Tag : sip
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All,I set up Homer SIPCapture and Captagent 6 on Asterisk box. All works fine but SIP records are duplicated.I tried to user extra filter and not src host into captagent config but no success.Can you point me how to figure it out?Kind reg..
all, sorry if the subject is a bit confusing, but I just couldn’t think of a good way of better describing the situation… Basically, I travel a lot and have several SIM cards for my phone from local carriers. What I’d like to do now is to se..
I am installing Asterisk in a small office with just 4 lines and 8 Extensions. I have two choices from my local telco (Fairpoint): 1) Old School ISDN BRI lines which I would connect to Asterisk with a OpenVOX B200P2) Telco supplied SIP trunks ove..
everyone,I am sending out a multicast page using the following in my dialplan:Page(MulticastRTP/linksys/${MULTICAST_IP_ADDR}:6061/${MULTICAST_IP_ADDR}:6061,q)Everything works great, but I had a question about SIP and SDP:Should I be seeing a SIP/..
Olle,Redirecting the question to users mailing list. Could you point out how can I *dynamically* pass both the SIP peer and request-URI in the dial command. I want be able to use same SIP peer to route to different SIP end points. Im currently do..
!I have a problem where SIP packets sent by Asterisk do not hit the wire, andI dont know what could cause this.Im running Asterisk 1.8.28_cert5 with full SIP debug. At the same time, Imdoing a tcpdump of the traffic on the network interface. I can ..
We currently have a production Asterisk box running 1.8.20.1 which uses MeetMe and chan_sip for conferences. I have been testing the new versions of Asterisk with PJSIP and ConfBridge but have run into an issue which is preventing us from moving forwa..
I need helpThis is the error:Really destroying SIP dialog NDMxOWRmYTRhMWVkMGFhMjllMzU4YmNmNjQwN2NlM2Y.Method: SUBSCRIBE– Executing [00919885497796@internal:1] Set(SIP/1001-0000000b,CAL..
Ive read SIP Connect 2.0 draft lately.It mentions specific use if either of the following values is present in the From: field of an INVITE message. The values are:sip:unavailable@unkown.invalid sip:anonymous@anonymous.invalidIm using Asterisk 13 ..
Greetings.I am using the PJSIP driver with TLS transport, and my endpoints are SIP mobile apps operating in environments that I do not control.I would like Asterisk to default to sending INVITES and all other SIP signals to endpoints via the exist..