06-16-2012 09:07 AM - edited 03-16-2019 11:42 AM
Im using a SIP provider (voip.ms) to a H323 gateway (2801) and from there go to my CUCM 8.6 server. Inbound calls work fine, i get ringback and outbound calls connect and work fine, i just hear no ringback on my outbound calls via my SIP provider.
Attached are my debug ccsip messages, debug ccsip calls and debug ccapi inout as well as my running config
Here is also a link to another thread for other stuff ive tried:
http://www.techexams.net/forums/ccnp-voice/78494-outbound-ringback-issues.html
If anyone could solve this, you would be my hero!!!!
Solved! Go to Solution.
06-19-2012 01:01 PM
Yes. On cucm setup a sip trunk, the detsination address will be your cube gateway. Assign MRGL with ANN to the sip trunk.
2. On your gateway configure the ff:
voice servie voip
sip
bind control source-interface "xxxx"-----xxxx= your inerface eg gig0/0 of fa0/0
bind media source-interface "xxxx"
options-ping 60
early-offer forced.......................If you want to use early offer otherwise you can remove this
voice class codec 1---------------------------------to advertise different codecs, change preference as you want
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 1 voip---------------------------------This will be your default dial-peer for incoming calls from cucm and sip provider
session portocol sipv2
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
I believe you already have a dial-peer with sip protocl going to your ITSP.
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-16-2012 02:45 PM
Hi John,
By default, Cisco Unified Communications Manager will signal the calling phone to play local ringback if SDP is not received in the 180 or 183 response. If SDP is included in the 180 or 183 response, instead of playing ringback locally, Cisco Unified Communications Manager will connect media, and the calling phone will play whatever the called device is sending (such as ringback or busy signal). If you do not receive ringback, the device to which you are connecting may be including SDP in the 180 response, but it is not sending any media before the 200OK response. In this case
The log you sent does not contain the debug ccsip messages, so I am unable to determine if you receive 183 with SDP. If your provider is sending 183 without SDP, then cucm should be teliing the phone to play ringback locally. However CUCM uses Annunciator to play ringback for sip trunks. So you need to ensure that you have ANNunciator in your MRG and assign this MRG to a MRGL and inturn assign the MRGL to your phones and cube gateway
If you are receiving 183 with SDP but still not getting ringback, then try you can use a feature called disable early media to force cucm to play ringback locally..(dont forget to enable Annunciator and assign to phones and cube gateway)
disable-early-media 180
To specify which call treatment, early media or local ringback, is provided for 180 responses with 180
responses with Session Description Protocol (SDP), use the disable-early-media 180 command in
sip-ua configuration mode. To enable early media cut-through for 180 messages with SDP, use the no
form of this command. disable-early-media 180
config t
sip-ua
disable-early-media 180
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-16-2012 07:13 PM
06-16-2012 07:54 PM
OK i have ANN_2 and ANN_3 under Announciator, i put them in a group then put them in a list. On my gateway i applied the list and checked MTPR checkbox. I selected the Media Resource Group list under my phones and device pool. I also went to Service Parameters and set
Any more ideas or debugs you guys want?
06-16-2012 11:50 PM
You dont need to check MTP box. Please uncheck otherwise you may create additional problems..
Can you do a test call and send me a
show voip rtp connection (this will show us if the annunciator is been called upon to provide ringback)
NB: to see the annuciator in the voip rtp connection command to need to be refreshing the command. Dont just call and type the command once. Because the ann is called when its time to provide ringback.
also please send a full debug ccsip messages. The one you sent, the call did not complete
Do oyu know how to collect cucm traces? If you can send that will be great.
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-18-2012 12:47 AM
John,
Have you made any progress with this? Did you see my last post
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-18-2012 07:44 AM
Havent had a chance to run through this yet. Will tonight and post results that you requested. Im not too familiar with CUCM traces
06-13-2017 04:47 PM
Hi All,
Even i have the same issue . There will be one ring then then a fax tone then it will give a message" We are sorry your call cannot be completed this time, please hang up and dial after sometime" this message will play twice in quick session of two times and then the call will disconnect. The call is reaching the gateway .
Working calls are receiving 183 message with SDP but the failing calls from this location does not have 183 instead it have 180 without any SDP message in it and getting a 200 OK message soon.
Is it ok for us to involve ITSP.
Regards,
Shinaj cm
06-14-2017 08:26 AM
You should engage your ITSP and I hope it is not TATA. Provide traces to them and push them to look at why you are seeing this behaviour
06-14-2017 11:45 AM
Hi Ayodeji,
Customer is asking why it is service provider issue as all the numbers are going to the same provider and it is working.
this is occurring only for a set or block of DID.
Regards,
Shinaj cm
06-18-2012 04:03 PM
Here is the requested output. Anytime you see the output of show voip rtp connection it is when the outbound call should be providing ringback, and the number i am calling is ringing on the other end.
In the mean time i have also tried this:
System > Service Parameters > Under Clusterwide Parameters (Device - H323) set Check Progress Indicator Before Establishing Media to True (False as default) and setting Send H323 User Info Message to H225 Info for Call Progress Tone, restarted callmanager service, reset the gateway, all with no luck.
06-19-2012 01:51 AM
John,
Looking at your trace I dont see the communication with CUCM. I dont see your gateway receiving an invite from cucm or sending sip trying or 200 ok to cucm. If the call originated from cucm then this must happen.
Did you omit this traces?
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-19-2012 06:53 AM
No those were from the router and nothing was omitted. Were those suppose to come from CUCM traces?
06-19-2012 07:33 AM
John,
I know why I dont see the trace, its beacise the leg to cucm is h323 leg, its not sip.
May I ask, why are you not using SIP all the way.? i.e. configure your sip trunk from cucm to your gateway. Have you tried this.
Let me know if you want to try with the sip trunk setup. Otherwise please do another test call and send
debug voip ccapi inout.
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-19-2012 12:23 PM
I would try converting it to SIP the whole way. How do i go about configuring that in CUCM and on the router>?
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