Hello All,
I have Polycom IP 7000 and I need to contact with external conference. When I call the numer +48 608 608 608and then I put the room numer and confirm sign #, I can hear the message that the call is transferred and it was impossible to connect. The asterisk logs show the following message:
I am able to contact to external conference using another SIP phone we have in our company. It seems to me that the there is an issue with the configuration of Polycom IP 7000.
Any help appreciated!
Regards
Hello zool,
welcome to the Polycom Community.
It is always useful to include the currently used software version as issue experienced may already be addressed in a newer release. This also allows yourself and others to check against current software release notes.
The community's VoIP FAQ contains this post here:
Oct 7, 2011 Question: Phone unable to send DTMF to an IVR system or how to troubleshoot DTMF issues
Resolution: Please check => here <=
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
If official support is required please check how to phone or open a case here
----------------Hello SteffenBaierUK,
The version of my Polycom IP 7000 is 3.1.2.0392.
Shall I update the software to a newer version: http://support.polycom.com/global/documents/support/setup_maintenance/products/voice/ssip7000_HDX_In... (page 12)? Is this type of update safe for my device?
Hello zool,
The community's VoIP FAQ contains this post here:
Oct 7, 2011 Question: How can I setup my Phone / Provisioning / Download / Upgrade / Update / Downgrade Software?
Resolution: Please check => here <=
For the SSIP7000 I recommend to use UCS 4.0.4
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
If official support is required please check how to phone or open a case here
----------------hello,
I have very old version 3.1.2.0392. Should I update it to 3.3 and after that update to 4.0.4?
The easier way maybe restarting the phone and set up provisioning server: downloads.polycom.com/voice/IP7000
. the actualization to na newest version should download automaticaly.
what do you think about it?
Hello Zool,
You can directly update from UCS 3.3.x to UCS 4.0.4 via a locally setup provisioning server.
The Polycom hosted server does currently not offer UCS 4.0.4 but this is the version I recommend.
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
If official support is required please check how to phone or open a case here
----------------Hello Steffen Baier,
I have the last question. Can I directly update from UCS 3.2.4.0244 to UCS 4.0.4 via a <removed by the moderator> setup provisioning server?
Regards;
Hello zool,
I have removed the reference to this Polycom hosted server as this is for Polycom support entitled users only and has not got the bandwidth for a larger audience.
Yes you can use the server address that you are aware of to upgrade from 3.2.x to UCS 4.0.4
For any follow up questions please work with your Polycom reseller and / or Polycom support directly.
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
If official support is required please check how to phone or open a case here
----------------Hello,
Thanks for your help. The actualization to the new version UCS 4.0.4 was successful. When it comes to my secodn issue (contact with external conference), I did exactly according to your instructions (http://community.polycom.com/t5/VoIP/FAQ-Phone-unable-to-send-DTMF-to-an-IVR-system-or-how-to/td-p/4... but the problem occurs. I still cannot connect to external conference.
best regards
Hello zool,
there are various troubleshooting suggestions in the FAQ including the DTMF post.
In addition Astersik can be used to test DTMF.
Example:
Exten => 501,1,Macro(DTMFTest) [macro-DTMFTest] ;for additional sounds see : ;http://downloads.asterisk.org/pub/telephony/sounds/releases/?C=N;O=A ;http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-en-gsm-1.4.7.tar.g... Exten => s,1,PlayBack(please-enter-your) Exten => s,2,PlayBack(number) Exten => s,3,Read(NUMBER,,5) Exten => s,4,SayDigits(${NUMBER}) Exten => s,5,Goto(3)
In above example 501 can be dialed and then you can press a maximum of 5 keys on the phone and the Asterisk server will repeat the Digits that you have pressed.
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
If official support is required please check how to phone or open a case here
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