Call Pickup sip trunk no audio

[Anju Josua]

Hi All,

currently deploying cme (aslo act as cube) with sip phone then connect to itsp sip trunk.  topology as below:

ITSP --- (sip trunk) --- CME (CUBE) ---(register using sip)--- ip phones

call from itsp to ip phone and ip phone to itsp already running well.

problem with pickup feature from itsp.

if i call from internal extension to other extension then pickup running well.

if i call from itsp to internal extension (already set up DID) then pickup from other ip phone (within same pickup group) then call connected but no audio on both way. call still connect until one of party hang up the call.

already do some test, i call from itsp to internal extension, then pickup from other ip phone, call connect with no audio, then i "hold" the call from ip phone, then i resume the call, call connect with both audio.

someone facing same issue with me in this link https://community.cisco.com/t5/ip-telephony-and-phones/cube-no-rtp-after-pickup/td-p/3714079 . already follow his guide to block the sip updates message, already try config in "voice service voip" and "dial-peer" but somehow cube keep send update message to itsp.

also already try to block with the midcall signaling, using this link https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-midcall-reinvite.pdf , but still no luck.

is there anyone out there facing same issue? any suggestion will be appreciated. need help on this.

thanks.

Anju Josua

[mHadi]

hi dear friend,

a few days ago, i had this problem with a cisco 3845 router with a SIP-TRUNK line from telco

when we pickup a call which was ringing from telco, we have no audio at all..

i view this topic:

https://community.cisco.com/t5/ip-telephony-and-phones/cube-no-rtp-after-pickup/td-p/3714079

>> and FIX the problem with this config on our cisco 3845 router:

### Remove the "UPDATE" Method Support to Avoid Interoperability Issues ###

config terminal
voice class sip-profiles 200
 request ANY sip-header Allow-Header modify ", UPDATE" ""

### Then apply it on dial peer to CUCM ###

dial-peer voice 150 voip
destination-pattern 67301
session protocol sipv2
session target ipv4:192.168.30.2
voice-class sip profiles 200
dtmf-relay rtp-nte
codec g711ulaw
no vad