Plantronics + Polycom. Now together as Poly Logo

No tone on Auto Answer

Occasional Visitor

No tone on Auto Answer

Moved by Mod from => here <=

 

Hi,

 

I am not too sure if it is the same problem that I encountered or not.  My client is running 3.2.4.0267 on 550 with Asterisk 1.6.  I have set up the intercom and BLF and it works correctly.  However, whenever the boss intercoms the 7plus extension (7102) - says ext. 102, it did not generalte beep tone (it goes silent) then speaker phone comes up.  How can I get the beep tone generated?  The followings is my sip.cfg setup.

 

<alertInfo voIpProt.SIP.alertInfo.1.value="intercom" voIpProt.SIP.alertInfo.1.class="4" />

 

<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="200" se.rt.4.ringer="1" se.rt.4.callWait="6" se.rt.4.mod="1" />

I also tried setting the alertinfo class to be "3", and add timeout and ringer type = 1, however, the result is the same.

My client just wanted to hear beep sound and talk just like 3.3.4 revision.

 

<alertInfo voIpProt.SIP.alertInfo.1.value="intercom" voIpProt.SIP.alertInfo.1.class="3" />

 

<AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer" se.rt.3.timeout="200" se.rt.3.ringer="1" />


====

Thanks for your help.

 

Jim

 

 

Message 1 of 11
10 REPLIES 10
Polycom Employee & Community Manager

Re: No tone on Auto Answer

Hello Jim,

welcome to the Polycom Community.

Running an older software like SIP 3.2.x or older uses the sip.cfg and phone1.cfg.

 

You could either add it to the existing or create a new cfg file that is loaded before the above.

 

Adding it to the sip.cfg:

 

Open the phone’s sip.cfg file with a text editor and locate the line starting with the text  <alertinfo. Then, add the key voIpProt.SIP.alertinfo.2.value with the value you set in the Asterisk dial plan in step 1. 
As well, add the key voIpProt.SIP.alertinfo.2.class with a value of 4.

 

 

.
voIpProt.SIP.alertInfo.2.value="intercom" voIpProt.SIP.alertInfo.2.class="4"
..
Fragment from “sip.cfg”

 

These changes allow the phone to recognize the alert in the invite message sent by Asterisk and to give it an internal ring class of 4. This internal class is used to reference the ring type to use, as shown below.

 

.
<VISUAL_ONLY se.rt.2.name="Visual" se.rt.2.type="visual" />
 <AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer" />
  <RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"
..
Fragment from “sip.cfg”

 

With an internal ring class of 4, the line starting with <RING_ANSWER> is referenced. If the was set to voIpProt.SIP.alertInfo.1.class a value of 3, the line starting with would have been referenced, prompting the phone to answer without any ringing at all.

 

The default setting for se.rt.4.timeout="2000"  (the amount of time the phone will ring before auto-answering) is 2000 (milliseconds). For the intercom application, the setting is changed for a much shorter ring time of 200 milliseconds.

 

The asterisk extensions.conf could use a similar entry to below:

 

.
[polycom]
..
exten => _9XXX,1,SIPAddHeader(Alert-Info: info=intercom)
exten => _9XXX,n,Dial(SIP/${EXTEN:1})
..
Fragment from “extensions.conf”

 

The value of the ALERT_INFO variable is not important, but it must match the phone configuration performed in step 2 (below). Ideally, it should be a short descriptive string. (In this example, the value is set to intercom.)

 

In the above example, when the digits 98036 are dialed, the variable is set, and then Asterisk sends the SIP invite (with the set variable) to extension 8036.

 

Please check the Admin Guide if you need further information.

 

 

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 2 of 11
Occasional Visitor

Re: No tone on Auto Answer

Hi Steffen,

 

Thanks for the help.  However, your instructions have generated a RING tone.  I was wondering how can I generate BEEP tone instead?  Exactly like intercom annoucement tone?  The customer is used to BEEP tone.

 

Jim

Message 3 of 11
Polycom Employee & Community Manager

Re: No tone on Auto Answer

Hello Jim,

 

I recommend that as advised you check the SIP 3.2.2 Admin Guide on Page A - 41.

 

Currently se.rt.4.ringer="2" uses Low Trill. Have a play with these settings.

 

Best Regards

 

Steffen Baier

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 4 of 11
Occasional Visitor

Re: No tone on Auto Answer

Steffen,

 

Yes, understood.  The customer wants to have a beep sound (intercom announcement tone) instead.  Can I customize it and how on 3.2.4.x release?  I have a beep.wav file but couldn't get it work.

 

Thanks,

 

Jim

Message 5 of 11
Polycom Employee & Community Manager

Re: No tone on Auto Answer

Hello Jim,

 

according to the SIP Admin Guide:

 

Sampled audio files 1-21 all use the same built-in file unless that file has been replaced with a downloaded file. For more information, refer to Sampled Audio for Sound Effects <saf/> on page A-37.

 Above chapter mentions the limitations (filetype etc.)

 

You could test the LoudRing.wav that comes usually with the phone software.

 

If you need further help I can only advise you that as a Polycom employee I am unable to provide free support.

 

Either a member of the community is willing to help you or your customer or you can call Polycom support and will be charged PPI.

 

This is described in the FAQ.

 

Best Regards

 

Steffen Baier

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 6 of 11
Visitor

Re: No tone on Auto Answer

Steffen,

 

We are finding ourselves in need of something similar to what Jim wants to do.

