HELLO & WELCOME

[ThinkYEAH]

Hi,

I have Elastix 2.5.0-17, and it was working great, lately my colegu reported me that calling outside works fine, but some number we call has extension and when we press "number" it won't redirect us.

For e example i call 03xxxxxx and i hear IVR that says for English press 1, for other language press 2, When i press 1 or 2 nothing happnes, i tried with different operators by same problem for all.

Thanks for help :)

[WelshPaul]

Sounds like an issue with the DTMF settings.

Have you added "dtmfmode=auto" and "dtmf=auto" to the trunks peer details?

Choices available to you are: inband, rfc2833, info or auto

inband: The device that you press the key on will generate the DTMF tones. - If the codec is not ulaw or alaw then the DTMF tones will be distorted by the audio compression and will not be recognised. If the phone is set for RFC2833 and asterisk is set for inband then you may not hear anything.

rfc2833: http://www.ietf.org/rfc/rfc2833.txt

info: See SIP method info and SIP info DTMF or http://www.ietf.org/rfc/rfc2976.txt

auto: Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones if the remote side does not indicate support of rfc2833 in SDP. This feature was added on Sep 6, 2005 and is not available in Asterisk 1.0.x.

[ThinkYEAH]

dtmfmode

Hi,

In peer details was "dtmfmode=inband" --> i have change it to auto, "dtmf=auto" was missing so i added it after dtmfmode and now is working.

It's kinda weird because only me has access in to Elastix admin panel, and as far i know i havent edit any config lately.

Thanks alot for your help WelshPaul

I have 3 more question if you can help me:
1. We have IVR, for sales press 1, for service press 2 ... when some1 press 1 or other number it transfer call in to mobile phone number, but the problem is that in that person who awnser calls can only see our fix number, every call which is transfered in that mobile phone shows same phone number 03xxxxxx (our company number) and we need sometimes to get clients phone number. is there any chance to show caller phone number to person who awnsers after transfer is made?

2. How to increase IVR allowed calls number? At this time only 1 person can call our company number, in case user1 is calling, user2 can't call our phone number...

3. We have 10 Grandstream DP750 phones, most of time we can't hear when we call outside or when some1 call us, sound quality is very poor or it stops few seconds... Is there any way to fix this issue?


Thanks again for your help!

[ThinkYEAH]

Edit:
We have 2 DP750 DECT Base Station with 10 phones ^ (can't edit my last post)

[ThinkYEAH]

Hi Paul,

After i added "dtmfmode=auto" and "dtmf=auto" now when i try to call outside i hear this "All circuits are busy now. Please try your call again later"

When i reboot virtual machine (Elastix) it starts working again. So, i have to reboot this virtual machine everytime it stops working.

Sorry to bother you with such problems :s

[WelshPaul]

I have 3 more question if you can help me:
1. We have IVR, for sales press 1, for service press 2 ... when some1 press 1 or other number it transfer call in to mobile phone number, but the problem is that in that person who awnser calls can only see our fix number, every call which is transfered in that mobile phone shows same phone number 03xxxxxx (our company number) and we need sometimes to get clients phone number. is there any chance to show caller phone number to person who awnsers after transfer is made?

The reason you see your own fixed number is because the call your receiving is from your PBX which is using the trunk assigned to your fixed number to make the call.

Let me explain:

The original caller calls your fixed number, your PBX answers the call and does whatever you have configured it to do, play IVR etc. Now, if they press 1 and get transferred to another landline or mobile, it's not them calling the transferred number, it's your PBX and so whatever caller ID of the trunk used to transfer the call is used.

There is no way to pass the original callers caller ID unfortunately. The reason for this is because providers don't allow us to alter CLI's via our PBX. Well none that I know of!

2. How to increase IVR allowed calls number? At this time only 1 person can call our company number, in case user1 is calling, user2 can't call our phone number...

Does your trunk/sip provider have a limit on how many calls you can receive or make at any one given time? If not, have you anything set under the Maximum Channels option?

3. We have 10 Grandstream DP750 phones, most of time we can't hear when we call outside or when some1 call us, sound quality is very poor or it stops few seconds... Is there any way to fix this issue?

Check what Codecs the phones are currently using. Configure them to use the G.711U only and make sure that your PBX is configured to use the same.

Hi Paul,

After i added "dtmfmode=auto" and "dtmf=auto" now when i try to call outside i hear this "All circuits are busy now. Please try your call again later"

When i reboot virtual machine (Elastix) it starts working again. So, i have to reboot this virtual machine everytime it stops working.

Sorry to bother you with such problems :s

No problem, adding the DTMF option above wouldn't cause your trunk to go offline (which is why your probably getting the "All circuits are busy now. Please try your call again later". You sure nobody has been playing with the configuration? :P

You need to watch the console while dialing or look at the log information in there. Anyway, try reverting back to the previous configuration and see if you still get this problem. Le me know...

[ThinkYEAH]

Does your trunk/sip provider have a limit on how many calls you can receive or make at any one given time? If not, have you anything set under the Maximum Channels option?

Nope, there is not limit by provider, and "Maximum Channels" in to elastix settings is blank

Check what Codecs the phones are currently using. Configure them to use the G.711U only and make sure that your PBX is configured to use the same.

Here you can view audio settings for our grandstream phones: https://imgur.com/a/uJb7Hx3 - G.711U is missing, also here you can see FreePBX Audo codecs: https://i.imgur.com/q0uZY5x.png

No problem, adding the DTMF option above wouldn't cause your trunk to go offline (which is why your probably getting the "All circuits are busy now. Please try your call again later". You sure nobody has been playing with the configuration? :P

After i removed dtmfmode=auto" and "dtmf=auto" i can call but i can't press numbers (Ex. press 1 for sales)...

I would like to thank you for help Paul!

[WelshPaul]

Nope, there is not limit by provider, and "Maximum Channels" in to elastix settings is blank

Odd, try setting it to 100.

Here you can view audio settings for our grandstream phones: https://imgur.com/a/uJb7Hx3 - G.711U is missing, also here you can see FreePBX Audo codecs: https://i.imgur.com/q0uZY5x.png

PCMU is G.711U :P
http://www.grandstream.com/support/faq/ ... ech-codecs

After i removed dtmfmode=auto" and "dtmf=auto" i can call but i can't press numbers (Ex. press 1 for sales)...

Yea, but we are waiting to see if problem you reported above re-appears. Adding dtmfmode=auto" and "dtmf=auto" to the trunk worked initially, correct? If so, best to wait and see if the problem re-occures without these settings...

You could try adding just dtmfmode=auto" to the trunk configuration but I would hold off for a few hours.