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CS410 and Yealink T20P ring tone problem

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Hi there,

We just installed a brand new CS410 and 4 Yealink T20P handset at a client site and come accross a strange problem. Eventhough that we have set the ringtone in the handset to ring5.wav. When there is a call come in the CS410 is forcing the handset to ring with ring3.wav option. We have tried to setup the handset with other available ring tone options but the CS410 always force the handset to ring with only ring3.wav option. The same problem happened across all 4 connected handsets. Is there any way we could specify the ringtone on the CS410 configuration? Really appreciate if you could help us to modify this as the client find this ring3.wav tone is particularly annoying to him :( . Thanks very much for your help

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Not familiar with that handset, but can you simply rename the ring tones within the phone to what you want? I like simple, can you tell?

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Not familiar with that handset, but can you simply rename the ring tones within the phone to what you want? I like simple, can you tell?

Yes, we did the setting within the phone but when calls come through the CS410 still forcing all connected handset to ring with that particular ringtone. :( .

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Yes, we did the setting within the phone but when calls come through the CS410 still forcing all connected handset to ring with that particular ringtone. :( .

 

It's appear to us that this ringtone is stored somewhere on the CS410 and is pass on to the handset when call come through.

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It's appear to us that this ringtone is stored somewhere on the CS410 and is pass on to the handset when call come through.

 

Can you send the SIP INVITE log that is sent from the PBX to the phone?

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How do you do it?

 

Here are the messages that I can find on the SIP log (Logging level is 9)

 

