162118

voice-class codec 2

session protocol sipv2

session target ipv4:172.31.251.1

session transport udp

dtmf-relay rtp-nte

fax rate 14400

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw

no vad

!

dial-peer voice 201 voip

description prueba_caso2433581

translation-profile outgoing A_CUCM_VOZ

preference 2

service session

answer-address +582127450447

destination-pattern 8T

voice-class codec 1

session protocol sipv2

--More-- session target ipv4:172.26.254.41

session transport udp

dtmf-relay rtp-nte

no vad

!

dial-peer voice 202 voip

description prueba_caso2433581

translation-profile outgoing A_CUCM_VOZ

preference 3

service session

answer-address +582127450447

destination-pattern 8T

voice-class codec 1

session protocol sipv2

session target ipv4:172.26.254.40

session transport tcp

dtmf-relay rtp-nte

no vad

!

sip-ua

g729-annexb override

!

The log of a bad call is :

Received:

INVITE sip:915786322@172.31.252.4:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 172.31.252.1:5060;branch=z9hG4bK6uqovh206o6h6n4ia6g0.1

From: <sip:50765500505@172.31.252.1:5060;user=phone>;tag=SDj7jb001-172.16.221.2651000+1+adae0002+7fa94043

To: <sip:915786322@172.31.252.4:5060;user=phone>

CSeq: 483705438 INVITE

Expires: 180

Min-SE: 90

Session-Expires: 1800

Supported: replaces, 100rel, timer

Content-Length: 291

Request-Disposition: fork, parallel

Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER, UPDATE

Call-ID: SDj7jb001-dcfdc333ba35543ea110591226d01dc1-c54qtk0

P-Asserted-Identity: <sip:50765500505@172.16.221.26;user=phone>

Privacy: none

Max-Forwards: 69

Contact: <sip:50765500505@172.31.252.1:5060;transport=udp>

Content-Type: application/sdp

v=0

o=- 3536323280 3536323280 IN IP4 172.31.252.1

s=-

c=IN IP4 172.31.252.1

t=0 0

m=audio 11226 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=maxptime:30

a=silenceSupp:on - - - -

SIP: (601844) Attribute mid, level 1 instance 1 not found.

an 23 17:13:24.015: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.31.252.1:5060;branch=z9hG4bK6uqovh206o6h6n4ia6g0.1

From: <sip:50765500505@172.31.252.1:5060;user=phone>;tag=SDj7jb001-172.16.221.2651000+1+adae0002+7fa94043

To: <sip:915786322@172.31.252.4:5060;user=phone>

Date: Mon, 23 Jan 2012 16:13:23 GMT

Call-ID: SDj7jb001-dcfdc333ba35543ea110591226d01dc1-c54qtk0

CSeq: 483705438 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

190037: .Jan 23 17:13:24.019: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:915786322@172.26.254.41:5060 SIP/2.0

Via: SIP/2.0/UDP 172.26.254.30:5060;branch=z9hG4bK24613FCC

From: <sip:50765500505@172.26.254.30>;tag=7755AAB4-2325

To: <sip:915786322@172.26.254.41>

Date: Mon, 23 Jan 2012 16:13:24 GMT

Call-ID: 26BE30D-451411E1-8C32A158-B79B2307@172.26.254.30

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE: 90

Cisco-Guid: 40384383-1158943201-2351735128-3080397575

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1327335204

Contact: <sip:50765500505@172.26.254.30:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 68

Session-Expires: 1800

P-Asserted-Identity: <sip:50765500505@172.26.254.30>

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 249

v=0

o=CiscoSystemsSIP-GW-UserAgent 5422 413 IN IP4 172.26.254.30

s=SIP Call

c=IN IP4 172.26.254.30

t=0 0

m=audio 18598 RTP/AVP 8 101

c=IN IP4 172.26.254.30

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

190038: .Jan 23 17:13:24.023: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.26.254.30:5060;branch=z9hG4bK24613FCC

From: <sip:50765500505@172.26.254.30>;tag=7755AAB4-2325

To: <sip:915786322@172.26.254.41>

Date: Mon, 23 Jan 2012 16:01:20 GMT

Call-ID: 26BE30D-451411E1-8C32A158-B79B2307@172.26.254.30

CSeq: 101 INVITE

Allow-Events: presence

Content-Length: 0

190039: .Jan 23 17:13:24.027: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 172.26.254.30:5060;branch=z9hG4bK24613FCC

