search for: broadsoft

Greetings to all. I am not sure of this is a "user" question or a "business" so apologies in advance if it should be asked in the business list. A client of mine has a UK branch that is served by a provider that uses the Broadsoft solution. I want to create a sip trunk from a remote asterisk pbx to the client. The provider claims that Broadsoft has a strict "protocol" on what "devices" can connect on their network and asterisk is not listed Does anyone know if asterisk (1.8 to be more precise) has pass...

I am looking for a sample sip configuration of a SIP provider that runs Broadsoft VoIP switch. My sip provider is Conecta from Brasil, that only give me a SIP IP address to configure my asterisk box, when I call them for support or authentication data to load on my sip.conf, they say me that I don?t need such data, so, anyone knows how I would configure my Asterisk box against...

Hello, I've seen several references in IP phones manuals to Broadsoft's BLF-List URI feature (also referred to as List-Oriented BLF). With this mechanism, a server is able to update the BLFs an IP Phone is supervising without asking the IP phone to reboot, as for a reason I don't know, most phones send BLF-related SUBSCRIBEs during boot time. Is this featur...

...the switch in the same way) - 8 pc's running XP Pro, all plugged into the switch port on the back of the IP-500's Location 2: - Full rate data T1 - Dell PowerEdge SC1420 (Since replaced with a clone pc) with no TDM hardware at all (this location connects SIP directly to the T1 providers Broadsoft switch and does not go over the public internet) - Asterisk 1.0.7 (AAH 0.9) - using SIP to connect with my provider (not across public internet, not natted since the Cisco IAD does the SIP mangling for us) - Average ping time to the broadsoft switch: 42 ms - 8 Polycom IP-500's running SIP 1.4.1...

...c, and he referrred me to Roger Austin, Regional Channel Manager for Voice. Roger's reply to my inquiry is as follows: "Cory, We appreciate your interest in Polycom VoIP Phones. Polycom deploys our VoIP phones with our VoIP Platform partners and at this time those Partners are Sphere, Broadsoft, Sylantro, and Interactive Intelligence. Unfortunately we are not supporting the Asterisk solution at this time." I am going to continue to pursue this, but this is the pushback I have gotten thusfar. We are a Polycom authorized reseller, and have compiled some pretty detailed documentatio...

...OK Via: SIP/2.0/UDP 83.30.64.252:5060;branch=z9hG4bK70ac947c From: "asterisk" <sip:asterisk@xxx.xxx.xxx.xxx>;tag=as0de7b82c To: <sip:sip.broadvoice.com>;tag=SD384v999- Call-ID: 45767cf0222b692d27bfa4633e09a735@83.30.64.252 CSeq: 102 OPTIONS Accept: application/sdp,application/broadsoft,text/plain Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE,NOTIFY,UPDATE Supported: 100rel Content-Length: 0 I'm registered with few others suppliers, SIP and IAX2 without any problem. Have someone an idea on what's happend? My ping time is 140 ms. Than...

Hi all I'm trying to get asterisk 10 spandsp get faxes from 012 in israel (they use broadsoft switches) using T.38 more reliable and would like to know if anyone knows of any changes I could make or ask them to make. As it stands now I get much more reliability receiving faxes with iaxmodem with no T.38 which is funny than when having asterisk receive using T.38 and if I try using gateway...

...5060;received=165.166.232.49;branch=z9hG4bK60286504;rport=5060 From: "asterisk" <sip:asterisk@192.168.1.108>;tag=as6d521e51 To: <sip:sip.broadvoice.com>;tag=SD3giuc99- Call-ID: 0a72198652f1d9677bbb1c19350ec6f9@192.168.1.108 CSeq: 102 OPTIONS Accept: application/sdp,application/broadsoft,text/plain Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE,NOTIFY,UPDATE Supported: 100rel,timer Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '0a72198652f1d9677bbb1c19350ec6f9@192.168.1.108' <-- SIP read from 147.135.0.128:5060: SIP/2....

Still looking for some direction with this subject: I think the term is called multi-line appearance.... Is this something that is directly supported in Asterisk? I can't seem to find any information on it or how to actually use it.... This is where you have several sipura-841 SIP phones for example... if someone pickes up 'line1' I'd like the light to come on on ALL phones to

Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most

Is there any "easy" way to add a custom subscribe handler? I have a set of users with Polycom phones that attempt to Events that Asterisk/PJSIP doesn't recognize, "call-info" and "as-feature-event". It just generates a warning, but it got me wondering if I could add my own handlers for those that didn't actually do anything but simply responded with a 200 OK.

I recently got a SNOM 360 and have been trying to get the extension lights to work. I can see the subscriptions with sip show subscriptions but I don't see any notifies when a call is made. I must be missing something because I've tried looking to see if anyone else has had this problem but the only solutions I've seen have been to put hints in and I have those. Any suggestions?

I set up hints and presence monitoring on some Polycom phones connected to an asterisk server with the expectation that the phones that are "watching" other extensions would be notified when the other extension sis ringing, in addition to the other statuses (on the phone, statuses set by the user on the phone, not registered, etc). I can see when the line is in use, and when it is