[9] 2009/10/26 13:17:42: Resolve 6758: a udp 10.1.1.83 5062

[9] 2009/10/26 13:17:42: Resolve 6758: udp 10.1.1.83 5062

[9] 2009/10/26 13:19:16: Resolve 6759: aaaa udp 10.1.1.81 5062

[9] 2009/10/26 13:19:16: Resolve 6759: a udp 10.1.1.81 5062

[9] 2009/10/26 13:19:16: Resolve 6759: udp 10.1.1.81 5062

[5] 2009/10/26 13:20:09: Voicemail from to too short, discarding it

[9] 2009/10/26 13:20:09: Resolve 6760: url sip:127.0.0.1:5062;line=1

[9] 2009/10/26 13:20:09: Resolve 6760: udp 127.0.0.1 5062

[7] 2009/10/26 13:20:09: Call a5eb93b0@fxo#f0875f4fe9: Clear last request

[5] 2009/10/26 13:20:09: BYE Response: Terminate a5eb93b0@fxo

[9] 2009/10/26 13:20:16: Resolve 6761: aaaa udp 10.1.1.82 5062

[9] 2009/10/26 13:20:16: Resolve 6761: a udp 10.1.1.82 5062

[9] 2009/10/26 13:20:16: Resolve 6761: udp 10.1.1.82 5062

[9] 2009/10/26 13:20:27: Resolve 6762: aaaa udp 10.1.1.83 5062

[9] 2009/10/26 13:20:27: Resolve 6762: a udp 10.1.1.83 5062

[9] 2009/10/26 13:20:27: Resolve 6762: udp 10.1.1.83 5062

[9] 2009/10/26 13:22:16: Resolve 6763: aaaa udp 10.1.1.81 5062

[9] 2009/10/26 13:22:16: Resolve 6763: a udp 10.1.1.81 5062

[9] 2009/10/26 13:22:16: Resolve 6763: udp 10.1.1.81 5062

[9] 2009/10/26 13:23:09: Resolve 6764: aaaa udp 10.1.1.82 5062

[9] 2009/10/26 13:23:09: Resolve 6764: a udp 10.1.1.82 5062

[9] 2009/10/26 13:23:09: Resolve 6764: udp 10.1.1.82 5062

[9] 2009/10/26 13:23:11: Resolve 6765: aaaa udp 10.1.1.83 5062

[9] 2009/10/26 13:23:11: Resolve 6765: a udp 10.1.1.83 5062

[9] 2009/10/26 13:23:11: Resolve 6765: udp 10.1.1.83 5062

[9] 2009/10/26 13:23:32: UDP: Opening socket on 0.0.0.0:59398

[9] 2009/10/26 13:23:32: UDP: Opening socket on 0.0.0.0:59399

[5] 2009/10/26 13:23:32: Identify trunk (IP address/port and domain match) 1

[9] 2009/10/26 13:23:32: Resolve 6766: aaaa udp 127.0.0.1 5062

[9] 2009/10/26 13:23:32: Resolve 6766: a udp 127.0.0.1 5062

[9] 2009/10/26 13:23:32: Resolve 6766: udp 127.0.0.1 5062

[7] 2009/10/26 13:23:32: Set packet length to 20

[6] 2009/10/26 13:23:32: Sending RTP for b0fb3d30@fxo#90b3a2e6c8 to 1.1.1.2:2112

[5] 2009/10/26 13:23:32: Trunk PSTN (not global) sends call to account 40 in domain localhost

[7] 2009/10/26 13:23:32: Attendant: Calling extension 40

[9] 2009/10/26 13:23:32: UDP: Opening socket on 0.0.0.0:54724

[9] 2009/10/26 13:23:32: UDP: Opening socket on 0.0.0.0:54725

[9] 2009/10/26 13:23:32: Resolve 6767: url sip:40@10.1.1.81:5062

[9] 2009/10/26 13:23:32: Resolve 6767: udp 10.1.1.81 5062

[8] 2009/10/26 13:23:32: Play audio_moh/noise.wav

[7] 2009/10/26 13:23:32: Set packet length to 20

[8] 2009/10/26 13:23:32: Play audio_en/ringback.wav

[6] 2009/10/26 13:23:32: Send codec pcmu/8000

[9] 2009/10/26 13:23:32: Resolve 6768: aaaa udp 127.0.0.1 5062

[9] 2009/10/26 13:23:32: Resolve 6768: a udp 127.0.0.1 5062

[9] 2009/10/26 13:23:32: Resolve 6768: udp 127.0.0.1 5062

[8] 2009/10/26 13:23:42: Attendant: Timeout (extension)

[9] 2009/10/26 13:23:42: Resolve 6769: udp 10.1.1.81 5062

[7] 2009/10/26 13:23:42: Call 27a0924a@pbx#231435015: Clear last request

[7] 2009/10/26 13:23:42: Call 27a0924a@pbx#231435015: Clear last INVITE

[9] 2009/10/26 13:23:42: Resolve 6770: url sip:40@10.1.1.81:5062

[9] 2009/10/26 13:23:42: Resolve 6770: udp 10.1.1.81 5062

[5] 2009/10/26 13:23:42: INVITE Response 487 Request Terminated: Terminate 27a0924a@pbx

[7] 2009/10/26 13:23:42: Other Ports: 1

[7] 2009/10/26 13:23:42: Call Port: b0fb3d30@fxo#90b3a2e6c8

[5] 2009/10/26 13:23:42: Attendant: Redirect to 41

[8] 2009/10/26 13:23:42: Play audio_moh/noise.wav

[9] 2009/10/26 13:23:42: UDP: Opening socket on 0.0.0.0:64300

[9] 2009/10/26 13:23:42: UDP: Opening socket on 0.0.0.0:64301

[9] 2009/10/26 13:23:42: Resolve 6771: url sip:41@10.1.1.82:5062

[9] 2009/10/26 13:23:42: Resolve 6771: udp 10.1.1.82 5062

[8] 2009/10/26 13:23:42: Play audio_en/ringback.wav

[6] 2009/10/26 13:23:42: Send codec pcmu/8000

[9] 2009/10/26 13:23:42: Resolve 6772: aaaa udp 127.0.0.1 5062

[9] 2009/10/26 13:23:42: Resolve 6772: a udp 127.0.0.1 5062

[9] 2009/10/26 13:23:42: Resolve 6772: udp 127.0.0.1 5062

[7] 2009/10/26 13:23:49: Call ac7cb30d@pbx#1919158502: Clear last INVITE

[6] 2009/10/26 13:23:49: Send codec=pcmu/8000 afrer answer

[6] 2009/10/26 13:23:49: Sending RTP for ac7cb30d@pbx#1919158502 to 10.1.1.82:11786

