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CME voip dial peer call issue

Mohammed Al-odhari
Enthusiast
Enthusiast

Dear all,

i have a cme routers in 4 branches using site codes as they have common extension numbers.

Attached is the show run and debug commands for HQ branch in which a fast busy tone (unknown number) is shown when i call the branches using :

011123, 008111 or 004123 in which the first three digits is the site code.

other branches can call the HQ successfully, routing and firewall is well configured.

can you please go through the attached and help.

thanks for all. 

5 Replies 5

Nadeem Ahmed
Cisco Employee
Cisco Employee

What is the Inter-Sites codec being set to  from HQ---Branches.

 

On HQ router you have dial-peer with voice class codec

Disconnect cause which in the logs is 17,57 which is probable cause of codec mismatch.

 

If you try testing one HQ to one branch call with G711 call with hard code on dial-peer .

 

Regards

Nadeem Ahmed

Br, Nadeem Please rate all useful post.

Nadeem is likely on to something here with the codec issue.

Your outbound dial peer is 9, which defines the SIP codec but uses H323 methods for DTMF. Try changing this on both ends of the call to use SIP with rtp-nte, or H323 with h245-alpha. 

dial-peer voice 9 voip
 translation-profile outgoing To-DhAMAR
 destination-pattern 011...
 session protocol sipv2
 session target ipv4:10.100.11.20
 voice-class codec 5  
 dtmf-relay h245-alphanumeric h245-signal
 no vad

     

     If you're using SIP, use these debugs and the issue should be much easier to identify:

    debug ccsip messages
    debug ccsip error
    debug ccsip call
    debug voip ccapi

     

    Run the debugs on both ends if you manage both devices. And make sure the CME/CUBE on the receiving end is set to allow the codecs at the inbound dial-peer and, if applicable, with the "allow-connections" commands in 'voice service voip' params.

     

    If you're still hitting issues, paste your debugs.

    Hi Nick and thanks for replay,

    the other 3 branches uses sip & sccp phones and uses the below dial peer to dial voip call.

    Branch commands:

    !
    voice class codec 5
     codec preference 1 g711ulaw
     codec preference 2 g729r8
    !

    for sip phone:

    !
    voice register pool  32
     id mac 6C99.8985.1DBA
     type 3905
     number 1 dn 32
     template 1
     dtmf-relay rtp-nte sip-notify
     voice-class codec 5
     username SFD32 password SFD32
    !

    for voip dial peer:

    !
    dial-peer voice 91 voip
     translation-profile outgoing To-HQ
     destination-pattern 001...
     session protocol sipv2
     session target ipv4:10.100.1.20
     voice-class codec 5  
     dtmf-relay h245-alphanumeric h245-signal
     no vad
    !

     

    thanks nadeem,

    i`ll change the codec and i`ll inform you back.

     

    regards,

    Hi,

    yes there is a voice class command in the dial peer.

    !
    voice class codec 5
     codec preference 1 g711ulaw
     codec preference 2 g729r8
    !

    HQ dial peer

    dial-peer voice 9 voip
     translation-profile outgoing To-DhAMAR
     destination-pattern 011...
     session protocol sipv2
     session target ipv4:10.100.11.20
     voice-class codec 5
     dtmf-relay sip-notify rtp-nte
     no vad

    Branch dial peer

    dial-peer voice 92 voip
     translation-profile outgoing To-HQ
     destination-pattern 001...
     session protocol sipv2
     session target ipv4:10.100.1.20
     voice-class codec 5
     dtmf-relay h245-alphanumeric h245-signal
     no vad

     

    even i hard code the codec with G711 but still the issue there.

     

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