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Cisco CME and Calls through SIP provider

Hello, friends.

There are Cisco (C2801-ADVENTERPRISEK9_IVS-M), Version 15.1 (4) M7.

Telephones connected to SCCP, registered SIP from the provider.

When I try to call to test number 4444 through sip in debug I see:

*Feb 10 01:51:25.317: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP XXXXXXXXXXX:5060;branch=z9hG4bK100D02077;rport=5060

From: "TEST" <sip:61@sip.zadarma.com>;tag=131CC60C-1D40

To: <sip:4444@sip.zadarma.com>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7

Call-ID: 27F407C2-910B11E3-969BFFC5-883A8A0E@92.63.108.115

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvf1OFL39Awnou/oMiaFQrf9jyybhFmf", qop="auth"

Server: kamailio (4.0.3 (x86_64/linux))

Content-Length: 0


*Feb 10 01:51:25.325: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:4444@sip.zadarma.com:5060 SIP/2.0

Via: SIP/2.0/UDP XXXXXXXXXX:5060;branch=z9hG4bK100D02077

From: "TEST" <sip:61@sip.zadarma.com>;tag=131CC60C-1D40

To: <sip:4444@sip.zadarma.com>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7

Date: Sun, 09 Feb 2014 21:51:25 GMT

Call-ID: 27F407C2-910B11E3-969BFFC5-883A8A0E@92.63.108.115

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0


Cisco при этом зарегана у провайдера SIP

DC#show sip-ua register status

Line peer expires(sec) registered P-Associ-URI

Configuration:

voice service voip

ip address trusted list

  ipv4 178.16.26.122 255.255.255.255

  ipv4 144.76.42.108 255.255.255.255

  ipv4 176.9.145.115 255.255.255.255

  ipv4 5.9.108.25 255.255.255.255

  ipv4 78.46.95.118 255.255.255.255

  ipv4 89.249.23.194 255.255.255.255

  ipv4 178.16.26.124 255.255.255.255

  ipv4 176.9.85.133 255.255.255.255

  ipv4 46.4.53.86 255.255.255.255

  ipv4 5.9.84.165 255.255.255.255

  ipv4 78.16.26.122 255.255.255.255

  ipv4 77.235.62.222 255.255.255.255

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

sip

  registrar server

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

codec preference 3 g711alaw

!

!

voice register global

max-dn 10

max-pool 10

!

voice register dn  1

number 150

!

voice register dn  2

number 151

!

voice translation-rule 9

rule 1 /^95/ //

!

voice translation-rule 1020

rule 1 /^.$/ /40232/

!

!

voice translation-profile outgoing

translate calling 1020

translate called 9

!

mgcp fax t38 ecm

!

mgcp profile default

!

!

dial-peer voice 2 voip

translation-profile outgoing outgoing

destination-pattern 95....

session protocol sipv2

session target sip-server

voice-class codec 1

no voice-class sip outbound-proxy

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

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

dtmf-relay rtp-nte

no vad

!

!

sip-ua

credentials username 40232 password 7 XXXXXXXXXX realm sip.zadarma.com

authentication username 40232 password 7 XXXXXXXXXXXX realm sip.zadarma.com

registrar dns:sip.zadarma.com:5060 expires 3600

sip-server dns:sip.zadarma.com:5060

connection-reuse

host-registrar

DC#show sip-ua register status

Line                             peer       expires(sec) registered P-Associ-URI

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

150                              40001      12           no

40232                            -1         550          yes

SIP provider says cisco trying to call with the internal call number, and it is necessary in order that have an SIP provider:

Wrong Remote-Party-ID: "Vankuver" <sip:61@<my ip>>;party=calling;

Should be so sip:40232@<my ip>

Please help me!

2 ACCEPTED SOLUTIONS

Accepted Solutions

Cisco CME and Calls through SIP provider

You are missing a parameter called  "modify"

request INVITE sip-header From modify "\"(.*)\" <>" "\"\" <40232>"

Hi.Under sip-uano remote

Hi.

Under sip-ua

no remote-party-id

 

 

HTH

 

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"
24 REPLIES

Cisco CME and Calls through SIP provider

Hi,

Can you please post "debug voice ccapi inout" for an outbound call with "debug voice translation".

You only have single dial-peer configured?