[9] 2009/10/26 13:23:49: Resolve 6773: url sip:41@10.1.1.82:5062

[9] 2009/10/26 13:23:49: Resolve 6773: udp 10.1.1.82 5062

[7] 2009/10/26 13:23:49: Determine pass-through mode after receiving response

[9] 2009/10/26 13:23:49: Resolve 6774: aaaa udp 127.0.0.1 5062

[9] 2009/10/26 13:23:49: Resolve 6774: a udp 127.0.0.1 5062

[9] 2009/10/26 13:23:49: Resolve 6774: udp 127.0.0.1 5062

[7] 2009/10/26 13:23:49: ac7cb30d@pbx#1919158502: RTP pass-through mode

[7] 2009/10/26 13:23:49: b0fb3d30@fxo#90b3a2e6c8: RTP pass-through mode

[9] 2009/10/26 13:23:55: Resolve 6775: aaaa udp 10.1.1.82 5062

[9] 2009/10/26 13:23:55: Resolve 6775: a udp 10.1.1.82 5062

[9] 2009/10/26 13:23:55: Resolve 6775: udp 10.1.1.82 5062

[7] 2009/10/26 13:23:55: b0fb3d30@fxo#90b3a2e6c8: Media-aware pass-through mode

[7] 2009/10/26 13:23:55: Other Ports: 1

[7] 2009/10/26 13:23:55: Call Port: b0fb3d30@fxo#90b3a2e6c8

[9] 2009/10/26 13:23:55: Resolve 6776: url sip:127.0.0.1:5062;line=1

[9] 2009/10/26 13:23:55: Resolve 6776: udp 127.0.0.1 5062

[7] 2009/10/26 13:23:55: Call b0fb3d30@fxo#90b3a2e6c8: Clear last request

[5] 2009/10/26 13:23:55: BYE Response: Terminate b0fb3d30@fxo

[9] 2009/10/26 13:25:15: Resolve 6777: aaaa udp 10.1.1.81 5062

[9] 2009/10/26 13:25:15: Resolve 6777: a udp 10.1.1.81 5062

[9] 2009/10/26 13:25:15: Resolve 6777: udp 10.1.1.81 5062

[9] 2009/10/26 13:25:56: Resolve 6778: aaaa udp 10.1.1.83 5062

[9] 2009/10/26 13:25:56: Resolve 6778: a udp 10.1.1.83 5062

[9] 2009/10/26 13:25:56: Resolve 6778: udp 10.1.1.83 5062

[9] 2009/10/26 13:26:02: Resolve 6779: aaaa udp 10.1.1.82 5062

[9] 2009/10/26 13:26:02: Resolve 6779: a udp 10.1.1.82 5062

[9] 2009/10/26 13:26:02: Resolve 6779: udp 10.1.1.82 5062

[6] 2009/10/26 13:26:20: Webserver: Could not read file img/main_logo.gif

[9] 2009/10/26 13:28:15: Resolve 6780: aaaa udp 10.1.1.81 5062

[9] 2009/10/26 13:28:15: Resolve 6780: a udp 10.1.1.81 5062

[9] 2009/10/26 13:28:15: Resolve 6780: udp 10.1.1.81

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Here are the messages that I can find on the SIP log (Logging level is 9)

 

Turn on the logging on for "other" SIP messages. Then you should see the whole SIP packet.

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this is a bug from the lastest and previous version of firmware (3.4.0.3201 and 3.4.0.3194). We have found that if a call is directed to a hunt group or agent group then it working fine with the handset ringtone setting. However if a call directed directly to an extension then a default ringtone was handed over from the PBX which could not be changed

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We are in same situation. Running Yealink T28's with Feb 2010 Firmware (latest) hunt group ring tone is ok, but dialling direct from a trunk (ie a DID from PSTN) is not as set in phone. Did you even work out a solution?

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we also have the same situation using TIPTEL IP286 ( = Yealink T-28 )

 

if you call the extension direct then a default ringtone :angry: is handed over which cannot be changed.

 

Is there a solution ?

