Hello all,
 
I have just come across the below email by Andreas Graning re the above subject, and its just exactly the same problem I am currently faced with. In this respect, I was wondering if anyone has in the meantime come across a solution to the issue. 
 
Any help on this subject shall be greatly appreciated.
 
 
Thankssssssssssss.
 
 
Regards,
 
 
Karl 

[cisco-voip] No early media for ISDN->SIP->ISDN

Andreas Granig a.granig at inode.at
Wed May 12 09:52:19 EDT 2004


Hi,

I've a Cisco 5300 running firmware image c5300-js-mz.122-15.T12.bin that 
interacts with SipExpressRouter (SER) from iptel.org.

Following scenario: a caller from ISDN calls a number which is routed 
thru the C5300 to SER, where it is forwarded back to ISDN via the C5300 
again. In this, and only in this scenario, the caller doesn't hear any 
early media (no ringback, no announcements etc).

It works like charm when I for example call from a Cisco ATA to ISDN via 
C5300 and vice versa.

When I configure "voice call send-alert" at the C5300, the PROGRESS is 
converted to ALERT and I hear a ringback tone, but other early media 
like announcements are also overwritten by a ringback tone.
I've also tried "progress_ind setup enable 3", 8 for alert and 8 for 
proceed, without any success.

I've alread studied (hopefully) all available Cisco documentation (voice 
commands for Firmware 12.2-T, the "Interworking Signaling Enhancements 
for H.323 and SIP VoIP", the "PSTN Callers not Hearing any Ring Back 
When they Call IP Phones", "Troubleshooting No Ringback Tone on 
ISDN-VoIP (H.323) Calls" and so on).

Any hints?

Regards,
Andy

PS: I've attached the Q931-debug and here are also the relevant parts of 
the C5300 configuration:

voice call send-alert
!
voice-port 0:D
  input gain 2
  echo-cancel coverage 32
  echo-cancel suppressor
  timeouts interdigit 3
!
dial-peer voice 9 pots
  application session
  destination-pattern 0.
  no digit-strip
  direct-inward-dial
  port 0:D
!
dial-peer voice 99 voip
  destination-pattern [1-9]...T
  translate-outgoing calling 5
  translate-outgoing called 1
  voice-class codec 1
  session protocol sipv2
  session target dns:my.sipserver.com
  dtmf-relay h245-signal
  no call fallback
  no vad
!
gateway
  timer receive-rtcp 5
!
sip-ua
  retry invite 2
  retry response 2
  retry bye 2
  retry cancel 2
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