Thanks

Manish

Cisco CME and Calls through SIP provider

Hi Aleksandr.

You can modify your FROM value on your INVITE through a sip profile applyed to your outgoing dialpeer

eg

Voice class sip-profile 10

request INVITE sip-header From modify "\"(.*)\" <>" "\"\" <40232>"

dial-peer voice 2 voip

voice-class sip profile 10

HTH

Regards

Carlo

Please rate all helpful posts

"The more you help the more you learn"

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"The more you help the more you learn"

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New Member

Cisco CME and Calls through SIP provider

I confused something in quotes ... I can not give the proper command

New Member

Cisco CME and Calls through SIP provider

i tried:

request INVITE sip-header From "\"(.*)\" <>" "\"\" <40232>"

and

request INVITE sip-header From "\"(.*)\" <>" "\"\" <40232>"

% Invalid input detected at '^' marker.

Cisco CME and Calls through SIP provider

You are missing a parameter called  "modify"

request INVITE sip-header From modify "\"(.*)\" <>" "\"\" <40232>"

New Member

Cisco CME and Calls through SIP provider

Thank you!

It works!

But I do not hear a voice,silence.

Registering on the inside interface cisco via IPSEC VPN with split tunnel (some networks)

*Feb 10 17:12:17.394: //54273/CC3A49EB9B7E/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x6A59E708

State of The Call        : STATE_ACTIVE

TCP Sockets Used         : NO

Calling Number           : 150

Called Number            : 4444

Source IP Address (Sig  ):

Destn SIP Req Addr:Port  : 178.16.26.122:5060

Destn SIP Resp Addr:Port : 178.16.26.122:5060

Destination Name         : sip.zadarma.com

*Feb 10 17:12:17.394: //54273/CC3A49EB9B7E/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g711alaw

Negotiated Codec Bytes   : 160

Nego. Codec payload      : 8 (tx), 8 (rx)

Negotiated Dtmf-relay    : 6

Dtmf-relay Payload       : 101 (tx), 101 (rx)

Source IP Address (Media):

Source IP Port    (Media): 16632

Destn  IP Address (Media): 178.16.26.124

Destn  IP Port    (Media): 15826

Orig Destn IP Address:Port (Media): [ - ]:0

*Feb 10 17:12:17.554: //54272/CC3A49EB9B7E/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x6A5B4EE8

State of The Call        : STATE_ACTIVE

TCP Sockets Used         : NO

Calling Number           : 150

Called Number            : 954444

Source IP Address (Sig  ): 92.63.108.115

Destn SIP Req Addr:Port  : :42294

Destn SIP Resp Addr:Port : <real ip of my lan from i connect to cisco vpn>:42294

Destination Name : <192.168.11.14 - ip in vpn tunnel>

Cisco CME and Calls through SIP provider

Are you behind nat?

Please post the output of a debug ccsip message during a call setup.

Thanks

Regards

Carlo

Please rate all helpful posts

"The more you help the more you learn"

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New Member

Cisco CME and Calls through SIP provider

Yes, I behind nat.

*Feb 10 18:11:53.425: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:954444@192.168.1.1 SIP/2.0

Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport

Max-Forwards: 70

Contact: <150>

To: "954444"<954444>

From: "150"<150>;tag=7b409f06

Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.

CSeq: 1 INVITE

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

Content-Type: application/sdp

User-Agent: X-Lite release 1104o stamp 56125

Content-Length: 314

v=0

o=- 2 2 IN IP4 192.168.11.14

s=CounterPath X-Lite 3.0

c=IN IP4 192.168.11.14

t=0 0

m=audio 5724 RTP/AVP 107 0 8 101

a=alt:1 2 : gNONJ/Dj BaLJhmb/ 10.200.16.55 5724

a=alt:2 1 : DQ3e8qud c1qVrWui 192.168.11.14 5724

a=fmtp:101 0-15

a=rtpmap:107 BV32/16000

a=rtpmap:101 telephone-event/8000

a=sendrecv

*Feb 10 18:11:53.477: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:4444@sip.zadarma.com:5060 SIP/2.0

Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038E7FF

From: "" <40232>>;tag=169E6BC4-1E16

To: <>4444@sip.zadarma.com>

Date: Mon, 10 Feb 2014 14:11:53 GMT

Call-ID: 2051121A-919411E3-9C13FFC5-883A8A0E@92.63.108.115

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0541864002-2442400227-2618163141-2285537806