 

Regards,

Markus

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we also have the same situation using TIPTEL IP286 ( = Yealink T-28 )

 

if you call the extension direct then a default ringtone :angry: is handed over which cannot be changed.

 

Check the ringtones.xml file (in admin/email/texts), maybe you need to edit the content there.

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I do not find any ringtones.xml file on the computer where the PBX is installed.

 

Where should it be located ?

 

Should be visible in the web interface in the menu mentioned above.

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ok i found the file:

 

<?xml version="1.0" ?>

- <ringtones>

- <tone name="custom1">

- <!-- vendor snom off

-->

<vendor ua="Polycom.*">Custom 1</vendor>

<vendor ua="Cisco-.*"><Bellcore-dr4></vendor>

- <!-- vendor snom on

-->

<vendor><http://127.0.0.1/Bellcore-dr4></vendor>

</tone>

- <tone name="custom2">

- <!-- vendor snom off

-->

<vendor ua="Polycom.*">Custom 2</vendor>

<vendor ua="Cisco-.*"><Bellcore-dr4></vendor>

- <!-- vendor snom on

-->

<vendor><http://127.0.0.1/Bellcore-dr4></vendor>

</tone>

- <tone name="custom3">

- <!-- vendor snom off

-->

<vendor ua="Polycom.*">Custom 3</vendor>

<vendor ua="Cisco-.*"><Bellcore-dr4></vendor>

- <!-- vendor snom on

-->

<vendor><http://127.0.0.1/Bellcore-dr4></vendor>

</tone>

- <tone name="custom4">

- <!-- vendor snom off

-->

<vendor ua="Polycom.*">Custom 4</vendor>

<vendor ua="Cisco-.*"><Bellcore-dr4></vendor>

- <!-- vendor snom on

-->

<vendor><http://127.0.0.1/Bellcore-dr4></vendor>

</tone>

- <tone name="internal" type="internal">

- <!-- vendor snom off

-->

<vendor ua="Polycom.*">Internal</vendor>

<vendor ua="Cisco-.*"><Bellcore-dr2></vendor>

<vendor ua="Grandstream HT-.*" />

- <!-- vendor snom on

-->

<vendor><http://127.0.0.1/Bellcore-dr2></vendor>

</tone>

- <tone name="external" type="external">

- <!-- vendor snom off

-->

<vendor ua="Polycom.*">External</vendor>

<vendor ua="Cisco-.*"><Bellcore-dr3></vendor>

<vendor ua="Grandstream HT-.*" />

- <!-- vendor snom on

-->

<vendor><http://127.0.0.1/Bellcore-dr3></vendor>

</tone>

- <tone name="intercom" type="intercom">

- <!-- vendor snom off

-->

<vendor ua="Polycom.*">Auto Answer</vendor>

<vendor ua="Cisco-.*" type="call-info">auto-answer=0</vendor>

<vendor ua="optiPoint .*" type="call-info">auto-answer=0</vendor>

- <!-- vendor snom on

-->

<vendor ua="snom.*" type="call-info"><{from-uri}>;answer-after=0</vendor>

- <!-- vendor snom off

-->

<vendor ua="Linksys.*" type="call-info"><{from-uri}>;answer-after=0</vendor>

<vendor ua="Aastra.*" type="call-info"><{from-uri}>;answer-after=0</vendor>

<vendor ua="Yealink.*" type="call-info"><{from-uri}>;answer-after=0</vendor>

- <!-- vendor snom on

-->

<vendor type="answer-mode">Auto</vendor>

</tone>

</ringtones>

 

how should i configure the usage of a special ringtone for Yealink / Tiptel phones ?

