system toolbox » voip

dosta informacija o siemens telefonima, kao i o povezivanju sa asterisk-om na ovom forumu, http://forums.whirlpool.net.au/forum-replies-archive.cfm/927308.html

na #13925 sam u features listi našao:

• Up to 3 calls in parallel: 2 VoIP calls and 1 fixed-line call with multiple handsets*

da li ovo znači da sa ovim telefonima preko jedne baze samo mogu da se obavljaju dva simultana voip razgovora ?

u tom slučaju možemo ići samo na kombinaciju 1 telefon sa bazom + 1 dodatna slušalica = 2 lokala.

Za četiri radna mjesta onda trebamo dva ovakva kompleta

na gore pomenutom forumu čovjek prenosi svoje iskustvo o konekciji tri slušalice na asterisk:

Connecting to Asterisk¶

All I did in Asterisk was define 3 more SIP extensions, intending them to correspond to the 3 handsets. In the Siemens C470IP all I did was connect via a web browser and went to the Settings -> Telephony -> Connection section. In that section there was space to enter 6 SIP providers. There was also one fixed configuration for Gigaset.net, which I disabled and might experiment with later on. All I had to do was to define 3 SIP providers being the three new extensions that I just configured in Asterisk.

It was really quite straight forward. The only thing different to what I've seen in others is that it has settings for domain, proxy server, registrar server and outbound proxy. I had to set them all to the address of my Asterisk system.

The interesting thing in the C470IP is the way they treat the SIP providers. Each provider, including the PSTN, can be accessed from any handset by adding a suffix at the end of the dialed number, #2 for example will force it use the 2nd SIP provider. This is not quite what I want as I want each handset to be tied to only one SIP provider, being the extension in Asterisk. To do this you need to go into the Settings > Telephony -> Number Assignment section. Here each handset can select one of the SIP providers for out going calls, and any number of the SIP providers for incoming calls plus the PSTN and Gigaset. This means each handset must choose one and only default provider to make outgoing calls, but can have any number of providers for incoming calls (from none to all), so that the handset rings when any of those providers gets an incoming call. Apparently all this can be overridden if there was dialplan as well. I will test all this at later when I test with direct connections to the VSPs without the Asterisk box.

So In my case I would select handset 1 to have SIP provder 1 as the outgoing, and only one selection for the incoming, being SIP provider 1. Similarly, I would do the same for the other 2 handsets. It's all very easy to do on the Web GUI with radio buttons and checkboxes.

ostaje samo pitanje o broju simultanih poziva

naš simens koja imamo za testiranje ima izgleda iste karakteristike u ovom pogledu

na #13122 Gigaset C455 IP - Technical Data piše

• simultaneous calls (conferencing) 2 VoIP and 1 PSTN Call simultaneously (+ 1 internal call or AM acces; VoIP resp. PSTN 3 party conference (1 external/ 2 internal participants) ; Two 3-party-conferences simultaneously (1 external VoIP or PSTN/2 internal participants each + AM access)
• Transfer of external calls to another handset yes
• Phonebook transfer between handsets yes

ali on ovdje govori conferencing-u ( simultaneous calls (conferencing) ) sad nisam načisto. ostaje pitanje:

Da li sa 3,4 slušalice možemo istovremeno obavljati 3,4 razgovora ?