Mitel Forums - The Unofficial Source
Mitel Forums => Mitel MiVoice Business/MCD/3300 => Topic started by: steve75 on July 22, 2014, 04:51:58 AM
-
Hello everyone,
I need help on a problem on outgoing calls via SIP Trunk.
I created a sip trunk with 15 contemporary connecting Mitel 3300 MCD 4.2 with Asterisk system.
Everything works fine. But making an outgoing call the Mitel 3300 provides immediately CONNECT. This may be fine on outgoing calls manuals. On outbound calls made automatically by the system asterisk, if I have the CONNECT instant the system is not able to understand if the called number has answered the call or not.
Do you have any help that I can give? Need to change some parameter in the SIP Peer Profile for this type of problem?
Any help is appreciated!
Thank you
This is my SIP Peer Profile:
Calling Line ID Options
Default CPN
CPN Restriction No
Public Calling Party Number Passthrough No
Use Diverting Party Number as Calling Party Number No
Authentication Options
User Name
Password *******
Confirm Password *******
Authentication Option for Incoming Calls No Authentication
Subscription User Name
Subscription Password *******
Subscription Confirm Password *******
SDP Options
Allow Peer To Use Multiple Active M-Lines No
Allow Using UPDATE For Early Media Renegotiation No
Avoid Signaling Hold to the Peer No
Enable Mitel Proprietary SDP Yes
Force sending SDP in initial Invite message No
Force sending SDP in initial Invite - Early Answer No
Limit to one Offer/Answer per INVITE No
NAT Keepalive No
Prevent the Use of IP Address 0.0.0.0 in SDP Messages No
Renegotiate SDP To Enforce Symmetric Codec No
Repeat SDP Answer If Duplicate Offer Is Received No
RTP Packetization Rate Override No
RTP Packetization Rate 20ms
Special handling of Offers in 2XX responses (INVITE) No
Suppress Use of SDP Inactive Media Streams No
Signaling and Header Manipulation Options
Allow Display Update No
Build Contact Using Request URI Address No
De-register Using Contact Address not * No
Disable Reliable Provisional Responses Yes
Disable Use of User-Agent and Server Headers No
Enable sending '+' for E.164 numbers No
Force Max-Forward: 70 on Outgoing Calls No
Ignore Incoming Loose Routing Indication No
Use P-Asserted Identity Header No
Use P-Preferred Identity Header No
Use Restricted Character Set For Authentication No
Use To Address in From Header on Outgoing Calls No
Use user=phone No
Timers
Registration Period 3600
Registration Period Refresh (%) 50
Session Timer 90
Subscription Period 3600
Subscription Period Minimum 300
Subscription Period Refresh (%) 80
Key Press Event Options
Allow Inc Subscriptions for Local Digit Monitoring No
Allow Out Subscriptions for Remote Digit Monitoring No
Force Out Subscriptions for Remote Digit Monitoring No
Request Outbound Proxy to Handle Out Subscriptions No
KPML Transport default
KPML Port 0
-
My guess is a timer change would tweak this, but not sure which one. See what "Help" gives you, or hover the mouse pointer at the timers and look at the info. If someone else knows more, please chime in. Thanks.
-
Huh, what are the chances.. I was going to post about the same problem in a different scenario. :o
I have a MCD 6.0 SP2 system sip trunked to a 3rd party IVR app. The IVR wants to make outbound calls to customers and give them information they requested. Obviously, it needs to know when the B party answers. I'm seeing the same thing as steve75 - for some unknown reason the Mitel is responding with 200 OK at about the time it receives a ring signal from the external telephone network. Basically, the IVR sends an INVITE, the Mitel replies with 100 Trying, then a few seconds later it sends 183 Session Progress, quickly followed by 200 OK. If we test to an extension, it works perfectly. The outbound trunks are E1 ISDN.
I've been all over my config, and have read a bunch of the sample configs for different providers on MOL, but I'm still in the dark. The only thing that looks like what I want is in Digital CO Trunk Circuit Descriptors there is a setting for Fake Answer Supervision After Outpulsing, but it doesn't apply to my trunk types as far as I can tell. It's like that setting is Yes on E1 trunks.
jrg0852 - I don't think this is timer related as I see it at random times roughly 3 to 6 seconds after INVITE.
-
Is there a way to tell from CCS traces if the answer supervision is coming from the E1?
Ralph
-
I've been in touch with the amazing guys at L2 Support. They have suggested I enable "Suppress Simulated CCM after ISDN Progress" in CoS for anything in the path. The online help for that feature is instructive, but has a couple of warnings that are giving me pause. I'm just going through change management with the customer and will update the thread with how things turn out - probably sometime next week.
-
Earlier update than expected - this new SIP trunk is not in production so we gave it its own COS and gave it a try there only. The IVR dev says "Works a treat". :)
-
Good to know,
Thanks for the update
-
They have suggested I enable "Suppress Simulated CCM after ISDN Progress" in CoS for anything in the path. The online help for that feature is instructive, but has a couple of warnings that are giving me pause.
Ah! I remember this now. The issues I've experienced with this enabled were that there were some auto attendants i couldn't dial through. It wouldn't hear the touch tones. You'd call an airline, it would answer with an auto attendant but you couldn't do anything.
With this enabled you won't be able to use touch tones until you get answer supervision from the far end. In some cases it never comes through.
Ralph
-
Thanks, ralph. That puts some meat onto the warnings in help. I'll pass it on to the devs to be sure to use a broad range of test numbers during rollout.
Hopefully steve75 will let us know if it's helped his issue, too.