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IP Dialing

Visitor

IP Dialing

I have a VVX 400 and I'm trying to set up a Line Key or Soft Key that will be labeled "Page" that would do a direct IP Dial to the analog paging adapter on the network. For example, it would literally dial 100@10.0.0.50, where 10.0.0.50 is the analog paging adapter on the same network as the VVX 400.

 

The phone is also registered to a server for regular calls, which seems to be confusing it, as it's sending an INVITE of 100@10.0.0.50 to the server it's registered with rather than contacting the paging adapter at 10.0.0.50.

 

Any ideas on this? I do not want to use the built-in multicast paging feature, the analog adapter doesn't support it. I just need to do a direct IP dial from a Line Key?

 

Thanks ...

Message 1 of 8
7 REPLIES 7
Visitor

Re: IP Dialing

Anyone have any ideas on this? I really need to be able to make a SIP call directly to an ATA for overhead paging, without use of the server the phone is registered to?

 

Thanks!

Message 2 of 8
Polycom Employee & Community Manager

Re: IP Dialing

Hello socalpilot78,

welcome to the Polycom Community.

 

Could I kindly ask you not to bump your post as people will respond if they have anything to add. This is also noted in the must read first.


It is always useful to include the currently used UC Software version as issues experienced or a question asked may already be addressed in a newer release.

This also allows yourself and others to check against current software release notes, Administrator Guides or FAQ post’s.

The above is also stated in the "Must Read First" and is the absolute minimum requirement every new post should include.

Therefore the Polycom VoIP FAQ contains this post here:

Question: How can I find out my SIP or UC Software Version of my Phone?
Resolution: Please check here

 

You can simply select a 2nd Line and select the following:

 

LineWithoutRegistration.PNG

 

Pressing Line 2 on the Phone and siply dialing the IP will send:

 

812607.877|sip  |0|00|Trying to send data to Destination 10.252.149.100 on socket 134
812607.880|sip  |0|00|>>> Data Send to 10.252.149.100:5060
812607.880|sip  |0|00|    INVITE sip:10.252.149.100 SIP/2.0
812607.880|sip  |0|00|    Via: SIP/2.0/UDP 10.252.149.61;branch=z9hG4bKb94d8ac1DAD76F83
812607.880|sip  |0|00|    From: "100" <sip:100@10.252.149.61>;tag=E5635DC9-D51B755B
812607.880|sip  |0|00|    To: <sip:10.252.149.100>
812607.880|sip  |0|00|    CSeq: 1 INVITE
812607.880|sip  |0|00|    Call-ID: c51840f9-b618836b-40df8e71@10.252.149.61
812607.880|sip  |0|00|    Contact: <sip:100@10.252.149.61>
812607.880|sip  |0|00|    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
812607.880|sip  |0|00|    User-Agent: PolycomVVX-VVX_410-UA/5.3.0.12074
812607.880|sip  |0|00|    Accept-Language: en
812607.880|sip  |0|00|    Supported: replaces,100rel
812607.880|sip  |0|00|    Allow-Events: conference,talk,hold
812607.880|sip  |0|00|    Max-Forwards: 70
812607.880|sip  |0|00|    Content-Type: application/sdp
812607.880|sip  |0|00|    Content-Length: 338
812607.880|sip  |0|00|    
812607.880|sip  |0|00|    v=0
812607.880|sip  |0|00|    o=- 264 264 IN IP4 10.252.149.61
812607.880|sip  |0|00|    s=Polycom IP Phone
812607.880|sip  |0|00|    c=IN IP4 10.252.149.61
812607.880|sip  |0|00|    t=0 0
812607.880|sip  |0|00|    a=sendrecv
812607.880|sip  |0|00|    m=audio 2222 RTP/AVP 9 102 0 8 18 127
812607.880|sip  |0|00|    a=rtpmap:9 G722/8000
812607.880|sip  |0|00|    a=rtpmap:102 G7221/16000
812607.880|sip  |0|00|    a=fmtp:102 bitrate=32000
812607.880|sip  |0|00|    a=rtpmap:0 PCMU/8000
812607.880|sip  |0|00|    a=rtpmap:8 PCMA/8000
812607.880|sip  |0|00|    a=rtpmap:18 G729/8000
812607.880|sip  |0|00|    a=fmtp:18 annexb=no
812607.880|sip  |0|00|    a=rtpmap:127 telephone-event/8000

Once you done the above store a local favorite in your directory with the IP of your paging system and then press the Line 2 and the newly created favorite.


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

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

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 3 of 8
Visitor

Re: IP Dialing

If I want it on button number 12 (bottom right), would I do the same, but for "Line 12" ?

 

Thanks, I will give a try and post back!

 

Sorry for bumping the post, but it had been 5 days without response.

 

 

Message 4 of 8
Visitor

Re: IP Dialing

I tried as you suggested but I'm not having luck creating the directory entry.

