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Valcom IP Solutions Setup Tool

Warning

Unfortunately at this time, the Valcom SIP software is not compatible with OnSIP at this time. There are two points of failure.

The first is that the Valcom software does not do DNS lookups for the outbound proxy server. The OnSIP network has a distributed proxy server which relies upon GeoDNS information to send traffic to the most efficient server location relative to the device registering. The Valcom software will only allow you to input a specific IP address rather than "sip.onsip.com" for this field. We provide a workaround for this error in our configuration, but this is not meant to be an endorsement of the product.

The second issue is that the Valcom software will supplant information in the "To:" field not allowing users to dial directly to the device. This means that you cannot set up a Valcom speaker with an extension directly as any attempt to dial an extension to reach the speaker to make your announcements. We have a workaround for this as well.

Here is an example call where you can see the Valcom software supplant and then reject the direct invite.

**Initial inviate from Hiros's phone at port 1034 to OnSIP at 199.7.175.101:5060

2011-09-14 16:35:29.555016
173.186.100.90:1034 -> 199.7.175.101:5060

INVITE sip:888@example.onsip.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.1.101;branch=z9hG4bK2851d7253CB6B7D6
From: "Hiro Protagonist" ;tag=9EC3133B-B270725C
To: 
CSeq: 1 INVITE
Call-ID: e6ca24af-f01fd7f0-6253c5d9@10.0.1.101
Contact: 
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.3.1.0769
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 243

v=0
o=- 1316016925 1316016925 IN IP4 10.0.1.101
s=Polycom IP Phone
c=IN IP4 10.0.1.101
t=0 0
a=sendrecv
m=audio 2232 RTP/AVP 9 0 8 127
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000

_________________________________________________________________________________

**OnSIP Reply to Hiro's phone with "trying" acknowledging the receipt of the INVITE.

2011-09-14 16:35:29.558680
199.7.175.101:5060 -> 173.186.100.90:1034

SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 10.0.1.101;branch=z9hG4bK2851d7253CB6B7D6;rport=1034;received=173.186.100.90
From: "Hiro Protagonist" ;tag=9EC3133B-B270725C
To: 
CSeq: 1 INVITE
Call-ID: e6ca24af-f01fd7f0-6253c5d9@10.0.1.101
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0


_________________________________________________________________________________

**INVITE from OnSIP (66.227.100.25:5060) to the Valcom device at 173.186.100.90:1036

2011-09-14 16:35:29.593899
66.227.100.25:5060 -> 173.186.100.90:1036

INVITE sip:valcom@10.0.1.235:5060;aor=valcom%40example.onsip.com SIP/2.0
Record-Route: 
Record-Route: 
Via: SIP/2.0/UDP 66.227.100.25;branch=z9hG4bK7e9e.f6c43a82.0
Via: SIP/2.0/UDP 199.7.175.101;branch=z9hG4bK7e9e.556ae8b4.0
Via: SIP/2.0/UDP 10.0.1.101;rport=1034;received=173.186.100.90;branch=z9hG4bK2851d7253CB6B7D6
From: "Hiro Protagonist" ;tag=9EC3133B-B270725C
To: 
CSeq: 1 INVITE
Call-ID: e6ca24af-f01fd7f0-6253c5d9@10.0.1.101
Contact: 
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.3.1.0769
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 243

v=0
o=- 1316016925 1316016925 IN IP4 10.0.1.101
s=Polycom IP Phone
c=IN IP4 10.0.1.101
t=0 0
a=sendrecv
m=audio 2232 RTP/AVP 9 0 8 127
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000

_________________________________________________________________________________
** Response from Valcom device at port 1036

2011-09-14 16:35:29.661897
173.186.100.90:1036 -> 66.227.100.25:5060

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 66.227.100.25;branch=z9hG4bK7e9e.f6c43a82.0
Via: SIP/2.0/UDP 199.7.175.101;branch=z9hG4bK7e9e.556ae8b4.0
Via: SIP/2.0/UDP 10.0.1.101;rport=1034;received=173.186.100.90;branch=z9hG4bK2851d7253CB6B7D6
Record-Route: 
Record-Route: 
From: "Hiro Protagonist" ;tag=9EC3133B-B270725C
To: 
Call-ID: e6ca24af-f01fd7f0-6253c5d9@10.0.1.101
CSeq: 1 INVITE
Content-Length: 0

_________________________________________________________________________________

**We ACK the 404 back to the Valcom device.

2011-09-14 16:35:29.662390
66.227.100.25:5060 -> 173.186.100.90:1036

ACK sip:valcom@10.0.1.235:5060;aor=valcom%40example.onsip.com SIP/2.0
Via: SIP/2.0/UDP 66.227.100.25;branch=z9hG4bK7e9e.f6c43a82.0
From: "Hiro Protagonist" ;tag=9EC3133B-B270725C
Call-ID: e6ca24af-f01fd7f0-6253c5d9@10.0.1.101
To: 
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0
_________________________________________________________________________________

You can reach the Valcom device if you call it by the SIP Address. This is easy enough with a softphone, but more difficult to do with a desk phone since a desk phone typically dials extensions, not SIP addresses.  A call to extension 7999 for example, does not work for Valcom directly.

The workaround is to go to "Apps" and create an announcement which when terminated will transfer to the device and assign the desired extension of the Valcom unit to that announcement. Then a user may dial the extension which will call the announcement which will transfer the call to the Valcom device's SIP address. Please be aware that there is a $4.95 a month charge for an announcement and that you will not be able to call the device directly from the PSTN. That's a minor point as most paging is internal only.

SIP address and extensions

Step 1: Gather information for each user.

Each user has a set of credentials which will be needed to configure each phone. For each phone that you are configuring, obtain the following:

  • "SIP Address" (Address of Record)
  • "SIP Password"
  • "Auth Username"
  • "Username"
  • "Proxy/Domain"

You can find this information in the user detail pages under the "Users" tab in the "Phone Configuration" section.

Phone configuration section

Step 2: Configure your Valcom software settings.

Under the "SIP" tab, enter the following information from Step 1 above:

User Details

  • Phone Number> "Username"
  • Description> However You Want to Distinguish This Device
  • Authentication Name > "Auth Username"
  • SECRET > "SIP Password." This will remain unencrypted
  • Realm > sip.onsip.com
  • SIP Server > sip.onsip.com
  • "Outbound Proxy" > 199.7.173.101
  • Register > Make certain this box is checked

User details

Step 3. Confirm that your phone is registered.

In the User portal, click on the "Users" tab. You will see a green "online" notation next to each user with a registered phone.
You should now be able to place and receive calls.

Download the 2017 Business Phone Guide

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