Ring-Back Tone

Callers to my Freephoneline.ca trunk here a BEEP......BEEP instead of Ring........Ring.CallCentric trunk is fine.Using SIPDroid.

RE: Ring-Back Tone

Do you have access to the SIP credentials of your Freephoneline.ca account? Have you actually purchased their $50 configuration settings file, as mentioned on their F.A.Q.?

Have you tried provisioning these credentials on an IP-phone, ATA or soft-phone? Apparently, their service works properly with the Linksys PAP2 device, which they also sell.

RE: Ring-Back Tone

I have SIP credentials.Also with softphone it works fine. I have a HTC Dream running SIPDroid. Inbound calls to Callcentric trunk rings fine. Inbound calls to Freephoneline trunk give european type ringback tone.

I was wondering if there is something I can configure in my PBXes setup?

This post has been edited 1 time(s), it was last edited by Dia on 03.02.2010 at 20:38.

RE: Ring-Back Tone

So the issue at hand is, the type of ring-back your callers hear when calling your Freephoneline.ca DID form the PSTN, right?

RE: Ring-Back Tone

Zitat:

Originally posted by Diafora So the issue at hand is, the type of ring-back your callers hear when calling your Freephoneline.ca DID form the PSTN, right?

Yes .And from any other origin, SIP,cell phone,majicjack etc.

This post has been edited 1 time(s), it was last edited by fre on 03.02.2010 at 19:32.

RE: Ring-Back Tone

Then here is what you should look for in your System Log entries. After every "SIP response 100 to standard invite" there is either:

a "SIP response 180 to standard invite" or a "SIP response 183 to standard invite" which instructs the SIP UAs about the impending ring-back type.

In the case of a 180 the SIP UAs are instructed to create their own ring-back tones locally. If however a 183 is present, the ring-back audio is carried over from the remote end, on a one-way audio channel.

The actual SIP message is "183 Session Progress" and it includes an SDP specifically for the one-way audio channel. The 183 message originates from the PSTN gateway which initiated the call, and every B2BUA is obliged to pass it on to the SIP endpoint, so the "early media" can be carried over.

So if on an inbound call a 183 is sent instead of a 180 the PBXes SIP Proxy has to pass it along, therefore your endpoints end up hearing European ring-back instead of N. American.

Try a few calls and paste the relevant System Log entries to verify which particular message is present.

RE: Ring-Back Tone

Zitat:

Originally posted by Diafora Then here is what you should look for in your System Log entries. After every "SIP response 100 to standard invite" there is either: a "SIP response 180 to standard invite" or a "SIP response 183 to standard invite" which instructs the SIP UAs about the impending ring-back type. In the case of a 180 the SIP UAs are instructed to create their own ring-back tones locally. If however a 183 is present, the ring-back audio is carried over from the remote end, on a one-way audio channel. The actual SIP message is "183 Session Progress" and it includes an SDP specifically for the one-way audio channel. The 183 message originates from the PSTN gateway which initiated the call, and every B2BUA is obliged to pass it on to the SIP endpoint, so the "early media" can be carried over. So if on an inbound call a 183 is sent instead of a 180 the PBXes SIP Proxy has to pass it along, therefore your endpoints end up hearing European ring-back instead of N. American. Try a few calls and paste the relevant System Log entries to verify which particular message is present.

Does this tell us ?
Called freeline-100
Feb 3 16:31:24 VERBOSE[10252] chan_sip.c: SIP response 100 to standard invite
Feb 3 16:31:27 VERBOSE[10252] chan_sip.c: SIP response 180 to standard invite
Feb 3 16:31:27 VERBOSE[10252] logger.c: Found RTP audio format 3
Feb 3 16:31:27 VERBOSE[10252] logger.c: Found RTP video format 103
Feb 3 16:31:27 VERBOSE[10252] logger.c: Peer audio RTP is at port 192.168.1.100:21000
Feb 3 16:31:27 VERBOSE[10252] logger.c: Peer video RTP is at port 192.168.1.100:21070
Feb 3 16:31:27 VERBOSE[10252] logger.c: Found description format GSM
Feb 3 16:31:27 VERBOSE[10252] logger.c: Found description format h263-1998
Feb 3 16:31:27 VERBOSE[10252] logger.c: Capabilities: us - 0x18041e (gsm|ulaw|alaw|g726|ilbc|h263|h263p), peer - audio=0x2 (gsm)/video=0x100000 (h263p), combined - 0x100002 (gsm|h263p)
Feb 3 16:31:27 VERBOSE[10252] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Feb 3 16:31:27 VERBOSE[10252] chan_sip.c: Oooh, we need to change our formats since our peer supports only 0x100002 (gsm|h263p) and not 0x4 (ulaw)
Feb 3 16:31:27 VERBOSE[3115] logger.c: -- SIP/freeline-100-9d6d is ringing
Feb 3 16:31:27 VERBOSE[3115] logger.c: We're at 67.231.245.210 port 44028
Feb 3 16:31:27 VERBOSE[3115] logger.c: Video is at 67.231.245.210 port 46858
Feb 3 16:31:27 VERBOSE[3115] logger.c: Adding codec 0x4 (ulaw) to SDP
Feb 3 16:31:27 VERBOSE[3115] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Feb 3 16:31:43 VERBOSE[10252] chan_sip.c: SIP response 200 to standard invite
Feb 3 16:31:43 VERBOSE[10252] logger.c: Found RTP audio format 3
Feb 3 16:31:43 VERBOSE[10252] logger.c: Found RTP video format 103
Feb 3 16:31:43 VERBOSE[10252] logger.c: Peer audio RTP is at port 192.168.1.100:21000
Feb 3 16:31:43 VERBOSE[10252] logger.c: Peer video RTP is at port 192.168.1.100:21070
Feb 3 16:31:43 VERBOSE[10252] logger.c: Found description format GSM
Feb 3 16:31:43 VERBOSE[10252] logger.c: Found description format h263-1998
Feb 3 16:31:43 VERBOSE[10252] logger.c: Capabilities: us - 0x18041e (gsm|ulaw|alaw|g726|ilbc|h263|h263p), peer - audio=0x2 (gsm)/video=0x100000 (h263p), combined - 0x100002 (gsm|h263p)
Feb 3 16:31:43 VERBOSE[10252] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Feb 3 16:31:43 VERBOSE[3115] logger.c: -- SIP/freeline-100-9d6d answered SIP/16475473801-