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Well the question is what "User-Agent" header the phone presents. If it is a Yealink OEM, it might be something like "Tiptel", then you have to change the entries below:

 

 

<?xml version="1.0"?>
<ringtones>
 <tone name="custom1">
  <vendor ua="Polycom.*">Custom 1</vendor>
  <vendor ua="Cisco-.*"><Bellcore-dr4></vendor>
  <vendor><http://127.0.0.1/Bellcore-dr4></vendor>
</tone>
 <tone name="custom2">
  <vendor ua="Polycom.*">Custom 2</vendor>
  <vendor ua="Cisco-.*"><Bellcore-dr4></vendor>
  <vendor><http://127.0.0.1/Bellcore-dr4></vendor>
</tone>
 <tone name="custom3">
  <vendor ua="Polycom.*">Custom 3</vendor>
  <vendor ua="Cisco-.*"><Bellcore-dr4></vendor>
  <vendor><http://127.0.0.1/Bellcore-dr4></vendor>
</tone>
 <tone name="custom4">
  <vendor ua="Polycom.*">Custom 4</vendor>
  <vendor ua="Cisco-.*"><Bellcore-dr4></vendor>
  <vendor><http://127.0.0.1/Bellcore-dr4></vendor>
</tone>
 <tone name="internal" type="internal">
  <vendor ua="Polycom.*">Internal</vendor>
  <vendor ua="Cisco-.*"><Bellcore-dr2></vendor>
  <vendor ua="Grandstream HT-.*"></vendor>
  <vendor><http://127.0.0.1/Bellcore-dr2></vendor>
</tone>
 <tone name="external" type="external">
  <vendor ua="Polycom.*">External</vendor>
  <vendor ua="Cisco-.*"><Bellcore-dr3></vendor>
  <vendor ua="Grandstream HT-.*"></vendor>
  <vendor><http://127.0.0.1/Bellcore-dr3></vendor>
</tone>
 <tone name="intercom" type="intercom">
  <vendor ua="Polycom.*">Auto Answer</vendor>
  <vendor ua="Cisco-.*" type="call-info">auto-answer=0</vendor>
  <vendor ua="optiPoint .*" type="call-info">auto-answer=0</vendor>
  <vendor ua="snom.*" type="call-info"><{from-uri}>;answer-after=0</vendor>
  <vendor ua="Linksys.*" type="call-info"><{from-uri}>;answer-after=0</vendor>
  <vendor ua="Aastra.*" type="call-info"><{from-uri}>;answer-after=0</vendor>
  <vendor ua="Yealink.*" type="call-info"><{from-uri}>;answer-after=0</vendor>
  <vendor type="answer-mode">Auto</vendor>
</tone>
</ringtones>

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well the extension is registered as:

 

REGISTER 200 <sip:200@192.168.10.218:5062> (udp:192.168.10.218:5062) Tiptel IP 286 2.51.13.2 00156513e98d 212 check-sync

 

what entry is needed to use the ringtone ring2.wav ( which is a internal ringtone in the phone )

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well the extension is registered as:

 

REGISTER 200 <sip:200@192.168.10.218:5062> (udp:192.168.10.218:5062) Tiptel IP 286 2.51.13.2 00156513e98d 212 check-sync

 

what entry is needed to use the ringtone ring2.wav ( which is a internal ringtone in the phone )

 

 

I think this question was repeated few times here and there – The selected Ringer on the snom phone does not work, why? (you can apply the same for other vendor phones too)

 

This is because, when PBX sends the INVITE, it also sends Alert-Info header to the phone. This forces the phone to use ringer specified in that header. This header is controlled by the ‘ringtones.xml’ file that is built into PBX binary.

 

You can change this behavior by placing a modified ‘ringtones.xml’ file to ‘html’ folder and restarting the PBX. (modified section is shown below)

 

If you observe the “<tone name=”internal” … > section in the file, you will get some clue how this works. This section has a line for “vendor ua=snom” without any specific ring tone information. So, the phone will use whatever ringer is set on the device itself, instead of getting the directive from the PBX.

 

If you want this to be controlled by PBX, then delete “vendor ua=snom” line and restart the PBX.

 

 

<tone name="internal" type="internal">

<vendor ua="Polycom.*">Internal</vendor>

<vendor ua="Cisco-.*"><Bellcore-dr2></vendor>

<vendor ua="Grandstream HT-.*"/>

<vendor ua="snom.*"/>

<vendor><http://127.0.0.1/Bellcore-dr2></vendor>

</tone>

 

For more info please refer to the ringtones chapter (chap 17) on the admin guide http://www.pbxnsip.com/docs/pbxnsip_adminguide_v4.pdf

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