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SIP Trunk to FXS no problem however once answered need to flash hook for transfer of call.

Mark Verwey
Level 1
Level 1

I have a site that does not have a PBX, because the calls are answered by an IVR, depending on the option the caller selects the call needs to be flash hook transfered.   When I had pots lines it works now that the site was converted to SIP, I need flash hook to work once again.  Below is part of the config on the router and the dial-peers for the lines assistance with figuring out what I missed would be greatly appreciated.  I already found in Cisco's documentation where it talks about hook flash and attempted to follow what it was saying to add the application service dsapp then made it global, then added the offer call hold settings under SIP-UA settings (see below) the document from Cisco is, https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-hookflash.pdf.  The router is running ios 15.6

 

application
service dsapp
param disc-toggle-time 15
param callHold True
param callWaiting TRUE
param callConference TRUE
!
global
service default dsapp

!!

dial-peer voice 10 pots
description store line 1
service dsapp
destination-pattern 22222222222
port 0/0/0
forward-digits 0

!!

dial-peer voice 1000 voip
description Outbound Store line 1
service dsapp
session protocol sipv2
session target ipv4:152.23.3.235
session transport udp
incoming called-number 222222222
voice-class sip dtmf-relay force rtp-nte
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711ulaw
no vad

!!

sip-ua
credentials username
authentication username
retry invite 2
retry bye 2
retry cancel 2
retry register 10
registrar dns:bstnyc.granitevoip.com expires 3600
sip-server dns:bstnyc.granitevoip.com
offer call-hold conn-addr
offer call-hold

 

 

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