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New Member

SIP SRST

Hi,

I tried to configure srst for sip phones.

 

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
 modem passthrough nse codec g711ulaw
 sip
  registrar server expires max 600 min 60
!
voice class h323 1
  h225 timeout tcp establish 1
  h225 timeout setup 5
!
!
voice register global
 mode srst
 timeouts interdigit 5
 system message Mode Secours
 max-dn 144
 max-pool 42
 timezone 23
!
voice register pool  1
 translation-profile outgoing outgoing_calls
 id network 10.103.20.0 mask 255.255.255.0
 number 1 4436 preference 1
 preference 1
 proxy 10.103.20.11 preference 1
 alias 1 33389676968 to 4436 preference 1

Outgoing calls works fine, but incoming calls fails.

Dynamic dial-peer created (for the phone which I use to test) :

dial-peer voice 40007 voip
 preference 1
 destination-pattern 33389676968$
 redirect ip2ip
 session target ipv4:10.103.20.186:5060     ---> ip phone address
 session protocol sipv2
 dtmf-relay rtp-nte
  after-hours-exempt   FALSE

dial-peer voice 40018 voip
 preference 1
 destination-pattern 4436$
 redirect ip2ip
 session target ipv4:10.103.20.11:5060     ---> h323 gateway ip address
 session protocol sipv2
 dtmf-relay rtp-nte

Thanks for help

  • IP Telephony
10 REPLIES

Hello 1- Did you configure

Hello

Can you add the below:-

voice service voip

sip

  bind all  source-interface GigabitEthernet0/0

2- Can you enable "debug ccsip messages"?  for incoming calls .

 

Thanks

please rate all useful information

New Member

Hello 1) SRST reference is

Hello

 

1) SRST reference is configured on CUCM

2) I will do it

3) call-manager-fallback is not configured. But is this not only used in SCCP SRST registration.

 

Phones registers but incoming calls fails.

 

BR

HelloYes , just try as i

Hello

Yes , just try as i mentioned above + under voice register pool 1 " add codec g711ulaw , and dtmf-relay rtp-nte". If did not help , please share "debug CCSIP messages".

 

Thanks

please rate all useful information

New Member

Hello,Could the output

Hello,

Could the output attached help to solve my problem ?

 

Thanks

VIP Gold

Can you attach all dial-peers

Can you attach all dial-peers, as currently its taking 'Outgoing Dial-peer=40017` (not 40007) to complete the call and its disconnecting with reason code 1, that mean number is unassigned.

Suresh

New Member

HI Suresh, I think the

HI Suresh,

 

I think the problem is with my translations... I have more than 15 DDI and no common digits with the internal phone number, so all translations are made in the CUCM.

Is it always true that we cannot define more than 15 rules in a translation-rule ?

Or I have to find a way to convert all DDIs in internal phone number in the gateway.

I have a 2921 gateway with an 15.3 IOS.

Thanks for your help

Hello Look my friedn , based

Hello

 

Look my frienddn , based on your Debugs , your incoming calls hit dial-peer 50 , but you have error unallocated number which means that your destination is unreachable.Kindly share your inbound dial-peer , your translation rules , brief about your goal. Did you try bind under SIP?.

 

Thanks

please rate all useful information

New Member

Hello,Attached, you will find

Hello,

Attached, you will find the configuration of the voice gateway.

 

On CUCM, the gateway is configured as H323 gateway.

IP Phones are configured as SIP (7945G).

On voice ports, 389XXXXXX is translated to 33389XXXXXX.

33389XXXXXX is translated to DN in the CUCM.

With my configuration, in SRST mode, outgoing calls to PSTN work but not incoming calls from PSTN.

In SRST mode, all incoming calls should be directed to a specific phone.

BR

HelloCan you try undervoice

Hello

Can you try under

voice register global

dialplan-pattern 1 XXXXXXXXX extension-length  4 extension-pattern 4436

 

Thanks

please rate all useful information

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