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

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

CSeq: 101 INVITE

Timestamp: 1392041513

Contact: <150>outside ip cisco cme:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 262

v=0

o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115

s=SIP Call

c=IN IP4 92.63.108.115

t=0 0

m=audio 18534 RTP/AVP 0 8 101

c=IN IP4 92.63.108.115

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

*Feb 10 18:11:53.481: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport

From: "150"<150>;tag=7b409f06

To: "954444"<954444>

Date: Mon, 10 Feb 2014 14:11:53 GMT

Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

*Feb 10 18:11:53.625: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP outside ip cisco cme:5060;branch=z9hG4bK1038E7FF;rport=5060

From: "" <40232>;tag=169E6BC4-1E16

To: <>4444@sip.zadarma.com>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066

Call-ID: 2051121A-919411E3-9C13FFC5-883A8A0E@92.63.108.115

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn", qop="auth"

Server: kamailio (4.0.3 (x86_64/linux))

Content-Length: 0

*Feb 10 18:11:53.633: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:4444@sip.zadarma.com:5060 SIP/2.0

Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038E7FF

From: "150" <>150@sip.zadarma.com>;tag=169E6BC4-1E16

To: <>4444@sip.zadarma.com>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066

Date: Mon, 10 Feb 2014 14:11:53 GMT

Call-ID: 2051121A-919411E3-9C13FFC5-883A8A0E@92.63.108.115

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

*Feb 10 18:11:53.637: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:4444@sip.zadarma.com:5060 SIP/2.0

Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038F25FC

From: "" <40232>;tag=169E6BC4-1E16

To: <>4444@sip.zadarma.com>

Date: Mon, 10 Feb 2014 14:11:53 GMT

Call-ID: 2051121A-919411E3-9C13FFC5-883A8A0E@92.63.108.115

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0541864002-2442400227-2618163141-2285537806

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

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

CSeq: 102 INVITE

Timestamp: 1392041513

Contact: <150>:5060>

Expires: 180

Allow-Events: telephone-event

Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:4444@sip.zadarma.com:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001

Max-Forwards: 69

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 262

v=0

o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115

s=SIP Call

c=IN IP4 92.63.108.115

t=0 0

m=audio 18534 RTP/AVP 0 8 101

c=IN IP4 92.63.108.115

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

*Feb 10 18:11:53.981: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 trying -- your call is important to us

Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038F25FC;rport=5060

From: "" <40232>;tag=169E6BC4-1E16

To: <>4444@sip.zadarma.com>

Call-ID: 2051121A-919411E3-9C13FFC5-883A8A0E@92.63.108.115

CSeq: 102 INVITE

Server: kamailio (4.0.3 (x86_64/linux))

Content-Length: 0

*Feb 10 18:11:54.385: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 92.63.X:5060;rport=5060;branch=z9hG4bK1038F25FC

Record-Route: <178.16.26.122>

From: "k40232" ;tag=169E6BC4-1E16

To: <>4444@sip.zadarma.com>;tag=as7e8de8e5

Call-ID: 2051121A-919411E3-9C13FFC5-883A8A0E@92.63.108.115

CSeq: 102 INVITE

Server: Zadarma Voip

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

Supported: replaces, timer

Contact: <4444>

Content-Type: application/sdp

Content-Length: 281

v=0

o=root 1942395501 1942395501 IN IP4 178.16.26.124

s=Asterisk PBX

c=IN IP4 178.16.26.124

t=0 0

m=audio 12164 RTP/AVP 8 0 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

*Feb 10 18:11:54.409: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:4444@178.16.26.124:5060 SIP/2.0

Via: SIP/2.0/UDP 92.63.xxxx.xxxx:5060;branch=z9hG4bK10390E63

From: "150" <>150@sip.zadarma.com>;tag=169E6BC4-1E16

To: <>4444@sip.zadarma.com>;tag=as7e8de8e5

Date: Mon, 10 Feb 2014 14:11:53 GMT

Call-ID: 2051121A-919411E3-9C13FFC5-883A8A0E@92.63.108.115

Route: <178.16.26.122>

Max-Forwards: 70

CSeq: 102 ACK

Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:4444@sip.zadarma.com:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001

Allow-Events: telephone-event

Content-Length: 0

*Feb 10 18:11:54.429: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport

From: "150"<150>;tag=7b409f06

To: "954444"<954444>;tag=169E6F78-88E

Date: Mon, 10 Feb 2014 14:11:53 GMT

Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.