From: <sip:50765500505@172.26.254.30>;tag=7755AAB4-2325

To: <sip:915786322@172.26.254.41>;tag=2c6ead86-57b1-4553-8bec-f27e30084b15-35003492

Date: Mon, 23 Jan 2012 16:01:20 GMT

Call-ID: 26BE30D-451411E1-8C32A158-B79B2307@172.26.254.30

CSeq: 101 INVITE

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

Allow-Events: presence

Contact: <sip:915786322@172.26.254.41:5060>

Supported: X-cisco-srtp-fallback

Supported: Geolocation

P-Asserted-Identity: "20948 PABLO LAPLANA" <sip:20948@172.26.254.41>

Content-Length: 0

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 172.31.252.1:5060;branch=z9hG4bK6uqovh206o6h6n4ia6g0.1

From: <sip:50765500505@172.31.252.1:5060;user=phone>;tag=SDj7jb001-172.16.221.2651000+1+adae0002+7fa94043

To: <sip:915786322@172.31.252.4:5060;user=phone>;tag=7755AAC8-CBD

Date: Mon, 23 Jan 2012 16:13:23 GMT

Call-ID: SDj7jb001-dcfdc333ba35543ea110591226d01dc1-c54qtk0

CSeq: 483705438 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: <sip:915786322@172.31.252.4>;party=called;screen=no;privacy=off

Contact: <sip:915786322@172.31.252.4:5060>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 172.26.254.30:5060;branch=z9hG4bK24613FCC

From: <sip:50765500505@172.26.254.30>;tag=7755AAB4-2325

To: <sip:915786322@172.26.254.41>;tag=2c6ead86-57b1-4553-8bec-f27e30084b15-35003492

Date: Mon, 23 Jan 2012 16:01:20 GMT

Call-ID: 26BE30D-451411E1-8C32A158-B79B2307@172.26.254.30

CSeq: 101 INVITE

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

Allow-Events: presence

Contact: <sip:915786322@172.26.254.41:5060>

Supported: X-cisco-srtp-fallback

Supported: Geolocation

P-Asserted-Identity: "20781 PALOMA DIEZ" <sip:20781@172.26.254.41>

Content-Length: 0

190051: .Jan 23 17:13:29.007: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 172.26.254.30:5060;branch=z9hG4bK24613FCC

From: <sip:50765500505@172.26.254.30>;tag=7755AAB4-2325

To: <sip:915786322@172.26.254.41>;tag=2c6ead86-57b1-4553-8bec-f27e30084b15-35003492

Date: Mon, 23 Jan 2012 16:01:20 GMT

Call-ID: 26BE30D-451411E1-8C32A158-B79B2307@172.26.254.30

CSeq: 101 INVITE

Allow-Events: presence

Reason: Q.850;cause=47

Content-Length: 0

ACK sip:915786322@172.26.254.41:5060 SIP/2.0

Via: SIP/2.0/UDP 172.26.254.30:5060;branch=z9hG4bK24613FCC

From: <sip:50765500505@172.26.254.30>;tag=7755AAB4-2325

To: <sip:915786322@172.26.254.41>;tag=2c6ead86-57b1-4553-8bec-f27e30084b15-35003492

Date: Mon, 23 Jan 2012 16:13:24 GMT

Call-ID: 26BE30D-451411E1-8C32A158-B79B2307@172.26.254.30

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

190056: .Jan 23 17:13:29.027: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:915786322@172.26.254.41:5060 SIP/2.0

Via: SIP/2.0/UDP 172.26.254.30:5060;branch=z9hG4bK2461521BF

From: <sip:50765500505@172.26.254.30>;tag=7755BE44-BAB

To: <sip:915786322@172.26.254.41>

Date: Mon, 23 Jan 2012 16:13:29 GMT

Call-ID: 5680BDD-451411E1-8C34A158-B79B2307@172.26.254.30

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE: 90

Cisco-Guid: 40384383-1158943201-2351735128-3080397575

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1327335209

Contact: <sip:50765500505@172.26.254.30:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 68

Session-Expires: 1800

P-Asserted-Identity: <sip:50765500505@172.26.254.30>

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 250

v=0

o=CiscoSystemsSIP-GW-UserAgent 8308 4413 IN IP4 172.26.254.30

s=SIP Call

c=IN IP4 172.26.254.30

t=0 0

m=audio 17756 RTP/AVP 8 101

c=IN IP4 172.26.254.30

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

190057: .Jan 23 17:13:29.031: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.26.254.30:5060;branch=z9hG4bK2461521BF

