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Thiago Cella
Beginner

SIP Trunk - No audio

Hi,

I configured the follow , but during the call there is no audio on both sides, could you help me?


version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
!
!
no aaa new-model
dot11 syslog
ip source-route
!
!
!
!
!
ip cef
no ipv6 cef
multilink bundle-name authenticated
!
!
!
!
!
!
voice service voip
address-hiding
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
h323
emptycapability
no telephony-service ccm-compatible
sip
registrar server
!
voice class sip-profiles 100
request INVITE sip-header From modify "<sip:(.*)@1.1.1.1>" "<sip:1111.1111.1111@1.1.1.1>"
!
!
!
!
!
voice-card 0
!
crypto pki token default removal timeout 0
!
!
!
!
license udi pid CISCO2801 sn FHK1147F1JU
license accept end user agreement
!
!
!
!
!
!
!
interface FastEthernet0/0
ip address dhcp
duplex auto
speed auto
!
interface FastEthernet0/1
ip address 192.168.23.1 255.255.255.0
duplex auto
speed auto
!
ip forward-protocol nd
!
!
no ip http server
no ip http secure-server
!
!
!
!
control-plane
!
!
voice-port 0/2/0
!
voice-port 0/2/1
!
!
!
mgcp profile default
!
!

!
dial-peer voice 1010 voip
description OUTBOUND to ISP
destination-pattern .T
session protocol sipv2
session target sip-server
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 100
dtmf-relay rtp-nte
codec g711alaw
clid network-number 1122222222
!
!
sip-ua
credentials username 1111.1111.1111 password XXXXXXXX realm isp.com
authentication username 1111.1111.1111 password XXXXXXXX  realm isp.com
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar ipv4:1.1.1.1 expires 1800
sip-server ipv4:1.1.1.1
host-registrar
!
!
telephony-service
max-ephones 15
max-dn 15
ip source-address 192.168.23.1 port 2000
max-conferences 4 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn 1 dual-line
number 1000 no-reg primary

!
ephone 1
description joao-teste
mac-address B888.E3E0.A621
button 1:1
!
!

Follow debug :


*Aug 10 22:44:47.707: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:99999999@1.1.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK56B2F
From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485
To: <sip:99999999@1.1.1.1>
Date: Wed, 10 Aug 2016 22:44:47 GMT
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3734615084-1585582566-2156900920-2418771262
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1470869087
Contact: <sip:1122222222@192.168.1.30:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 271

v=0
o=CiscoSystemsSIP-GW-UserAgent 7869 7950 IN IP4 192.168.1.30
s=SIP Call
c=IN IP4 192.168.1.30
t=0 0
m=audio 17740 RTP/AVP 8 101 19
c=IN IP4 192.168.1.30
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20

*Aug 10 22:44:47.723: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 3.3.3.3:49152;branch=z9hG4bK56B2F;rport=64718;received=192.168.1.30
From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485
To: <sip:99999999@1.1.1.1>
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
CSeq: 101 INVITE
Server: nt
Content-Length: 0


*Aug 10 22:44:47.727: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 3.3.3.3:49152;branch=z9hG4bK56B2F;rport=64718;received=192.168.1.30
From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485
To: <sip:99999999@1.1.1.1>;tag=d8ba4cf51f1985334ea8c7911ac42cf4.5f85
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="isp.com", nonce="57abaffb00011b77d2f49ecf6a217052132bc67a5f4535a3"
Server: nt
Content-Length: 0


*Aug 10 22:44:47.735: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:99999999@1.1.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK56B2F
From: <sip:1122222222@1.1.1.1>;tag=8E61D8-485
To: <sip:99999999@1.1.1.1>;tag=d8ba4cf51f1985334ea8c7911ac42cf4.5f85
Date: Wed, 10 Aug 2016 22:44:47 GMT
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


*Aug 10 22:44:47.735: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:99999999@1.1.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK57150B
From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485
To: <sip:99999999@1.1.1.1>
Date: Wed, 10 Aug 2016 22:44:47 GMT
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3734615084-1585582566-2156900920-2418771262
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1470869087
Contact: <sip:1122222222@192.168.1.30:5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="1111.1111.1111",realm="isp.com",uri="sip:99999999@1.1.1.1:5060",response="aa74f1052b7797de81f57343cd025366",nonce="57abaffb00011b77d2f49ecf6a217052132bc67a5f4535a3",algorithm=md5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 271

v=0
o=CiscoSystemsSIP-GW-UserAgent 7869 7950 IN IP4 192.168.1.30
s=SIP Call
c=IN IP4 192.168.1.30
t=0 0
m=audio 17740 RTP/AVP 8 101 19
c=IN IP4 192.168.1.30
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20

