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Dial out from OCS does not work


Andrep
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Hi all,

 

I configured PBXNSIP for OCS.

I can dial in from a PSTN device to an extension and/or to the OCS, but i cannot dial out from OCS to a PSTN device (It works from an extension which is directly connected to PBXNSIP)

 

I have two trunks configured (1 to OCS and 1 to the GW for PSTN connectivity)

 

Here is the log (Communicator is +41434435683 and i want to dial +41763801234 which is a mobile phone):

 

[5] 2008/06/03 15:23:13: SIP port accept from 172.25.20.66:4749

[9] 2008/06/03 15:23:13: SIP Rx tcp:172.25.20.66:4749:

INVITE sip:+41763801234@172.25.20.110;user=phone SIP/2.0

FROM: <sip:+41434435683@mediation.collabcom.ch;user=phone>;epid=FF7797D3EA;tag=b42ca2ac31

TO: <sip:+41763801234@172.25.20.110;user=phone>

CSEQ: 13 INVITE

CALL-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8

CONTACT: <sip:MEDIATION.collabcom.ch:5060;transport=Tcp;maddr=172.25.20.66;ms-opaque=3d8ccc5b0ddbda89>

CONTENT-LENGTH: 302

SUPPORTED: 100rel

USER-AGENT: RTCC/3.0.0.0 MediationServer

CONTENT-TYPE: application/sdp; charset=utf-8

ALLOW: UPDATE

ALLOW: Ack, Cancel, Bye,Invite

 

v=0

o=- 0 0 IN IP4 172.25.20.66

s=session

c=IN IP4 172.25.20.66

b=CT:1000

t=0 0

m=audio 55808 RTP/AVP 97 101 0 8

c=IN IP4 172.25.20.66

a=rtcp:55809

a=label:Audio

a=rtpmap:97 RED/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=ptime:20

 

[9] 2008/06/03 15:23:13: Resolve 5679: tcp 172.25.20.66 4749

[9] 2008/06/03 15:23:13: SIP Tx tcp:172.25.20.66:4749:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8

From: <sip:+41434435683@mediation.collabcom.ch;user=phone>;epid=FF7797D3EA;tag=b42ca2ac31

To: <sip:+41763801234@172.25.20.110;user=phone>;tag=3b5230389e

Call-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094

CSeq: 13 INVITE

Content-Length: 0

 

 

[9] 2008/06/03 15:23:13: Resolve 5680: tcp 172.25.20.66 4749

[9] 2008/06/03 15:23:13: SIP Tx tcp:172.25.20.66:4749:

SIP/2.0 404 Not Found

Via: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8

From: <sip:+41434435683@mediation.collabcom.ch;user=phone>;epid=FF7797D3EA;tag=b42ca2ac31

To: <sip:+41763801234@172.25.20.110;user=phone>;tag=3b5230389e

Call-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094

CSeq: 13 INVITE

Contact: <sip:+41763801234@172.25.20.110:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.10.2474

Content-Length: 0

 

 

[9] 2008/06/03 15:23:13: Resolve 5681: tcp 172.25.20.66 4749

[9] 2008/06/03 15:23:13: SIP Tx tcp:172.25.20.66:4749:

SIP/2.0 404 Not Found

Via: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8

From: <sip:+41434435683@mediation.collabcom.ch;user=phone>;epid=FF7797D3EA;tag=b42ca2ac31

To: <sip:+41763801234@172.25.20.110;user=phone>;tag=3b5230389e

Call-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094

CSeq: 13 INVITE

Contact: <sip:+41763801234@172.25.20.110:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.10.2474

Content-Length: 0

 

 

[9] 2008/06/03 15:23:13: SIP Rx tcp:172.25.20.66:4749:

ACK sip:+41763801234@172.25.20.110;user=phone SIP/2.0

FROM: <sip:+41434435683@mediation.collabcom.ch;user=phone>;tag=b42ca2ac31;epid=FF7797D3EA

TO: <sip:+41763801234@172.25.20.110;user=phone>;tag=3b5230389e

CSEQ: 13 ACK

CALL-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8

CONTENT-LENGTH: 0

 

 

Thanks for your help. Could it be that my Dial Plan is not correct? I also need to replace the + to 00!

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Hi Andrep,

 

I assume that you have read my wiki page for Basic Setup for pbxnsip / Office Communications Server 2007 Interoperability OCS.

If not, please read it very careful.

 

I can see that the NOT FOUND is coming prompt from the pbxnsip. So I guess another trunk (outbound) is not involved, you need to check the trunk, configured to the OCS Meditiation Server again.

 

Please compare your configuration with the one in the wiki. Like described, make sure that the Account under "Assume that calls come from user:" is existing. It must only exist, a registration is not necessary.

 

btw.: I was trying to call your +41434435683 and it was succesfully ringing :D , but did no redirect to an Exchange 2007 Unified Messaging Server :rolleyes: Simply Decline after some rings. Are you using the Ferrari Office Master now with pbxnsip for PSTN?

 

You can check your call log, find my Berlin number and call me back if you like or simply look at my signature her ;)

 

Best regards,

 

Jan

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  • 1 month later...

Hi Jan,

 

It works now. I had to replace the ip-address of the mediation server with the fqdn.

 

Regarding the OfficeMaster Gateway I have still problems (no audio --> I guess codec mismatch). I'm in contact with their support and I will post the config as soon as the problems are solved.

 

Hi Andrep,

 

I assume that you have read my wiki page for Basic Setup for pbxnsip / Office Communications Server 2007 Interoperability OCS.

If not, please read it very careful.

 

I can see that the NOT FOUND is coming prompt from the pbxnsip. So I guess another trunk (outbound) is not involved, you need to check the trunk, configured to the OCS Meditiation Server again.

 

Please compare your configuration with the one in the wiki. Like described, make sure that the Account under "Assume that calls come from user:" is existing. It must only exist, a registration is not necessary.

 

btw.: I was trying to call your +41434435683 and it was succesfully ringing :D , but did no redirect to an Exchange 2007 Unified Messaging Server :D Simply Decline after some rings. Are you using the Ferrari Office Master now with pbxnsip for PSTN?

 

You can check your call log, find my Berlin number and call me back if you like or simply look at my signature her ;)

 

Best regards,

 

Jan

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  • 1 month later...

Hi all,

I have a problem...when I make a call from MOC, if the called doesn't answer, after 180+100rel I can not send the message "480 Temporally not Available": the connection to Mediation server is refused after 1 minute of inactivity.

 

Someone can help me?

Thanks

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Hi all,

I have a problem...when I make a call from MOC, if the called doesn't answer, after 180+100rel I can not send the message "480 Temporally not Available": the connection to Mediation server is refused after 1 minute of inactivity.

 

Someone can help me?

Thanks

 

One thing you should check in your trunk configuration "Remote Party/Privacy Indication", make sure it is setting to "No Indication".

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  • 7 months later...

Hi,

i have a same problem, i can't call out from communicator to iP-Phone,

i can call from iP-Phone to communicator and communicator to communicator,

i don't have a media gateway just mediation server with direct sip trunk to Mitel 3300.

do you have some suggestions ?

thank's

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