CSeq: 1 INVITE

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

Allow-Events: telephone-event

Contact: <4444>:5060;transport=tcp>

Supported: replaces

Server: Cisco-SIPGateway/IOS-12.x

Supported: timer

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 193

v=0

o=CiscoSystemsSIP-GW-UserAgent 149 3396 IN IP4 92.63.108.115

s=SIP Call

c=IN IP4 92.63.108.115

t=0 0

m=audio 17190 RTP/AVP 8

c=IN IP4 92.63.108.115

a=rtpmap:8 PCMA/8000

a=ptime:20

*Feb 10 18:11:54.653: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:4444@92.63.108.115:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 91.231.141.230:42294;branch=z9hG4bK-d8754z-95374017c126c928-1---d8754z-;rport

Max-Forwards: 70

Contact: <150>

To: "954444"<954444>;tag=169E6F78-88E

From: "150"<150>;tag=7b409f06

Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.

CSeq: 1 ACK

User-Agent: X-Lite release 1104o stamp 56125

Content-Length: 0

Cisco CME and Calls through SIP provider

Hi.

Can you please share a show voip rtp connections during an active call.

If you can, please attach also yuor running config

Thanks

Regards

Carlo

Please rate all helpful posts "The more you help the more you learn"
New Member

Cisco CME and Calls through SIP provider

DC#show voip rtp connections

VoIP RTP active connections :

No. CallId     dstCallId  LocalRTP RmtRTP     LocalIP                                RemoteIP

1     54611      54610      16604    19546                            178.16.26.124

Found 1 active RTP connections

I try to send config via email.

Cisco CME and Calls through SIP provider

Hi Aleksandr.

From the above command, the second rtp stream from VG to IP phone doesn't appear.

That's could be the root issue.

Please share your entire vg configuration.

Thanks

Regards

Carlo

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"The more you help the more you learn"

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New Member

Cisco CME and Calls through SIP provider

Hello!

I sent you a letter with the configuration cisco yesterday by mail. Did not receive?

Cisco CME and Calls through SIP provider

Hi Aleksandr.

Yes sorry, I checked my inbox just now.

Watching your config, I see different possible issues.

Which subnet is assigned to ip phones?

You have a route map rerouting traffic for subnet 192.168.1.0/24 to a different gateway.

Try also to add

voice service voip

media flow-around

Please let me know

Regards

Carlo

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"The more you help the more you learn"

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New Member

Cisco CME and Calls through SIP provider

Thank you for your reply!

Phones are in defferent subnes 192.168.0.0/24, 2.0/24, 4.0/24 connect via vpn and gre tunnel.

No problems with calls from them. Problems,when I connect throuth vpn to VG and register the softphone on 192.168.1.1(inside ip VG). My VPN lan 192.168.11.0/24.

New Member

Cisco CME and Calls through SIP provider

I added 

media flow-around

DC#show voip rtp connections

VoIP RTP active connections :

No. CallId     dstCallId  LocalRTP RmtRTP     LocalIP                                RemoteIP

1     55283      55280      17782    16230    outside-ip-cisco                          176.9.145.115

VIP Super Bronze

Cisco CME and Calls through SIP provider

From what I can see here, the Ip Phones are been told by your CCME gateway to send their RTP stream to the public ip of the gateway (92.63.108.115)..The question is can they reach this IP? Looking at the Request URI, the INVITE from the X-lite was sent to the private IP of CCME (192.168.1.1) That suggests to me that those phones may not be ale to reach that public IP.

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Cisco CME and Calls through SIP provider

Hi Aleksandr,

We recieve a  "SIP/2.0 407 Proxy Authentication Required", you need to check with your Service Provider:

*Feb 10 18:11:53.625: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP outside ip cisco cme:5060;branch=z9hG4bK1038E7FF;rport=5060

From: "" <40232>;tag=169E6BC4-1E16

To: <>4444@sip.zadarma.com>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066

Call-ID: 2051121A-919411E3-9C13FFC5-883A8A0E@92.63.108.115

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn", qop="auth"

Server: kamailio (4.0.3 (x86_64/linux))

Content-Length: 0

Kindly rate this post accordingly.

