Tag : sip
Group Members,I have one question regarding SIP-I/SIP-T support in any of Asterisk versions.We have client which send SIP-I/SIP-T request can asterisk handle it and serve as a normal SIP call.As per mine analysis SIP-I/SIP-T are variant of SIP proto..
Here is my scenario. I have a SIP call between two SIP endpoints. A calls B. During the ringing, B disconnects (network cable is unplugged).But A continue ringing forever (until the dial timeout) even if asterisk detects that B is disconnected with ..
I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configurat..
I havent been able to find the answer online, and am not currently able to conduct an experiment to find the answer…I understand that in a SIP call where G729 has been negotiated as the preferred codec, a G.729 licence is not consumed until there..
all We have an Asterisk s..
,I have a fresh install of Asterisk 12.0.0 and Im going to use it only as a client. Im trying to SIP REGISTER with a remote SIP provider.The situation is that Asterisk is running in a VMware VM with a RFC IP address (192.168.1.2). The provider of ..
Greetings-I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do have qualify=..
I am using voip with Vodafone as SIP peer for outbound telephony and i have a huge problem establishing calls to other people. It works like in 1 of 5 tries. The peer is sending SIP 480 temporarily not available. It took a while to identify this, beca..
If SIP channel driver needs to connect to a remote GW over a dedicated SIP trunk BUT the remote GW has a standby in case of failure, how can the sip configuration file be configured for the remote GW when there are actually two IP addresses. If the m..
All,Ive asked this on the asterisk-dev list, so sorry for cross-posting. So far Im not sure how to accomplish this without looking at the source code or looking at some other way to get around this issue.Im trying to have an automated call to an Aas..