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raymond

2005-Apr-26 00:22 UTC

[Asterisk-Users] NO ringback tone for VOIP call to another SIP server

All,

I found that there is no ringback to the caller (a-party) for VoIP call but when
I make call to registered user, I can hear the ringback tone.
Below are the debug log for the two cases:  

I wonder if anyone who can tell me why?

Thanks.

Raymond

Case 1:  no ringback to the caller (a-party) for outbond VoIP call to another
SIP server

Apr 26 07:04:09 VERBOSE[2607]:     -- Executing
Dial("SIP/30511694-abfa",
"SIP/99740185293137656@192.168.11.194") in new stack
Apr 26 07:04:09 DEBUG[2607]: Outgoing Call for 99740185293137656
Apr 26 07:04:09 DEBUG[2607]: 99740185293137656 is not a local user
Apr 26 07:04:09 VERBOSE[2607]:     -- Called 99740185293137656@192.168.11.194
Apr 26 07:04:09 DEBUG[2607]: (Provisional) Stopping retransmission (but
retaining packet) on '08dd46db0a283b54514fb22a2c1d1cf5@192.168.19.244'
Request 102: Found
Apr 26 07:04:13 DEBUG[2607]: (Provisional) Stopping retransmission (but
retaining packet) on '08dd46db0a283b54514fb22a2c1d1cf5@192.168.19.244'
Request 102: Found
Apr 26 07:04:13 VERBOSE[2607]:     -- SIP/192.168.11.194-8dc7 is making progress
passing it to SIP/30511694-abfa
Apr 26 07:04:13 DEBUG[2607]: Ooh, format changed from unknown to ulaw
Apr 26 07:04:13 DEBUG[2607]: Auto destroying call
'000f905a-df770678-288bbbfc-04cbc6ba@202.69.89.241'
Apr 26 07:04:13 DEBUG[2607]: RTP NAT: Using address 192.168.19.241:64868
Apr 26 07:04:13 DEBUG[2607]: Oooh, format changed to 8
Apr 26 07:04:13 DEBUG[2607]: Ooh, format changed from unknown to ulaw
Apr 26 07:04:13 DEBUG[2607]: Ooh, format changed from ulaw to alaw
Apr 26 07:04:15 NOTICE[2607]: RFC3389 support incomplete.  Turn off on client if
possible
Apr 26 07:04:32 DEBUG[2607]: update_user_counter(99740185293137656) - decrement
outUse counter
Apr 26 07:04:32 DEBUG[2607]: 99740185293137656 is not a local user
Apr 26 07:04:32 DEBUG[2607]: Exiting with DIALSTATUS=CANCEL.
Apr 26 07:04:32 VERBOSE[2607]:   == Spawn extension (siptest02, 85293137656, 1)
exited non-zero on 'SIP/30511694-abfa'
Apr 26 07:04:32 VERBOSE[2607]:     -- Executing
Hangup("SIP/30511694-abfa", "") in new stack
Apr 26 07:04:32 VERBOSE[2607]:   == Spawn extension (siptest02, h, 1) exited
non-zero on 'SIP/30511694-abfa'
Apr 26 07:04:32 DEBUG[2607]: cdr_mysql: inserting a CDR record.
Apr 26 07:04:32 DEBUG[2607]: cdr_mysql: SQL command as follows:  INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2005-04-26 07:04:09','\"cisco 7960\"
<30511694>','30511694','85293137656','siptest02',
'SIP/30511694-abfa','SIP/192.168.11.194-8dc7','Hangup','',23,0,'NO
ANSWER',3,'')
Apr 26 07:04:32 DEBUG[2607]: update_user_counter(30511694) - decrement inUse
counter
Apr 26 07:04:32 DEBUG[2607]: Acked pending invite 102
Apr 26 07:04:32 DEBUG[2607]: Stopping retransmission on
'08dd46db0a283b54514fb22a2c1d1cf5@192.168.19.244' of Request 102: Found
Apr 26 07:04:32 DEBUG[2607]: Stopping retransmission on
'08dd46db0a283b54514fb22a2c1d1cf5@192.168.19.244' of Request 102: Found
Apr 26 07:04:32 DEBUG[2607]: 99740185293137656 is not a local user
Apr 26 07:04:32 DEBUG[2607]: Stopping retransmission on
'000f905a-df770679-53fe2af6-57af4e6e@192.168.19.241' of Response 102:
Found


Case 2: When I make call to registered user, I can hear the ringback tone:

Apr 26 07:05:49 DEBUG[2607]: Auto destroying call
'E489B82D0E814906923363BF5F11C30A@192.168.19.244'
Apr 26 07:05:50 DEBUG[2607]: Setting NAT on RTP to 4
Apr 26 07:05:50 DEBUG[2607]: Stopping retransmission on
'000f905a-df77067a-11b43352-391ee0db@192.168.19.241' of Response 101:
Found
Apr 26 07:05:50 DEBUG[2607]: Setting NAT on RTP to 4
Apr 26 07:05:50 DEBUG[2607]: Check for res for 30511694
Apr 26 07:05:50 DEBUG[2607]: Call from user '30511694' is 1 out of 0
Apr 26 07:05:50 DEBUG[2607]: build_route: Contact hop:
<sip:30511694@192.168.19.241:5060>
Apr 26 07:05:50 VERBOSE[2607]:     -- Executing
Dial("SIP/30511694-581e", "SIP/30511690|20|tr") in new stack
Apr 26 07:05:50 DEBUG[2607]: SIMPLE DIAL (NO URL)
Apr 26 07:05:50 DEBUG[2607]: Setting NAT on RTP to 4
Apr 26 07:05:50 DEBUG[2607]: Outgoing Call for 30511690
Apr 26 07:05:50 DEBUG[2607]: Call from user '30511690' is 1 out of 0
Apr 26 07:05:50 VERBOSE[2607]:     -- Called 30511690
Apr 26 07:05:50 DEBUG[2607]: (Provisional) Stopping retransmission (but
retaining packet) on '463a6a3847d25dbd0d990f3b5e6635e3@192.168.19.244'
Request 102: Found
Apr 26 07:05:50 DEBUG[2607]: (Provisional) Stopping retransmission (but
retaining packet) on '463a6a3847d25dbd0d990f3b5e6635e3@192.168.19.244'
Request 102: Found
Apr 26 07:05:50 VERBOSE[2607]:     -- SIP/30511690-adb1 is ringing
Apr 26 07:06:00 DEBUG[2607]: update_user_counter(30511690) - decrement outUse
counter
Apr 26 07:06:00 DEBUG[2607]: Exiting with DIALSTATUS=CANCEL.
Apr 26 07:06:00 VERBOSE[2607]:   == Spawn extension (siptest02, 1690, 1) exited
non-zero on 'SIP/30511694-581e'
Apr 26 07:06:00 VERBOSE[2607]:     -- Executing
Hangup("SIP/30511694-581e", "") in new stack
Apr 26 07:06:00 VERBOSE[2607]:   == Spawn extension (siptest02, h, 1) exited
non-zero on 'SIP/30511694-581e'
Apr 26 07:06:00 DEBUG[2607]: cdr_mysql: inserting a CDR record.
Apr 26 07:06:00 DEBUG[2607]: cdr_mysql: SQL command as follows:  INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2005-04-26 07:05:50','\"cisco 7960\"
<30511694>','30511694','1690','siptest02',
'SIP/30511694-581e','SIP/30511690-adb1','Hangup','',10,0,'NO
ANSWER',3,'')
Apr 26 07:06:00 DEBUG[2607]: update_user_counter(30511694) - decrement inUse
counter
Apr 26 07:06:00 DEBUG[2607]: Acked pending invite 102
Apr 26 07:06:00 DEBUG[2607]: Stopping retransmission on
'463a6a3847d25dbd0d990f3b5e6635e3@192.168.19.244' of Request 102: Found
Apr 26 07:06:00 DEBUG[2607]: Stopping retransmission on
'463a6a3847d25dbd0d990f3b5e6635e3@192.168.19.244' of Request 102: Found
Apr 26 07:06:00 DEBUG[2607]: Stopping retransmission on
'000f905a-df77067a-11b43352-391ee0db@192.168.19.241' of Response 102:
Found



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raymond

2005-Apr-26 00:47 UTC

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[Asterisk-Users] Re: NO ringback tone for VOIP call to another SIP server

Hi all,

To my surprise, I change the Dial statement in extensions.conf from:

exten => _852.,1,Dial,SIP/123456${EXTEN}@192.168.11.194,r

to:

exten => _852.,1,Dial(SIP/123456${EXTEN}@192.168.11.194,20,r)

I can hear ringback tone now.  I don't know why but it just works.

Cheers.

Raymond

  ----- Original Message ----- 
  From: raymond 
  To: asterisk-users@lists.digium.com 
  Sent: Tuesday, April 26, 2005 3:22 PM
  Subject: NO ringback tone for VOIP call to another SIP server


  All,

  I found that there is no ringback to the caller (a-party) for VoIP call but
when I make call to registered user, I can hear the ringback tone.
  Below are the debug log for the two cases:  

  I wonder if anyone who can tell me why?

  Thanks.

  Raymond


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asterisk users - Apr 2005 - NO ringback tone for VOIP call to another SIP server

[Asterisk-Users] NO ringback tone for VOIP call to another SIP server

[Asterisk-Users] Re: NO ringback tone for VOIP call to another SIP server