Valcom IP Solutions Setup Tool
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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.
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.
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.
Step 2: Configure your Valcom software settings.
Access the software settings. On the Mac, from the menu, choose Bria > Preferences
the click on "Accounts". In Windows, from the Bria menu choose Softphone > Account Settings.
Under the Account 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
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.
