Posts tagged SIP
SIP Software Release 3.0 for IP Deskphones
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Avaya has released SIP software release 3.0 for their 1120E and 1140E IP deskphones. (There was no mention of the 1110E, 1150E, 1165E or 1200 series IP phones in any of the accompanying material).
Several enhancements have been included in SIP Release 3.0 for the 1100 series phones including User Interface and Preferences enhancements, Multi-user Login, Emergency Services support, USB device support, Wide-band Codec, Provisioning and Licensing.
The SIP software Release 3.0 for IP Deskphones also continues to improve the overall quality of the IP Deskphone software through the delivery of ongoing resolution of CRs. Numerous quality improvements have been delivered and 9 customer cases have been closed in SIP 3.0.
I’ve only performed very limited SIP testing with the 1120E, 1140E, and 1220 IP phones in non-production environments. I did notice a few feature called “Multi-user Login” which allows a SIP IP phone to connect to multiple SIP servers at the same time. Here’s the blurb from Avaya on the feature (it’s a direct quote from the release notes);
Multi-user Login
The Multiuser feature in SIP Release 3.0 allows multiple SIP user accounts to be in use on the IP Deskphone at the same time. Multiple users, each with their own account, can share a single IP Deskphone allowing each user to receive calls without logging off other users. One user can have multiple user accounts (for example, a work account and a personal account) active at the same time on the same IP Deskphone. You can register each account to a different server, and for each account, the IP Deskphone exposes the functionality available to that account. One account is considered a primary account and is used by default for most IP Deskphone operations. Each account is associated to a line key; the primary account is always on the bottom right line key of the IP Deskphone (this is the first key, Key 01), and an arbitrary key (including a key on an Expansion Module) can be selected for additional accounts.
The following operations are supported:
- Start dialing
- Place a call using the corresponding user account
- Answer an incoming call targeted to that account
- Initiate a call without pressing a line key (for example, by dialing digits at the idle screen and lifting the handset) uses the primary account.
A running IP Deskphone is associated to a single profile that represents one configuration of the IP Deskphone with all relevant persistent data such as preferences and call logs. A different profile is associated to each account used as a primary account. The IP Deskphone can store up to five different profiles; the IP Deskphone takes data from the profile associated to the current primary account. A number of configurations are independent of profiles and tied directly to an account making them available to that account regardless of the primary account you use (for example, voice mail ID).
The IP Deskphone receives and answers calls targeted at any of the registered accounts; the incoming call screen indicates who the call is for. You can place an outgoing call using any of the accounts; the account that you use is displayed on the dialing screen. When a call is active, information from both local and remote parties appear on the screen.
Regardless of which account receives the call, incoming call logs, outgoing call logs, and instant messages appear in a single list. The IP Deskphone indicates the local user in the detailed view of the entry.
Some features are only available to the primary account, such as instant messaging, retrieving parked calls by token, and establishing ad-hoc conference calls.
Please refer to the product bulletin and the release notes for all the details.
Cheers!
BCM50 SIP trunking for PSTN access
0I recently received an email message asking me how to configure the BCM50 for SIP trunking to PSTN providers.
Thankfully Avaya/Nortel has already provided plenty of configuration examples for a number of well-known carriers.
2010-00000227_1.1_BCM50_BCM450_R5_Configuration_Verizon.pdf
NN10000-103_Ver_1.4_Nortel_BCM50_3.0_SIP_Config_Guide.pdf
2009-00002459_1.0_BCM_5.0_Configuration_Guide_Skype_SIP.pdf
2010-00000229_1.0_M50R3_M450R1_Configuration_Guide_For_Bell.pdf
2010-00000219_1.0_Config_Guide_Bell_SIP_Trunking.pdf
2009-00002460_1.1_BCM ConfigurationGuide_PAETEC_SIP_Trunking.pdf
Have a look at the above documents if you are searching for how to configure the BCM50/BCM400 for SIP trunking to a public provider/carrier for PSTN access.
Cheers!
Voice Pulse – IP Phone Service
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With AT&T CallVantage soon to be canceled I had to go searching for an alternate solution for my home phone service. I contemplated going with Verizon’s Triple Play since I already have Verizon FiOS Internet and Verizon FiOS TV but in the end I decided to go with Voice Pulse.
I ordered the service online and it took about a 11 days for the Linksys PAP2 to arrive. I had to call four days after I placed the order to find out the status since there was no email message informing me that the equipment was back ordered).
The installation of the Linksys PAP2 was quite easy. I just connected it to the Verizon Actiontec router and plugged the RJ11 jack into my phone. Within seconds I had dial tone from the Linksys PAP2. I didn’t need to make any changes to the Verizon Actiontec router although it might be necessary later to apply some QoS settings.
It took 7 days to port my original AT&T CallVantage phone number to Voice Pulse. Prior to porting my phone number I just setup CallVantage to forward all calls to the temporary number assigned by Voice Pulse.
There are an amazing number of call routing and call filtering features including telemarketer block which promises to block automated and computerized dialing services used by a vast number of telemarketing companies.
So far the service has been great and very reliable. And would you know that I actually received a phone call from Voice Pulse confirming the port of my home phone number. And I even spoke to an actual human being that I could clearly understand. Did I mention that they called me?
If you have reliable Internet broadband and your looking for a good Internet phone provider you won’t go wrong with Voice Pulse. Voice Pulse will also provide you with SIP trunks for your Asterisk deployment.
Cheers!