*Aug 10 22:44:47.743: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 3.3.3.3:49152;branch=z9hG4bK57150B;rport=64718;received=192.168.1.30
From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485
To: <sip:99999999@1.1.1.1>
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
CSeq: 102 INVITE
Server: nt
Content-Length: 0


*Aug 10 22:44:52.171: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 3.3.3.3:49152;rport=64718;received=192.168.1.30;branch=z9hG4bK57150B
Record-Route: <sip:1.1.1.1:5060;lr=on;ftag=8E61D8-485;did=ff7.0f54d5d6;vsf=AAAAAAAABAMeAgkABx94CQgDbgcLGR8FAwAEAzAuNTA->
Require: 100rel
Contact: <sip:01199999999@172.16.0.1:5060>
To: <sip:99999999@1.1.1.1>;tag=ee7c6277
From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, INFO
RSeq: 1
Content-Length: 0


*Aug 10 22:44:52.179: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Sent:
PRACK sip:01199999999@172.16.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK581FD4
From: <sip:1122222222@1.1.1.1>;tag=8E61D8-485
To: <sip:99999999@1.1.1.1>;tag=ee7c6277
Date: Wed, 10 Aug 2016 22:44:47 GMT
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
CSeq: 103 PRACK
RAck: 1 102 INVITE
Route: <sip:1.1.1.1:5060;lr=on;ftag=8E61D8-485;did=ff7.0f54d5d6;vsf=AAAAAAAABAMeAgkABx94CQgDbgcLGR8FAwAEAzAuNTA->
Allow-Events: telephone-event
Proxy-Authorization: Digest username="1111.1111.1111",realm="isp.com",uri="sip:01199999999@172.16.0.1:5060",response="b8d26f88b22916bdfbc04c211c4b7fee",nonce="57abaffb00011b77d2f49ecf6a217052132bc67a5f4535a3",algorithm=md5
Max-Forwards: 70
Content-Length: 0


*Aug 10 22:44:52.195: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 3.3.3.3:49152;rport=64718;received=192.168.1.30;branch=z9hG4bK581FD4
Contact: <sip:01199999999@172.16.0.1:5060>
To: <sip:99999999@1.1.1.1>;tag=ee7c6277
From: <sip:1122222222@1.1.1.1>;tag=8E61D8-485
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
CSeq: 103 PRACK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, INFO
Content-Length: 0


*Aug 10 22:44:52.867: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 3.3.3.3:49152;rport=64718;received=192.168.1.30;branch=z9hG4bK57150B
Record-Route: <sip:1.1.1.1:5060;lr=on;ftag=8E61D8-485;did=ff7.0f54d5d6;vsf=AAAAAAAABAMeAgkABx94CQgDbgcLGR8FAwAEAzAuNTA->
Require: 100rel
Contact: <sip:01199999999@172.16.0.1:5060>
To: <sip:99999999@1.1.1.1>;tag=ee7c6277
From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, INFO
Content-Type: application/sdp
RSeq: 2
Content-Length: 284

v=0
o=MG4000|2.0 15467 15657 IN IP4 172.11.10.82
s=-
c=IN IP4 200.236.219.50
t=0 0
m=audio 38748 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0
a=X-vrzcap:identification bin=CMG1143 Prot=mgcp App=MG

*Aug 10 22:44:52.887: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Sent:
PRACK sip:01199999999@172.16.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK59858
From: <sip:1122222222@1.1.1.1>;tag=8E61D8-485
To: <sip:99999999@1.1.1.1>;tag=ee7c6277
Date: Wed, 10 Aug 2016 22:44:47 GMT
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
CSeq: 104 PRACK
RAck: 2 102 INVITE
Route: <sip:1.1.1.1:5060;lr=on;ftag=8E61D8-485;did=ff7.0f54d5d6;vsf=AAAAAAAABAMeAgkABx94CQgDbgcLGR8FAwAEAzAuNTA->
Allow-Events: telephone-event
Proxy-Authorization: Digest username="1111.1111.1111",realm="isp.com",uri="sip:01199999999@172.16.0.1:5060",response="b8d26f88b22916bdfbc04c211c4b7fee",nonce="57abaffb00011b77d2f49ecf6a217052132bc67a5f4535a3",algorithm=md5
Max-Forwards: 70
Content-Length: 0


