[cisco-voip] No Ringing on Some outbound Calls - at&t ipFlex

Matthew Loraditch MLoraditch at heliontechnologies.com
Tue Jun 3 16:04:13 EDT 2014


Thanks everyone. I am somewhat familiar with all of this, but the refresher in the finer details helps.


Matthew G. Loraditch – CCNP-Voice, CCNA-R&S, CCDA
1965 Greenspring Drive
Timonium, MD 21093

direct voice. 443.541.1518
fax.  410.252.9284

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From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Meade
Sent: Tuesday, June 03, 2014 3:51 PM
To: Peter Slow
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No Ringing on Some outbound Calls - at&t ipFlex

Correct, it's the PortReq SCCP message and PortRes return message that is used for this.

On Tue, Jun 3, 2014 at 3:22 PM, Peter Slow <peter.slow at gmail.com<mailto:peter.slow at gmail.com>> wrote:
most skinny endpoints running newish software support early offer
using the sccp getport message, fyi.

On Tue, Jun 3, 2014 at 2:10 PM, Amit Kumar <amit3.kum at gmail.com<mailto:amit3.kum at gmail.com>> wrote:
> Here are my two cents.
>
> Skinny endpoint mostly do an delayed offer, unless we force on sip trunk to
> do an an early offer ( mtp if needed ). as everyone said, ringback behavior
> changes from call manager, as per what we get from called party.
>
> 180 ringing without SDP - > Ringback needs to be generated locally.
> 180 / 183 with SDP - > Called party is going to play ringback for us. We
> just need to establish media by that time ( Similar to what we see in ISDN,
> when we get alerting with an PI of 3 or 8 ). In case of delayed offer, PRACK
> is really an solution to look forward to have media established beforehand.
>
>
>
>
> On Tue, Jun 3, 2014 at 10:50 PM, Brian Meade <bmeade90 at vt.edu<mailto:bmeade90 at vt.edu>> wrote:
>>
>> Definitely at least need the "debug ccsip messages" for one of the calls.
>> We'll need to see if AT&T is sending a 180 Ringing or if they're sending a
>> 183 Session Progress w/ SDP to play the ringback inband.  If they're sending
>> a 183 Session Progress, make sure the SIP Profile on the SIP Trunk has the
>> Rel1XX Options set to Send Prack if 18X contains SDP.
>>
>>
>> On Tue, Jun 3, 2014 at 12:43 PM, Matthew Loraditch
>> <MLoraditch at heliontechnologies.com<mailto:MLoraditch at heliontechnologies.com>> wrote:
>>>
>>> We are having a regular but not always issue with a client, where they
>>> are not hearing the ringing when dialing outbound on calls. These are SCCP
>>> 6945s to UCM 9.1.2 SIP Trunked to CUBE and then handoff to at&t’s ipFlex
>>> service.
>>>
>>> If anyone has some suggestions to look at before I spend time with TAC it
>>> would be appreciated!
>>>
>>> Thanks!
>>>
>>>
>>>
>>>
>>>
>>> Matthew G. Loraditch – CCNP-Voice, CCNA-R&S, CCDA
>>>
>>> 1965 Greenspring Drive
>>> Timonium, MD 21093
>>>
>>> direct voice. 443.541.1518<tel:443.541.1518>
>>> fax.  410.252.9284<tel:410.252.9284>
>>>
>>> Twitter  |  Facebook  | Website  |  Email Support
>>>
>>> Support Phone. 410.252.8830<tel:410.252.8830>
>>>
>>>
>>>
>>>
>>>
>>>
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>>
>>
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