Regards,

Kevin

Cisco CME and Calls through SIP provider

Hi.

Please add what follows and send a debug ccsip

voice service voip

sip

bind all source f0/1

HTH

Regards

Carlo

Please rate all helpful posts

"The more you help the more you learn"

Please rate all helpful posts "The more you help the more you learn"
New Member

Hello,friends!All worked, but

Hello,friends!

All worked, but again recently had a problem with outgoing calls with the same provider. 
My config:

sip-ua
 credentials username 141756 password 7 pass realm sip.zadarma.com
 authentication username 141756 password 7 pass
 registrar 1 dns:sip.zadarma.com expires 3600
 sip-server dns:sip.zadarma.com
 connection-reuse
 host-registrar

dial-peer voice 2 voip
 translation-profile outgoing outgoing
 destination-pattern 9...........
 session protocol sipv2
 session target sip-server
 voice-class codec 1
 no voice-class sip outbound-proxy
 voice-class sip profiles 20
 voice-class sip bind control source-interface FastEthernet0/0
 voice-class sip bind media source-interface FastEthernet0/0
 dtmf-relay rtp-nte sip-notify
 no vad

voice class sip-profiles 20
 request INVITE sip-header From modify "\"(.*)\" <sip:(.*)@(.*)>" "\"\" <sip:141756@92.63.108.115>"

 

voice translation-rule 9
 rule 1 /^9/ //
voice translation-rule 1030
 rule 1 /^.*/ /141756/

voice translation-profile outgoing
 translate calling 1030
 translate called 9

 

debug ccsip message

Jun  4 23:08:51.247: //17386/1C4006E18C9A/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:79118268147@sip.zadarma.com:5060 SIP/2.0

Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK4A51D1A

Remote-Party-ID: "Vankuver" <sip:141756@92.63.108.115>;party=calling;screen=no;privacy=off

From: "" <sip:141756@92.63.108.115>;tag=30056CAC-7EF

To: <sip:79118268147@sip.zadarma.com>

Date: Wed, 04 Jun 2014 18:08:51 GMT

Call-ID: 1DDB67F5-EB4A11E3-8C9FA529-5E0D91DD@92.63.108.115

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0473958113-3947500003-2358945065-1577947613

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: 1401905331

Contact: <sip:141756@92.63.108.115:5060>

Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 309

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 4250 6098 IN IP4 92.63.108.115

s=SIP Call

c=IN IP4 92.63.108.115

t=0 0

m=audio 17138 RTP/AVP 0 18 8 101

c=IN IP4 92.63.108.115

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

 

Jun  4 23:08:51.307: //17386/1C4006E18C9A/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK4A51D1A;rport=5060

From: "" <sip:141756@92.63.108.115>;tag=30056CAC-7EF

To: <sip:79118268147@sip.zadarma.com>;tag=b638310eda6e4a73cf10b7fe3c94c572.53f6

Call-ID: 1DDB67F5-EB4A11E3-8C9FA529-5E0D91DD@92.63.108.115

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="U49vk1OPbmeoTsQ5WUcB2MQbTNbuaDse", qop="auth"

Server: kamailio (4.1.2 (x86_64/linux))

Content-Length: 0

 

 

Jun  4 23:08:51.315: //17386/1C4006E18C9A/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:79118268147@sip.zadarma.com:5060 SIP/2.0

Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK4A51D1A

From: "Vankuver" <sip:141756@92.63.108.115>;tag=30056CAC-7EF

To: <sip:79118268147@sip.zadarma.com>;tag=b638310eda6e4a73cf10b7fe3c94c572.53f6

Date: Wed, 04 Jun 2014 18:08:51 GMT

Call-ID: 1DDB67F5-EB4A11E3-8C9FA529-5E0D91DD@92.63.108.115

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

 

 

Jun  4 23:08:51.315: //17386/1C4006E18C9A/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:79118268147@sip.zadarma.com:5060 SIP/2.0

Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK4A61F2E

Remote-Party-ID: "Vankuver" <sip:141756@92.63.108.115>;party=calling;screen=no;privacy=off

