Caller ID issue on SIP Trunks

[Xero]

I am facing a problem while configuring SIP trunks on OXE R9.0, my trunks are working properly and I have configured DID/DOD also; everything is working fine excpet the 3 issues mentioned below:

1) Caller ID is not displayed on the set whereas it is being displayed in the SIP traces.

2) I am unable to send DTMF digits using SIP trunks where as they are being transmitted using analog trunks; what is the protocol that DTMF is using? can we change the protocol to RTP?

3) Whenever I reset SIP motor all 4 process donot initialize and trunk group do not come in service, for that I have to perform bascul everytime.

[Konstantinos.E]

hi

1)make an a different entity with the supllement and instalation number and declare it to the sip trunk group.
2)compvisu sys command shows if the rtp is enabled. by default in r9 is enabled.
3)what command do you use to reset the process?

[mujib.alu]

HI,

1. Translator / Extn num Plan / Ext.Call translator : DEF
2. SIP / External GW : Dynamic payload : ???
3. kill the process

B O L

[Xero]

Thanks both of u...

1) CLI issue resolved by managing ext call back translation ... I forgot
2) Dynamic payload was 97 and not working I changed it to 101 still not working..
3) I use dhs3_init -R SIPMOTOR but it make the TG out of service and never make it inservice again.

My main issue left is for the dtmf digits sending

[Xero]

Inter-node protocol H323....... no
RTP Direct..................... yes
RTP Direct for H323 terminals.. no
! Be careful : Inter-node protocol H323 must be set to yes !!!
Fast Start..................... yes
VAD (Voice Activity Detection):
- G723/G729...... no
- G711........... no
ECE (Echo Canceller)........... yes
- LIO/LIOE........ 16 ms
- INTIP/GA/GD..... 128 ms
PFE (Post Filter).............. yes
Volume ........................ 8
Volume for IP Phone ........... 0dB
Volume for other device. ...... 0dB
VRE ........................... no
Law (Except Media Gateway)..... A law
Global compression type ....... G723
Multi-algorithm (for H323) .... no
Compression for INTIP/GD ..... with
Compression for IPP ........... with
Transit on IP Boards ...........no
ticket Stat IP................. no
IP version..................... IPv4
Transit compatibility.......... no
Voip Framing G711 ............. 20 ms
Voip Framing G723 ............. 30 ms
Voip Framing G729 ............. 20 ms
No RBT For Direct RTP H323..... no
T38 FAX........................ no

[mujib.alu]

Hi,

DTMF digit sending problem from analog user or digital user or IP users..?

if analog users able send DTMF digits, check with SIP service provider for the Payload value.!!