The first one gives europian ring ,this one is North American ring.


Feb 4 08:29:03 VERBOSE[30275] logger.c: -- Called freeline-100
Feb 4 08:29:04 VERBOSE[10252] chan_sip.c: SIP response 100 to standard invite
Feb 4 08:29:09 VERBOSE[10252] chan_sip.c: SIP response 180 to standard invite
Feb 4 08:29:09 VERBOSE[10252] logger.c: Found RTP audio format 3
Feb 4 08:29:09 VERBOSE[10252] logger.c: Found RTP video format 103
Feb 4 08:29:09 VERBOSE[10252] logger.c: Peer audio RTP is at port 192.168.1.100:21000
Feb 4 08:29:09 VERBOSE[10252] logger.c: Peer video RTP is at port 192.168.1.100:21070
Feb 4 08:29:09 VERBOSE[10252] logger.c: Found description format GSM
Feb 4 08:29:09 VERBOSE[10252] logger.c: Found description format h263-1998
Feb 4 08:29:09 VERBOSE[10252] logger.c: Capabilities: us - 0x18041e (gsm|ulaw|alaw|g726|ilbc|h263|h263p), peer - audio=0x2 (gsm)/video=0x100000 (h263p), combined - 0x100002 (gsm|h263p)
Feb 4 08:29:09 VERBOSE[10252] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Feb 4 08:29:09 VERBOSE[10252] chan_sip.c: Oooh, we need to change our formats since our peer supports only 0x100002 (gsm|h263p) and not 0x4 (ulaw)
Feb 4 08:29:09 VERBOSE[30275] logger.c: -- SIP/freeline-100-1b3d is ringing
Feb 4 08:29:09 VERBOSE[30275] logger.c: We're at 67.231.245.210 port 44596
Feb 4 08:29:09 VERBOSE[30275] logger.c: Video is at 67.231.245.210 port 38528
Feb 4 08:29:09 VERBOSE[30275] logger.c: Adding codec 0x4 (ulaw) to SDP
Feb 4 08:29:09 VERBOSE[30275] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Feb 4 08:29:29 NOTICE[30275] rtp.c: Unknown RTP codec 126 received
Feb 4 08:29:29 VERBOSE[10252] chan_sip.c: SIP response 200 to standard invite
Feb 4 08:29:29 VERBOSE[10252] logger.c: Found RTP audio format 3
Feb 4 08:29:29 VERBOSE[10252] logger.c: Found RTP video format 103
Feb 4 08:29:29 VERBOSE[10252] logger.c: Peer audio RTP is at port 192.168.1.100:21000
Feb 4 08:29:29 VERBOSE[10252] logger.c: Peer video RTP is at port 192.168.1.100:21070
Feb 4 08:29:29 VERBOSE[10252] logger.c: Found description format GSM
Feb 4 08:29:29 VERBOSE[10252] logger.c: Found description format h263-1998
Feb 4 08:29:29 VERBOSE[10252] logger.c: Capabilities: us - 0x18041e (gsm|ulaw|alaw|g726|ilbc|h263|h263p), peer - audio=0x2 (gsm)/video=0x100000 (h263p), combined - 0x100002 (gsm|h263p)
Feb 4 08:29:29 VERBOSE[10252] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Feb 4 08:29:29 VERBOSE[30275] logger.c: -- SIP/freeline-100-1b3d answered SIP/5060-b7616258

This line looks to be the difference
Feb 4 08:29:29 NOTICE[30275] rtp.c: Unknown RTP codec 126 received

This post has been edited 9 time(s), it was last edited by fre on 07.02.2010 at 04:57.