From: "" <sip:141756@92.63.108.115>;tag=30056CAC-7EF

To: <sip:79118268147@sip.zadarma.com>

Date: Wed, 04 Jun 2014 18:08:51 GMT

Call-ID: 1DDB67F5-EB4A11E3-8C9FA529-5E0D91DD@92.63.108.115

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0473958113-3947500003-2358945065-1577947613

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: 1401905331

Contact: <sip:141756@92.63.108.115:5060>

Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"

Expires: 180

Allow-Events: telephone-event

Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:79118268147@sip.zadarma.com:5060",response="2e910417008d2e2006242c70f531f177",nonce="U49vk1OPbmeoTsQ5WUcB2MQbTNbuaDse",cnonce="B1B29B16",qop=auth,algorithm=md5,nc=00000001

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 309

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 4250 6098 IN IP4 92.63.108.115

s=SIP Call

c=IN IP4 92.63.108.115

t=0 0

m=audio 17138 RTP/AVP 0 18 8 101

c=IN IP4 92.63.108.115

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

 

Jun  4 23:08:51.819: //17386/1C4006E18C9A/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:79118268147@sip.zadarma.com:5060 SIP/2.0

Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK4A61F2E

Remote-Party-ID: "Vankuver" <sip:141756@92.63.108.115>;party=calling;screen=no;privacy=off

From: "" <sip:141756@92.63.108.115>;tag=30056CAC-7EF

To: <sip:79118268147@sip.zadarma.com>

Date: Wed, 04 Jun 2014 18:08:51 GMT

Call-ID: 1DDB67F5-EB4A11E3-8C9FA529-5E0D91DD@92.63.108.115

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0473958113-3947500003-2358945065-1577947613

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: 1401905331

Contact: <sip:141756@92.63.108.115:5060>

Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"

Expires: 180

Allow-Events: telephone-event

Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:79118268147@sip.zadarma.com:5060",response="2u e910417008d2e2006242c70f531f177",nonce="U49vk1OPbmeoTsQ5WUcB2MQbTNbuaDse",cnonce="B1B29B16",qop=auth,algorithm=md5,nc=00000001

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 309

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 4250 6098 IN IP4 92.63.108.115

s=SIP Call

c=IN IP4 92.63.108.115

t=0 0

m=audio 17138 RTP/AVP 0 18 8 101

c=IN IP4 92.63.108.115

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

 

Jun  4 23:08:52.819: //17386/1C4006E18C9A/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:79118268147@sip.zadarma.com:5060 SIP/2.0

Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK4A61F2E

Remote-Party-ID: "Vankuver" <sip:141756@92.63.108.115>;party=calling;screen=no;privacy=off

From: "" <sip:141756@92.63.108.115>;tag=30056CAC-7EF

To: <sip:79118268147@sip.zadarma.com>

Date: Wed, 04 Jun 2014 18:08:52 GMT

Call-ID: 1DDB67F5-EB4A11E3-8C9FA529-5E0D91DD@92.63.108.115

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0473958113-3947500003-2358945065-1577947613

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: 1401905332

Contact: <sip:141756@92.63.108.115:5060>

Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"

Expires: 180

Allow-Events: telephone-event

Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:79118268147@sip.zadarma.com:5060",response="2all

All possible debugging has been turned off

DC#e910417008d2e2006242c70f531f177",nonce="U49vk1OPbmeoTsQ5WUcB2MQbTNbuaDse",cnonce="B1B29B16",qop=auth,algorithm=md5,nc=00000001

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 309

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 4250 6098 IN IP4 92.63.108.115

s=SIP Call

c=IN IP4 92.63.108.115

t=0 0

m=audio 17138 RTP/AVP 0 18 8 101

c=IN IP4 92.63.108.115

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

 

My router reseive one message from SIP-provider:

Received:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK4A51D1A;rport=5060

From: "" <sip:141756@92.63.108.115>;tag=30056CAC-7EF

To: <sip:79118268147@sip.zadarma.com>;tag=b638310eda6e4a73cf10b7fe3c94c572.53f6

Call-ID: 1DDB67F5-EB4A11E3-8C9FA529-5E0D91DD@92.63.108.115

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="U49vk1OPbmeoTsQ5WUcB2MQbTNbuaDse", qop="auth"

Server: kamailio (4.1.2 (x86_64/linux))

Content-Length: 0

 

IN debug ccsip errors I see:

SIP/Error/sipSPI_ipip_set_history_info_header: ccb->src_addr_str is NULL

 

Please,help me!!!