*Aug 10 22:44:52.903: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 3.3.3.3:49152;rport=64718;received=192.168.1.30;branch=z9hG4bK59858
Contact: <sip:01199999999@172.16.0.1:5060>
To: <sip:99999999@1.1.1.1>;tag=ee7c6277
From: <sip:1122222222@1.1.1.1>;tag=8E61D8-485
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
CSeq: 104 PRACK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, INFO
Content-Length: 0


*Aug 10 22:44:57.367: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 3.3.3.3:49152;rport=64718;received=192.168.1.30;branch=z9hG4bK57150B
Record-Route: <sip:1.1.1.1:5060;lr=on;ftag=8E61D8-485;did=ff7.0f54d5d6;vsf=AAAAAAAABAMeAgkABx94CQgDbgcLGR8FAwAEAzAuNTA->
Contact: <sip:01199999999@172.16.0.1:5060>
To: <sip:99999999@1.1.1.1>;tag=ee7c6277
From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, INFO
Content-Length: 0


*Aug 10 22:44:57.379: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:01199999999@172.16.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5AEB7
From: <sip:1122222222@1.1.1.1>;tag=8E61D8-485
To: <sip:99999999@1.1.1.1>;tag=ee7c6277
Date: Wed, 10 Aug 2016 22:44:47 GMT
Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30
Route: <sip:1.1.1.1:5060;lr=on;ftag=8E61D8-485;did=ff7.0f54d5d6;vsf=AAAAAAAABAMeAgkABx94CQgDbgcLGR8FAwAEAzAuNTA->
Max-Forwards: 70
CSeq: 102 ACK
Proxy-Authorization: Digest username="1111.1111.1111",realm="isp.com",uri="sip:99999999@1.1.1.1:5060",response="aa74f1052b7797de81f57343cd025366",nonce="57abaffb00011b77d2f49ecf6a217052132bc67a5f4535a3",algorithm=md5
Allow-Events: telephone-event
Content-Length: 0

9 REPLIES 9
Dennis Mink
Advisor

Can you route directly between 192.168.1.30 and 200.236.219.50  (the end point IP addresses of you SIP SDP)? Is there a Firewall in the path?

also does the call establish audio when initiated from the opposite direction?

Please remember to rate useful posts, by clicking on the stars below.

Hi

There is no firewall. This test is a outbound Call, i will test inbound Call and report the results.

Tks 

so can you route directly between the two IP addresses?

Please remember to rate useful posts, by clicking on the stars below.

Yes correct.

One more information ,  I made a test with  same account with X-lite client on my PC and works.

The follow LOG is appering on Cisco, when I try make a call. I

Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 201.69.228.73:49152;branch=z9hG4bK261412;rport=57056;received=192.168.1.30
From: <sip:1 1111.1111.1111@1.1.1.1>;tag=12E370-1A20
To: <sip:9999999@187.63.130.50>;tag=18f01cbe78aa7d69fd8f3e0a8ea294be.fda4
Call-ID: 6E23CBF1-5EF811E6-802CC533-85BC4FB0@192.168.1.30
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="isp.com", nonce="57ac7536000133f7398173a31fd71115b45acad43b6f9f70"
Server: nt
Content-Length: 0

Is this X-lite registered on call manager?

please attached the sip traces for a failed and a successful calll

Please remember to rate useful posts, by clicking on the stars below.

No, i register the X-lite with the SIP account provide by ISP.

Him

I don't see 'bind all source' command under voice service voip > sip. You need to define source interface for media and signaling which is reachable from your ITSP. Else, routing table will be used which might not be correct.

Also, during an active call please share the output of 'show call active voice brief' and 'show voip rtp conn'

Hi Friends,

Sorry for the late, the problem was on router of ISP. I made more tests using x-lite and the some calls calls didnt work too. So i configured the link directly on router and all works.

tks

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