10-27-2014 09:02 PM - edited 03-17-2019 12:42 AM
Hello Pros,
Incoming and outgoing PSTN calls work with no issue in SRST. The only problem is that as soon as we try to put an incoming PSTN call on hold the call drops. Please note that this is not the case for outgoing calls. An outgoing call can be put on hold either from calling phone or called phone without issue. Both Transfer and Conference soft keys work in both directions. I am attaching the show running configuration.
10-28-2014 05:23 AM
Try adding the following command "voice iec syslog" globally. This command will display any errors that may be occuring. If not we will need to collect some debugs.
Regards,
Yosh
10-28-2014 11:05 AM
Hi,
voice iec syslog didn't gererate any output on the screen
Here's the debug ccsip all.
Also here's the output of the debug ccm-manager music-on-hold all
q3labvo10#debug ccm-manager music-on-hold all
Call Manager music-on-hold all debugging is on
q3labvo10#
q3labvo10#
q3labvo10#
q3labvo10#
q3labvo10#
q3labvo10#
q3labvo10#
Oct 28 13:57:24.103: moh_update_rtp: callID 162471 dstCallID -1
Oct 28 13:57:24.147: %ISDN-6-CONNECT: Interface Serial0/0/0:6 is now connected to 5146027093 N/A
Oct 28 13:57:24.643: moh_update_rtp: callID 162472 dstCallID -1
Oct 28 13:57:26.599: moh_update_rtp: callID 162472 dstCallID 162469
Oct 28 13:57:33.943: moh_update_rtp: callID 162472 dstCallID 162469
Oct 28 13:57:33.943: moh_update_rtp: callID 162472 dstCallID 162469
Oct 28 13:57:33.967: moh_update_rtp: callID 162472 dstCallID 162
Thanks,
MK
10-29-2014 02:34 AM
Please configure the ff and send debug ccsip messages only.
service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit
Then..
<Enable debugs, then test again.>
debug ccsip messages
<Enable session capture to txt file in terminal program.> (such as Putty)
then do the ff:
terminal length 0
show logging
10-30-2014 09:21 AM
10-30-2014 10:09 AM
Hi again,
I removed the "service survivability" command under the incoming dail-peer 1 and everything worked.
dial-peer voice 1 pots
translation-profile incoming 1514
service survivability -- Removed
incoming called-number .
direct-inward-dial
Thanks for you help Ayodeji,
MK
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