From: <sip:50765500505@172.26.254.30>;tag=7755BE44-BAB

To: <sip:915786322@172.26.254.41>

Date: Mon, 23 Jan 2012 16:01:25 GMT

Call-ID: 5680BDD-451411E1-8C34A158-B79B2307@172.26.254.30

CSeq: 101 INVITE

Allow-Events: presence

Content-Length: 0

190058: .Jan 23 17:13:29.035: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 172.26.254.30:5060;branch=z9hG4bK2461521BF

From: <sip:50765500505@172.26.254.30>;tag=7755BE44-BAB

To: <sip:915786322@172.26.254.41>;tag=2c6ead86-57b1-4553-8bec-f27e30084b15-35003502

Date: Mon, 23 Jan 2012 16:01:25 GMT

Call-ID: 5680BDD-451411E1-8C34A158-B79B2307@172.26.254.30

CSeq: 101 INVITE

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

Allow-Events: presence

Contact: <sip:915786322@172.26.254.41:5060>

Supported: X-cisco-srtp-fallback

Supported: Geolocation

P-Asserted-Identity: "20948 PABLO LAPLANA" <sip:20948@172.26.254.41>

Content-Length: 0

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 172.31.252.1:5060;branch=z9hG4bK6uqovh206o6h6n4ia6g0.1

From: <sip:50765500505@172.31.252.1:5060;user=phone>;tag=SDj7jb001-172.16.221.2651000+1+adae0002+7fa94043

To: <sip:915786322@172.31.252.4:5060;user=phone>;tag=7755AAC8-CBD

Date: Mon, 23 Jan 2012 16:13:23 GMT

Call-ID: SDj7jb001-dcfdc333ba35543ea110591226d01dc1-c54qtk0

CSeq: 483705438 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: <sip:915786322@172.31.252.4>;party=called;screen=no;privacy=off

Contact: <sip:915786322@172.31.252.4:5060>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

1

190078: .Jan 23 17:13:44.239: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 172.26.254.30:5060;branch=z9hG4bK2461521BF

From: <sip:50765500505@172.26.254.30>;tag=7755BE44-BAB

To: <sip:915786322@172.26.254.41>;tag=2c6ead86-57b1-4553-8bec-f27e30084b15-35003502

Date: Mon, 23 Jan 2012 16:01:25 GMT

Call-ID: 5680BDD-451411E1-8C34A158-B79B2307@172.26.254.30

CSeq: 101 INVITE

Allow-Events: presence

Reason: Q.850;cause=47

Content-Length: 0

The log from a call that process well :

eceived:

INVITE sip:943008712@172.31.255.4:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 172.31.255.1:5060;branch=z9hG4bK6s906530ao0hmmgbp210.1

From: <sip:50765500505@172.31.255.1:5060;user=phone>;tag=SDl7m5f01-172.16.221.2651001+1+8fd00003+69d9dfcc

To: <sip:943008712@172.31.255.4:5060;user=phone>

CSeq: 20038592 INVITE

Expires: 180

Min-SE: 90

Session-Expires: 1800

Supported: replaces, 100rel, timer

Content-Length: 291

Request-Disposition: fork, parallel

Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER, UPDATE

Call-ID: SDl7m5f01-13b7f302d8f21d7905bab1dd543bf962-c54qtk0

P-Asserted-Identity: <sip:50765500505@172.16.221.26;user=phone>

Privacy: none

Max-Forwards: 69

Contact: <sip:50765500505@172.31.255.1:5060;transport=udp>

Content-Type: application/sdp

v=0

o=- 3536319816 3536319816 IN IP4 172.31.255.1

s=-

c=IN IP4 172.31.255.1

t=0 0

m=audio 10638 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=maxptime:30

a=silenceSupp:on - - - -

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.31.255.1:5060;branch=z9hG4bK6s906530ao0hmmgbp210.1

From: <sip:50765500505@172.31.255.1:5060;user=phone>;tag=SDl7m5f01-172.16.221.2651001+1+8fd00003+69d9dfcc

To: <sip:943008712@172.31.255.4:5060;user=phone>

Date: Mon, 23 Jan 2012 15:16:47 GMT

Call-ID: SDl7m5f01-13b7f302d8f21d7905bab1dd543bf962-c54qtk0

CSeq: 20038592 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

.Jan 23 16:16:47.965: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:943008712@172.26.253.41:5060 SIP/2.0