 

 

 

 

Hi Aleksandr.On your dialpeer

Hi Aleksandr.

On your dialpeer 2 remove voice class codec and force codec to g711ulaw.

 

Please try it and let me know.

 

HTH

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"
New Member

no effect:Sent:INVITE sip

no effect:

Sent:
INVITE sip:79118268147@sip.zadarma.com:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK2417FB
Remote-Party-ID: "Vankuver" <sip:141756@92.63.108.115>;party=calling;screen=no;privacy=off
From: "" <sip:141756@92.63.108.115>;tag=3035EC-1A5D
To: <sip:79118268147@sip.zadarma.com>
Date: Wed, 04 Jun 2014 20:44:50 GMT
Call-ID: E8B58412-EB5F11E3-8018FAC8-2C2E981E@92.63.108.115
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3877290282-3948876259-2148793032-0741251102
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: 1401914691
Contact: <sip:141756@92.63.108.115:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273

v=0
o=CiscoSystemsSIP-GW-UserAgent 1053 2671 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 19012 RTP/AVP 18 101
c=IN IP4 92.63.108.115
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Jun  5 01:44:51.067: //31/E71AC12A8013/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK2417FB;rport=5060
From: "" <sip:141756@92.63.108.115>;tag=3035EC-1A5D
To: <sip:79118268147@sip.zadarma.com>;tag=b638310eda6e4a73cf10b7fe3c94c572.ea16
Call-ID: E8B58412-EB5F11E3-8018FAC8-2C2E981E@92.63.108.115
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="U4+Bw1OPgJcXokYeoDfJkKMwzKC2C3v3", qop="auth"
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0


Jun  5 01:44:51.075: //31/E71AC12A8013/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:79118268147@sip.zadarma.com:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK2417FB
From: "Vankuver" <sip:141756@92.63.108.115>;tag=3035EC-1A5D
To: <sip:79118268147@sip.zadarma.com>;tag=b638310eda6e4a73cf10b7fe3c94c572.ea16
Date: Wed, 04 Jun 2014 20:44:50 GMT
Call-ID: E8B58412-EB5F11E3-8018FAC8-2C2E981E@92.63.108.115
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


Jun  5 01:44:51.079: //31/E71AC12A8013/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:79118268147@sip.zadarma.com:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK251416
Remote-Party-ID: "Vankuver" <sip:141756@92.63.108.115>;party=calling;screen=no;privacy=off
From: "" <sip:141756@92.63.108.115>;tag=3035EC-1A5D
To: <sip:79118268147@sip.zadarma.com>
Date: Wed, 04 Jun 2014 20:44:51 GMT
Call-ID: E8B58412-EB5F11E3-8018FAC8-2C2E981E@92.63.108.115
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3877290282-3948876259-2148793032-0741251102
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: 1401914691
Contact: <sip:141756@92.63.108.115:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:79118268147@sip.zadarma.com:5060",response="ceb43d310176422368146b753373de99",nonce="U4+Bw1OPgJcXokYeoDfJkKMwzKC2C3v3",cnonce="805CE107",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273

v=0
o=CiscoSystemsSIP-GW-UserAgent 1053 2671 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 19012 RTP/AVP 18 101
c=IN IP4 92.63.108.115
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Jun  5 01:44:51.579: //31/E71AC12A8013/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:79118268147@sip.zadarma.com:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK251416
Remote-Party-ID: "Vankuver" <sip:141756@92.63.108.115>;party=calling;screen=no;privacy=off
From: "" <sip:141756@92.63.108.115>;tag=3035EC-1A5D
To: <sip:79118268147@sip.zadarma.com>
Date: Wed, 04 Jun 2014 20:44:51 GMT
Call-ID: E8B58412-EB5F11E3-8018FAC8-2C2E981E@92.63.108.115
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3877290282-3948876259-2148793032-0741251102
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: 1401914691
Contact: <sip:141756@92.63.108.115:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:79118268147@sip.zadarma.com:5060",response="ceb43d310176422368146b753373de99",nonce="U4+Bw1OPgJcXokYeoDfJkKMwzKC2C3v3",cnonce="805CE107",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273

