Tag : sip
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Im trying to setup a sip trunk. The asterisk version is Asterisk certified/11.6-cert12. I had to change the CallerID to a pilot number of the trunk as provider only accept it. but when i dial Im getting below error. Cannot hear anything[Feb 11 14:19:..
so this is not Asterisk related maybe, only in so much, that one of the systems involved is a SIP PABX running on one…There are actually 2 parallel running systems on my site: one is SIP based, with hardphones and softphones; and the other is Siem..
Ive got a request from a prospective customer demanding a SIP hardphone able to provide 120 BLF to its operator.Each BLF should show current extension activity (blink when ringing, …)and allow speed dialing.Beside finding hardware matching these requiremen..
ive got calls coming into an 11.21.0 box. The internal phones are analogue off a TDM400 board, and SIP extensions.Using an analogue internal phone, the remote party always hears an echo on its side. We do not hear an echo. Doesnt matter who is the call..
Hello Lets assume we have this situation: Call => SIP TSP => Asterisk1 => IAX2 => Asterisk2 => SIP/ATA => Fax I have two Asterisk Servers in two branch offices, which are interconnected by IAX2 and the Switch functionality. Asterisk1 is connected..
I run an ISP with virtual services.We also offer VoIP.Currently we have our DNS set up so that virtual domains can have SIP addresses in their own domain.However, it comes to Asterisk as just the user name.For example, my SIP address is sip:darcy@VybeNetworks…
Hi! I wish you all e Happy New Year first! Allthough, Im relative new to Asterisk, I got our server up and Running, Softphones, ISDN, and a brand new Snom 821 are working flawlessly. 🙂 Platform is Debian 8/Asterisk Packages (11) from Debian Repo. ..
Happy new year!maybe off-topic… but maybe someone _knows_ a solution.I have a free SIP Account at linphone.org… calling other linphone.org users via SIP and receiving SIP calls from other users registered at linphone.org is no problem… just d..
Im having a strange problem with Asterisk 13 i cant seem to find out whats causing it. After a Dial call from one SIP peer to another, if the calling side hangs up, DIALSTATUS is not set, but when the called side hangs up, it does. The strangest th..
In setting up the GS-Wave softphone there are two id entries:SIP User IDSIP Authentication IDI would have thought SIP User ID was the devicename , i.e. [name].Then SIP Authentication ID was defaultuser.But not so. With[gs_5062](cell-phones)defaultuser=gs_62and..