 

I tried to create a local directory entry with contact "100@192.168.1.50", however it keeps reverting the mode to "Tel" and doesn't keep "URL" for mode. I'm assuming I must be doing something wrong with the contact and I can't seem to figure it out?

 

I made sure that feature.urlDialing.enabled="1" is set.

 

Also, if I try to place a call manually and select "URL" and type the IP address, it sends an INVITE to the server it's registered to with the IP address of the paging adapter as the contact address, rather than sending an INVITE to the adapter directly? See the snip below from the INVITE from the phone. 10.100.1.10 is the server it's registered to and 192.168.1.50 is the ip address of the ATA paging adapter. Any ideas?

 

U 192.168.1.10:5060 -> 10.100.1.10:5060
INVITE sip:192.168.1.50 SIP/2.0.
Via: SIP/2.0/UDP 10.254.184.242;branch=z9hG4bKfe791dbA6D7BB0E.
From: "Test 1" <sip:101@local.212.service>;tag=D89A9423-AE106A96.
To: <sip:192.168.1.50>.
CSeq: 2 INVITE.

 

Message 5 of 8
Polycom Employee & Community Manager

Re: IP Dialing

Hello socalpilot78,

So far you have not even answered my basic question what software version the phone is actually using.

 

I simply tried this again with a stored favorite aka 100@10.252.149.100 and it still uses Line 2 if you select the line before pressing the key.

 

921811.138|sip  |0|00|>>> Data Send to 10.252.149.100:5060
921811.138|sip  |0|00|    INVITE sip:100@10.252.149.100 SIP/2.0
921811.138|sip  |0|00|    Via: SIP/2.0/UDP 10.252.149.61;branch=z9hG4bK1c322f93A6350AA9
921811.138|sip  |0|00|    From: "100" <sip:100@10.252.149.61>;tag=50925BEB-F7B9A3F1
921811.138|sip  |0|00|    To: <sip:100@10.252.149.100>
921811.138|sip  |0|00|    CSeq: 1 INVITE
921811.138|sip  |0|00|    Call-ID: d54220fb-c29874a1-966e0a23@10.252.149.61
921811.138|sip  |0|00|    Contact: <sip:100@10.252.149.61>
921811.138|sip  |0|00|    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
921811.138|sip  |0|00|    User-Agent: PolycomVVX-VVX_410-UA/5.3.0.12074
921811.138|sip  |0|00|    Accept-Language: en
921811.139|sip  |0|00|    Supported: replaces,100rel
921811.139|sip  |0|00|    Allow-Events: conference,talk,hold
921811.139|sip  |0|00|    Max-Forwards: 70
921811.139|sip  |0|00|    Content-Type: application/sdp
921811.139|sip  |0|00|    Content-Length: 342
921811.139|sip  |0|00|    
921811.139|sip  |0|00|    v=0
921811.139|sip  |0|00|    o=- 39383 39383 IN IP4 10.252.149.61
921811.139|sip  |0|00|    s=Polycom IP Phone
921811.139|sip  |0|00|    c=IN IP4 10.252.149.61
921811.139|sip  |0|00|    t=0 0
921811.139|sip  |0|00|    a=sendrecv
921811.139|sip  |0|00|    m=audio 2254 RTP/AVP 9 102 0 8 18 127
921811.139|sip  |0|00|    a=rtpmap:9 G722/8000
921811.139|sip  |0|00|    a=rtpmap:102 G7221/16000
921811.139|sip  |0|00|    a=fmtp:102 bitrate=32000
921811.139|sip  |0|00|    a=rtpmap:0 PCMU/8000
921811.139|sip  |0|00|    a=rtpmap:8 PCMA/8000
921811.139|sip  |0|00|    a=rtpmap:18 G729/8000
921811.139|sip  |0|00|    a=fmtp:18 annexb=no
921811.139|sip  |0|00|    a=rtpmap:127 telephone-event/8000

 

This is now going into support territory so please work with support on this as there is only so much I can do within the community.

In order to raise a support ticket you need to work with your Polycom reseller as they need to do this for you.

If this is some sort of an Internet discounter please post either your phone's MAC address or your Polycom devices serial so I can look up who would be able to support you. This may not be who you purchased the Polycom device from.

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

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

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 8
Visitor

Re: IP Dialing

Running the latest software, 5.5.2.8571.

 

In your testing, is your phone also registered to a server? I am able to URL dial if it's not registered in my testing. However if it's registered, the phone seems to just send the server it's registered with an INVITE with the IP address of the ATA and not directly send the ATA an INVITE.

Message 7 of 8
Highlighted
Polycom Employee & Community Manager

Re: IP Dialing

Hello socalpilot78,

yes I have Line 1 registered and Line 2 is not registered.

 

PagingSpeedDial.PNG

 

I already explained the next step aka contact support or wait until someone else may try and help you with this.

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

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

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 8