Via: SIP/2.0/UDP 172.26.253.30:5060;branch=z9hG4bK23C54210FE

From: <sip:50765500505@172.26.253.30>;tag=EE04CC4C-1F39

To: <sip:943008712@172.26.253.41>

Date: Mon, 23 Jan 2012 15:16:47 GMT

Call-ID: 1A3859B3-450C11E1-8F7099E3-58230DFA@172.26.253.30

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE: 90

Cisco-Guid: 0439820490-1158418913-2406128099-1478692346

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1327331807

Contact: <sip:50765500505@172.26.253.30:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 68

Session-Expires: 1800

P-Asserted-Identity: <sip:50765500505@172.26.253.30>

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 282

v=0

o=CiscoSystemsSIP-GW-UserAgent 8083 0 IN IP4 172.26.253.30

s=SIP Call

c=IN IP4 172.26.253.30

t=0 0

m=audio 18180 RTP/AVP 8 18 101

c=IN IP4 172.26.253.30

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

.Jan 23 16:16:47.969: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.26.253.30:5060;branch=z9hG4bK23C54210FE

From: <sip:50765500505@172.26.253.30>;tag=EE04CC4C-1F39

To: <sip:943008712@172.26.253.41>

Date: Mon, 23 Jan 2012 15:03:36 GMT

Call-ID: 1A3859B3-450C11E1-8F7099E3-58230DFA@172.26.253.30

CSeq: 101 INVITE

Allow-Events: presence

Content-Length: 0

.Jan 23 16:16:47.977: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 172.26.253.30:5060;branch=z9hG4bK23C54210FE

From: <sip:50765500505@172.26.253.30>;tag=EE04CC4C-1F39

To: <sip:943008712@172.26.253.41>;tag=8c9c6e7c-528c-4fbf-a0b6-7aaf5f9921e5-41414082

Date: Mon, 23 Jan 2012 15:03:36 GMT

Call-ID: 1A3859B3-450C11E1-8F7099E3-58230DFA@172.26.253.30

CSeq: 101 INVITE

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

Allow-Events: presence

Contact: <sip:943008712@172.26.253.41:5060>

Supported: X-cisco-srtp-fallback

Supported: Geolocation

P-Asserted-Identity: "6000032 OC VITORIA" <sip:6000032@172.26.253.41>

Content-Length: 0

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 172.31.255.1:5060;branch=z9hG4bK6s906530ao0hmmgbp210.1

From: <sip:50765500505@172.31.255.1:5060;user=phone>;tag=SDl7m5f01-172.16.221.2651001+1+8fd00003+69d9dfcc

To: <sip:943008712@172.31.255.4:5060;user=phone>;tag=EE04CC58-0

Date: Mon, 23 Jan 2012 15:16:47 GMT

Call-ID: SDl7m5f01-13b7f302d8f21d7905bab1dd543bf962-c54qtk0

CSeq: 20038592 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: <sip:943008712@172.31.255.4>;party=called;screen=no;privacy=off

Contact: <sip:943008712@172.31.255.4:5060>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

.Jan 23 16:16:52.921: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.26.253.30:5060;branch=z9hG4bK23C54210FE

From: <sip:50765500505@172.26.253.30>;tag=EE04CC4C-1F39

To: <sip:943008712@172.26.253.41>;tag=8c9c6e7c-528c-4fbf-a0b6-7aaf5f9921e5-41414082

Date: Mon, 23 Jan 2012 15:03:36 GMT

Call-ID: 1A3859B3-450C11E1-8F7099E3-58230DFA@172.26.253.30

CSeq: 101 INVITE

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

Allow-Events: presence

Contact: <sip:943008712@172.26.253.41:5060>

Supported: replaces

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires: 1800;refresher=uas

Require: timer

P-Asserted-Identity: "6000032 OC VITORIA" <sip:6000032@172.26.253.41>

Content-Type: application/sdp

Content-Length: 237

v=0

o=CiscoSystemsCCM-SIP 2000 1 IN IP4 172.26.253.41

s=SIP Call

c=IN IP4 172.26.253.47

t=0 0

m=audio 18756 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=ptime:20

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

paolo bevilacqua · ‎01-25-2012

You should always use G.711. Thats gives the best call quality, interwoking, and ease of configuration.

clopezg78 · ‎01-26-2012

Hello Paolo,

I know that the best solution is implement codec G711.

But customer wants G729r8, if you read from the begining i mention that the issue is only with certain incoming calls .e.g Uruguay,Panama.

Regards.

paolo bevilacqua · ‎01-26-2012

Sorry, but considering how quick you've been in low rating my correct posting, made to help, I will refrain from participate again.