Hello Somphol/Justin,
I have resolved the issue by adding the command no supplementary-service
sip moved-temporarily.
Thanks a lot Somphol for pointing the document to me.
Thank you Justin for providing me the inputs.
Regards,
Viki
On Thu, Jan 30, 2014 at 6:15 AM, Justin Carney
Something doesn’t seem to add up in my head. Supp Services shouldn’t effect
DTMF. Did you change anything related to the SIP Trunk on CUCM? Or anything
DTMF related on a dial-peer?
On Jan 30, 2014, at 6:22 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote:
Hello Somphol/Justin,
I have
I understand how DTMF works on SIP Trunks, what I’m not clear on is how “no
supp services” would have an impact on his DTMF issue. I’m trying to understand
the logic of something changing with RFC2833 or SIP NOTIFY to the point where #
is now recognized, yet without changing anything related to
no supplementary service affect only call forwarding and call transfer , i do
not know how it solve DTMF
Regards,
Moataz Tolba
On Thursday, 30 January 2014, 15:17, Mark Holloway m...@markholloway.com
wrote:
I understand how DTMF works on SIP Trunks, what I’m not clear on is how “no
supp
Hello All,
I have attached the debug ccsip messages output before and after using the
command. I do not have the answer why it resolved the dtmf-issue. If you
guys find something, please share it.
Thanks,
Viki
On Thu, Jan 30, 2014 at 4:16 PM, Moataz moataz_m...@yahoo.com wrote:
no
In the larger debug attachment the SDP includes a=fmtp:18 in the 200 OK coming
from the CME site (IP 3.3.3.3). In the other capture I didn’t see any SDP. If
no DTMF offer is present during call setup, this would assume plain old in-band
DTMF, which won’t work on a compressed codec like G.729.
Hello All,
I am facing an issue with dtmf-relay. PhoneA registered to CUCM (SiteA)
calls PhoneD registered to CUCME (Site C). Between Regions G729 codec is
negotiated. PhoneD call-forward no answer to Voicemail. CUE integrated with
CME. After leaving the Voicemail from PhoneA to PhoneD, when I
On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman
sethuvign...@gmail.comwrote:
Media Termination Point Required (Checked)
MTP Preferred Originating CodecRequired Field: g711ulaw
Hi Vignesh,
I think if you can set these two to default settings which is MTP Required
[uncheck], and MTP
I concur with Somphol's suggestion and that mtp shouldn't be required.
You stated you can record the voicemail but I don't see the sdspfarm tag 1
BR2-IOS-XCODE command under telephony-service. Is your transcoder showing
its registered with show sccp command? I'm guessing that it is
registered
@onlinestudylist.com
Subject: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
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