No ringing on outbound SIP trunk but audio when answered

[G0G3tt3r]

We added two more sip channels to the system here. Same trunk we were using just 2 more paths. 9.1.10 500v2. Everything works great except, after the call is made there is no ringing sound on the IPO but audio cuts through fine once the called parts answers. Internal ringing works fine. Sip provider has don their checks..... We have rebooted router and PBX

 

[G0G3tt3r]

Actually I cant even park or put a call on hold until it is answered. Usually you can wait until the call starts to ring and then you get the chance to place the call on hold or park even before it is answered...Right??

 

[intrigrant]

Enable 'FORCE DIRECT EARLY MEDIA' on the SIP trunk and try again.

 

[G0G3tt3r]

Greyed out. I tried to figure out why but am stuck.

 

[G0G3tt3r]

1536067mS CMExtnEvt: v=(null) State, new=Idle old=Idle,0,0,217: Digit Key Pressed 4
10:53:19 1536068mS CMExtnRx: v=217, p1=0
CMInformation
Line: type=DigitalExtn 4 Call: lid=0 id=1120 in=0
Called[4] Type=Default (100) Reason=CMDRdirect Keypad[4]
10:53:19 1536068mS CMARS: FORM: Main - Received Number: 2703314924
10:53:19 1536068mS CMARS: FOUND A SHORT CODE - short_code: XXXXXXXXXXN - Tel: N - Called_Party: 2703314924 - Line Group Id: 7
10:53:19 1536068mS CMARS: SetCurrentTarget: Short_Code: XXXXXXXXXXN - Line_Group_ID: 7
10:53:19 1536068mS CMARS: FindActiveARSByGroupID GroupID=7 - Not Found
10:53:19 1536069mS CMLRQ: FindActiveLRQByGroupID GroupID=7 - Not Found
10:53:19 1536069mS CMMap: a=4.20 b=0.0 Mapper::AllocateCodec allocated CMRTVocoder(f560536c) resource busy 3, total 10
10:53:19 1536070mS CMCallEvt: 0000000000000000 0.1123.0 -1 BaseEP: NEW CMEndpoint f180be7c TOTAL NOW=9 CALL_LIST=4
10:53:19 1536071mS NAT: Info: Line 17: Not using STUN for media in this case.
10:53:19 1536073mS CMARS: ModifyCMARSTarget: Short_Code: XXXXXXXXXXN - Line_Group_ID: 7 set line status to CMARS_TRYING
10:53:19 1536074mS CMARS: FOUND LINE - Line Id: 17 - using line group id: 7 (code XXXXXXXXXXN, line 7) - Called Number: 2703314924 - Calling Number: 217
10:53:19 1536074mS CMARS: SEND Setup TO LINE
10:53:19 1536074mS CMCallEvt: c0a8061e00000463 17.1123.0 27 SIPTrunk Endpoint: StateChange: END=child CMCSIdle->CMCSOffering
10:53:19 1536076mS CMLineTx: v=0
CMSetup
Line: type=SIPLine 17 Call: lid=17 id=1123 in=0
Called[2703314924] Type=Default (100) Reason=CMDRdirect Calling[217] Type=Internal Plan=Default
BC: CMTC=3K1Audio CMTM=CallIndSigCon CMTR=64 CMST=Default CMU1=ULaw
IE CMIESupplementaryService (3)
Network Facility Extension
sourceEntity: 0
destinationEntity: 0
Interpretation APDU
discardAnyUnrecognisedInvokePdu
CallingName.Invoke.CodePageISO8859-1
invokeId 14649
user 'Charts' presentation Allowed
IE CMIEFastStartInfoData (6) 14 item(s)
IE CMIECallingPartyName (110)(Type=CMNameDefault) name=Charts
IE CMIECallingPartyKName (226)(Type=CMNameDefault) name=Charts
IE CMIERespondingPartyName (228)(Type=CMNameDefault) name=Charts
IE CMIERespondingPartyKName (229)(Type=CMNameDefault) name=Charts
IE CMIERespondingPartyNumber (230)(P:100 S:100 T:101 N:100 R:4) number=217
IE CMIEDeviceDetail (231) c0a8061e00000460 LOCALE=enu HW=15 VER=9 class=CMDeviceStdPhone type=108 number=49 channel=0 features=0x10000420 rx_gain=32 tx_gain=32
ep_callid=1120 ipaddr=192.168.6.30 apps=0 loc=0 em_a_loc=1 em_d_loc=0 features2=0x0 is_spcall=0 ignores_dtmf=0 avgsid=
IE CMIEMohSourceId (247) MOH Source = 255
IE CMIESecurityData (253) 2 item(s)
CMSecurityElement RX_ONLY {
CMSRTPCryptoCapability none
Codecs list (size 7)
}
CMSecurityElement TX_ONLY {
CMSRTPCryptoCapability none
Codecs list (size 7)
}
Locale: enu
10:53:19 1536078mS SIP Call Tx: 17
INVITE sip:270XXXXXXX@ XX.XX.XX.XXSIP/2.0
Via: SIP/2.0/UDP 192.168.6.30:5060;rport;branch=z9hG4bKe4fa71990c969b285e8d409acd534007
From: "xxxxxxxxxx" <sip:xxxxxxxxxx@XX.xx.xx.xx>;tag=7af8261827ef7385
To: <sip:xxxxxxxxxx@xx.xx.xx.xx>
Call-ID: e0ded2df177b7809dee047394317a09e
CSeq: 2023909805 INVITE

 

[G0G3tt3r]

[Force direct media] i turned it on and no change

 

[janni78]

You're not posting the complete trace.

"Trying is the first step to failure..." - Homer

 

[janni78]

Oh, and don't post it, include it as a txt file in that case.

"Trying is the first step to failure..." - Homer

 

[G0G3tt3r]

trace uploaded. call originated from ext 217

 

[kyle555]

So the flow is invite, 100 trying, and then 183 session progress or 180 ringing.

I can't remember, but there's a convention where I believe when I send you a 183 means "I'm ringing the far end. I'm not playing ringback to you. If you want ringback on your set, play it yourself" and and I send you 180 Ringing, I set up early media and send you the media of ringback.

It's not against RFC spec to provide or not provide ringback, but I'd be curious what you get from the far end after 100 trying and before 200ok.

 

[G0G3tt3r]

I acually dont see it as an audio issue now. It acts like the call isn't terminating until the caller answers. Not even a call timer counting.

 

[G0G3tt3r]

kyle555 - What?