v=0
o=CiscoSystemsSIP-GW-UserAgent 1053 2671 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 19012 RTP/AVP 18 101
c=IN IP4 92.63.108.115
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Jun  5 01:44:52.579: //31/E71AC12A8013/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:79118268147@sip.zadarma.com:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK251416
Remote-Party-ID: "Vankuver" <sip:141756@92.63.108.115>;party=calling;screen=no;privacy=off
From: "" <sip:141756@92.63.108.115>;tag=3035EC-1A5D
To: <sip:79118268147@sip.zadarma.com>
Date: Wed, 04 Jun 2014 20:44:52 GMT
Call-ID: E8B58412-EB5F11E3-8018FAC8-2C2E981E@92.63.108.115
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3877290282-3948876259-2148793032-0741251102
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: 1401914692
Contact: <sip:141756@92.63.108.115:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:79118268147@sip.zadarma.com:5060",response="ceb43d310176422368146b753373de99",nonce="U4+Bw1OPgJcXokYeoDfJkKMwzKC2C3v3",cnonce="805CE107",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273

v=0
o=CiscoSystemsSIP-GW-UserAgent 1053 2671 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 19012 RTP/AVP 18 101
c=IN IP4 92.63.108.115
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Jun  5 01:44:54.583: //31/E71AC12A8013/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:79118268147@sip.zadarma.com:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK251416
Remote-Party-ID: "Vankuver" <sip:141756@92.63.108.115>;party=calling;screen=no;privacy=off
From: "" <sip:141756@92.63.108.115>;tag=3035EC-1A5D
To: <sip:79118268147@sip.zadarma.com>
Date: Wed, 04 Jun 2014 20:44:54 GMT
Call-ID: E8B58412-EB5F11E3-8018FAC8-2C2E981E@92.63.108.115
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3877290282-3948876259-2148793032-0741251102
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: 1401914694
Contact: <sip:141756@92.63.108.115:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:79118268147@sip.zadarma.com:5060",response="ceb43d310176422368146b753373de99",nonce="U4+Bw1OPgJcXokYeoDfJkKMwzKC2C3v3",cnonce="805CE107",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273

v=0
o=CiscoSystemsSIP-GW-UserAgent 1053 2671 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 19012 RTP/AVP 18 101
c=IN IP4 92.63.108.115
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

dial-peer voice 2 voip
 translation-profile outgoing outgoing
 destination-pattern 9...........
 session protocol sipv2
 session target sip-server
 no voice-class sip outbound-proxy
 voice-class sip profiles 20
 voice-class sip bind control source-interface FastEthernet0/0
 voice-class sip bind media source-interface FastEthernet0/0
 dtmf-relay rtp-nte sip-notify
 no vad
dial-peer voice 3 voip
 translation-profile incoming incoming
 incoming called-number 141756
 voice-class codec 1
 voice-class sip bind control source-interface FastEthernet0/0
 voice-class sip bind media source-interface FastEthernet0/0
 dtmf-relay rtp-nte
 no vad

 

voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
 codec preference 3 g711alaw

 

 

Hi.Under sip-uano remote

Hi.

Under sip-ua

no remote-party-id

 

 

HTH

 

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

Hi.On dialpeer 2 you removed

Hi.

On dialpeer 2 you removed voice-class codec as asked but you didn't add "codec g711ulaw" as I asked you.

Please modify the config with codec and previous suggestion and let me know.

 

Thanks

 

Regards

 

 

Carlo

Please rate all helpful posts "The more you help the more you learn"
New Member

After adding no remote-party

After adding no remote-party-id all works.

My config:

sip-ua
 credentials username 141756 password 7 <pass> realm sip.zadarma.com
 authentication username 141756 password 7 <pass>
 no remote-party-id
 registrar 1 dns:sip.zadarma.com expires 3600
 sip-server dns:sip.zadarma.com
 connection-reuse
 host-registrar

 

dial-peer voice 2 voip
 translation-profile outgoing outgoing
 preference 1
 destination-pattern 98..........
 session protocol sipv2
 session target sip-server
 voice-class codec 1
 no voice-class sip outbound-proxy
 voice-class sip profiles 20
 voice-class sip bind control source-interface FastEthernet0/0
 voice-class sip bind media source-interface FastEthernet0/0
 dtmf-relay rtp-nte sip-notify
 no vad

 

Thank you!!!

 

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