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Replies

FAX not Working between CUCM to Voice gateway VG2911 using SIP confi

chaitusite
Level 1
Level 1

FAX not Working Between CUCM to VG2911 via SIP Trunk,

incoming Call flow: PSTN--->SIP trunk CUBE4331--->CUCM--->SIP trunk VG2911--->FXS Port --->FAX machine

outgoing call flow :FXS port--->SIP trunk(Dial Peer on 2911)--->CUCM----SIP trunk to CUBE4331--->PSTN SIP trunk

 

Tested Incoming Voice call Working but not able to Dial Out from the FAX FXS line

 

Please suggest Dial-Peer and FXS port Config as below.

 

dial-peer voice 119 pots
description *** HR FAX ***
destination-pattern 119
port 0/1/1
no sip-register

!

!

voice-port 0/1/1
description HR Fax

!

!

voice service voip
ip address trusted list
address-hiding
dtmf-interworking rtp-nte
mode border-element license capacity 30
allow-connections sip to sip
no supplementary-service sip moved-temporarily
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
sip
asserted-id pai
midcall-signaling passthru
privacy-policy passthru

1 Accepted Solution

Accepted Solutions

HI Roger,

 

After your Analysis i have found the issue is in CUCM not in VG2911, when i carefully examine the SIP trunk, i found the issue is with CSS assigned in SIP trunk is not having Correct Partitions to route the call Via CUBE4311, I have added the correct CSS and the calls start working, but i see the Media is not two way then i enable MTP on SIP trunk which did the trick and resolved the issue.

 

Thanks Roger for guiding me to correct path,

Krishna

View solution in original post

10 Replies 10

If that's all the dial peer that you have it's not that surprising that it does not work as there is no outbound dial peer to route the call from the 2911 to CM. Would you mind to post the entire configuration of all the gateways that are involved in the specific call scenario?



Response Signature


chaitusite
Level 1
Level 1

HI Roger ,

 

Please find the attached configuration 

 

I have remove some Config 

From what I can tell it includes the SBC configuration, but not the VG. Without this it would be hard to give you any valuable advice.



Response Signature


HI Roger,

 

Sorry i have missed the VG file, Please find the attached ZIP which is having both the documents

From what I can tell your sending calls from the VG to CM based on this configuration.

voice class e164-pattern-map 20
 description *** PSTN Numbers ***
  e164 +.T
!
dial-peer voice 501 voip
 description *** Outbound from CUBE to CUCM ***
 session protocol sipv2
 session target ipv4:XXXXXCUCM:5060
 session transport udp
 destination e164-pattern-map 20
voice-class codec 1 
 no voice-class sip outbound-proxy   
 voice-class sip early-offer forced
 voice-class sip profiles 3
 voice-class sip bind control source-interface GigabitEthernet0/1
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte sip-kpml
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 no vad
!
dial-peer voice 200 voip
 destination-pattern [1234]..$
 session protocol sipv2
 session target ipv4:XXXXXCUCM:5060
 session transport udp
 voice-class codec 1 
 no voice-class sip outbound-proxy   
 voice-class sip early-offer forced
 voice-class sip profiles 3
 voice-class sip bind control source-interface GigabitEthernet0/1
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte

Are you entering the called number on the fax to match either +T or [1234].. ?



Response Signature


Hi Roger,

 

Yes the call has to go to  VG2911---> CUCM-----> 4331 CUBE----->PSTN, but this path was not Working , not able to dial out. 

The call path is clear, but that’s not really an answer to the question I had. Can you please share details on the called number entered into the fax machine? Reason for why asking for this information is to see if it would match any of your dial peers in the VG that sends calls to the CM.



Response Signature


HI Roger,

 

After your Analysis i have found the issue is in CUCM not in VG2911, when i carefully examine the SIP trunk, i found the issue is with CSS assigned in SIP trunk is not having Correct Partitions to route the call Via CUBE4311, I have added the correct CSS and the calls start working, but i see the Media is not two way then i enable MTP on SIP trunk which did the trick and resolved the issue.

 

Thanks Roger for guiding me to correct path,

Krishna

Is it FAX not working or Voice call not working ? 

 

 



Response Signature


HI Nithin,

Incoming call to the Line is working fine , FAX and Outgoing calls are not working on the line 

 

Thanks,

Krishna