 

We have Asterisk 1.8 and Polycom IP 331 phones.  The phones are running 4.3.1.0440 boot, and 3.3.2.0413 load.

 

We are setting up paging.  Right now the behavior is that the phones will ring with low-trill, then go off-hook.  Unfortunately, our default ring tone is low-trill too, so users think that they are getting a call rather than a page.  We would like to move to a different ring pattern for paging, and long-term, possibly using no tone, just going off-hook.

 

I believe I am following the posts that you sent to Jim, but I have a few questions that might help me work through this process.

 

- My sip.cfg file is pretty sparse.  It has no entries in it that refer to the ringtype table for example.  This leads me to believe that all of the settings/tables (like the ring type table) are built into the firmware by default and that the sip.cfg file settings are there to simply "override" the default settings for those items that need it.  Is that correct?

 

- If I want to override a single table setting or add a new setting in a table (let's use ringtype table again), can I overwrite a single entry in the table or do I have to include the entire table in the sip.cfg with my changes in it?  For example, here is a table from a sample sip.cfg file that (I believe) matches up with my firmware version:

 

<ringType se.rt.enabled="1" se.rt.modification.enabled="1">

<DEFAULT se.rt.1.name="Default" se.rt.1.type="ring" se.rt.1.ringer="2" se.rt.1.callWait="6" se.rt.1.mod="1" />

<VISUAL_ONLY se.rt.2.name="Visual" se.rt.2.type="visual" />

<AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer" />

<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1" />

<INTERNAL se.rt.5.name="Internal" se.rt.5.type="ring" se.rt.5.ringer="2" se.rt.5.callWait="6" se.rt.5.mod="1" />

<EXTERNAL se.rt.6.name="External" se.rt.6.type="ring" se.rt.6.ringer="2" se.rt.6.callWait="6" se.rt.6.mod="1" />

<EMERGENCY se.rt.7.name="Emergency" se.rt.7.type="ring" se.rt.7.ringer="2" se.rt.7.callWait="6" se.rt.7.mod="1" />

<CUSTOM_1 se.rt.8.name="Custom 1" se.rt.8.type="ring" se.rt.8.ringer="5" se.rt.8.callWait="7" se.rt.8.mod="1" />

<CUSTOM_2 se.rt.9.name="Custom 2" se.rt.9.type="ring" se.rt.9.ringer="7" se.rt.9.callWait="7" se.rt.9.mod="1" />

<CUSTOM_3 se.rt.10.name="Custom 3" se.rt.10.type="ring" se.rt.10.ringer="9" se.rt.10.callWait="7" se.rt.10.mod="1"&#0;/>

<CUSTOM_4 se.rt.11.name="Custom 4" se.rt.11.type="ring" se.rt.11.ringer="11" se.rt.11.callWait="7" se.rt.11.mod="1" />

</ringType>

 

Let's say that I wanted to change the RIng Answer line to use ring pattern 11, triplet.  Can I add this to my sip.cfg:

 

<ringType se.rt.enabled="1" se.rt.modification.enabled="1">

<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="2000" se.rt.4.ringer="11" se.rt.4.callWait="6" se.rt.4.mod="1" />

</ringType>

 

Will that change *just* the Ring_Answer entry in the table?  Or will I have wiped out the rest of the entries?

 

And, can I do this:

 

<ringType se.rt.enabled="1" se.rt.modification.enabled="1">

<CUSTOM_5 se.rt.12.name="Custom 4" se.rt.12.type="ring" se.rt.12.ringer="11" se.rt.12.callWait="7" se.rt.12.mod="1" />

</ringType>

 

to add an entry to the table, again without wiping it out?

 

If the answer is 'yes' on able to modify/add, then I should be good-to-go.  I can use 11/triplet to test to make sure that I am effectively changing that behavior, then 1/silent for the case where we would want no ring, only off-hook.

 

Thanks for your help on this.

 

Bryan Hunt

Message 7 of 11
Polycom Employee & Community Manager

Re: No tone on Auto Answer

Hello Bryan,

welcome to the Polycom Community.

 

Without going into to much details (as this is described in the FAQ) since UCS 3.3.x you no longer have a sip.cfg or a phone1.cfg.

 

Any Parameter that should change from its standard value must be specified.

 

If only a parameter is changed and not the whole setting like duration etc. only that parameter must be provided via a configuration file.

 

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 8 of 11
Visitor

Re: No tone on Auto Answer

Steffan,

 

I am using an Asterisk/FreePBX distro called PBX In A Flash.  Under that, I am using the End Point Manager module version 2.10.3.7.

 

Even though EPM pushes out 3.3.2.0413 firmware, it still (appears) to use sip.cfg.  I have made other changes to the sip.cfg file and they do affect the phones.

 

To find the firmware version, I did: menu/status/platform/application/main.  That is the correct spot, is that right?

 

Is it possible for EPM to "override" the config files and still use the sip.cfg file and format?

 

Thanks.

 

Bryan Hunt

Message 9 of 11
Polycom Employee & Community Manager

Re: No tone on Auto Answer

Hello Bryan,

 

again, not going into to much details (some reference => here <= and as usual in the admin guide.) since UCS 3.3.x we do no longer provide a sip.cfg / phone1.cfg.

 

You can of course call the files the phone loads anything you want but you should no longer use any pre - UCS 3.3.x Parameters as explained => here <=

 

Whoever is responsible for your platform should be consulted as they are aware how they utilize the Polycom provisioning method.

 

Best Regards

 

Steffen Baier

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 10 of 11