search for: siptest

Hi List I am having trouble running the command siptest:~# asterisk -rx 'dialplan reload' most times it does what I expect and I get a response as below siptest:~# asterisk -rx 'dialplan reload' Dialplan reloaded. every now and then I get no response i.e. siptest:~# asterisk -rx 'dialplan reload' siptest:~# and a "...

...39;t accept the ACK and insists on retransmitting the 491 Response. Asterisk replies with the following 491 response: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 10.110.7.89:5070;branch=z9hG4bK5a668c33f196837c3602266b23b389e0;received=10.110.7.89 From: <sip:30001 at 10.110.7.20:5070>;tag=SIPTester To: <sip:30008 at 10.110.7.20>;tag=as2ea72122 Call-ID: 0dd43bb5a64eb5a2fb0114193821f037 at 10.110.7.89 CSeq: 5 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal...

Hi All, I'm working with regcontext and sip users/peers. In the wiki, the example shows you can put this parameter in the [sipuser] context, like so: [general] lots of general parameters [sipuser] regcontext=siptest regexten=1234 Now this does not create the Noop exten priority 1 in the dial plan when the sip user registers. Now if I put regcontext in the [general] section, the sip user registers and the Noop exten priority 1 does get entered into the dial plan: [general] regcontext=siptest [sipuser] The...

...Linphonec v1.2 (Debian). Every time i try to register within LAN i got 'Forbidden' message from Linphonec. Where to start searching for reason for this failure, is there more debuggin options available? Here is relevant part of sip.conf: [fujitsu] type=friend username=arimo context=siptest secret=n*****i host=dynamic sip debug output: Sip read: REGISTER sip:arimo.iki.fi SIP/2.0 Via: SIP/2.0/UDP 192.168.1.67:5060;rport;branch=z9hG4bK769868031 From: <sip:arimo@arimo.iki.fi>;tag=529821486 To: <sip:arimo@arimo.iki.fi> Call-ID: 230459319@192.168.1.67 CSeq: 1 REGISTER Contact...

...hinks there is a an inbound DTMF happening : [2011-06-02 21:05:41.847] DEBUG[1333] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [2011-06-02 21:05:41.847] DEBUG[1333] rtp.c: Sending dtmf: 52 (4), at 192.168.1.5 [2011-06-02 21:05:41.847] DTMF[1333] channel.c: DTMF begin '4' received on SIP/siptest-0000000c [2011-06-02 21:05:41.847] DTMF[1333] channel.c: DTMF begin passthrough '4' on SIP/siptest-0000000c [2011-06-02 21:05:41.847] DEBUG[1333] chan_dahdi.c: Started VLDTMF digit '4' [2011-06-02 21:05:41.847] DEBUG[1333] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [2011-06-02 21...

I've got a one-way audio problem, but I've looked through a few documents on the subject and I'm not sure that it's the same issue. User A calls a local Asterisk user B via a public SIP gateway (voiptalk.org) using (sip:110@siptest.dmclub.net) B is connected to the Asterisk server via VPN B is registered (and has successful bi-directional conversations with other users on the VPN) Asterisk correctly forwards the call via SIP and B's phone rings and is answered, but B can't hear A So there appears to be an audio-...

...for iCalendar 'cal1' [Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:150 fetch_icalendar: Unable to retrieve iCalendar 'cal1' from 'https://www.google.com/calendar/dav/info at mydomain.tld/events/': Could not authenticate to server: rejected Basic challenge siptest*CLI> calendar show calendars Calendar Type Status -------- ---- ------ cal1 ical free Am I missing something obvious here ? Kind regards Jonas On 26-10-15 17:02, Marek ?ervenka wrote: > try ical url > > caldav switched...

...the phone to register and dumps the sip phone into sip show peers, works great. I can take the sip entry out of the database and the phone will not resister in realtime. Works great. Now the dial plan setup. All the extension info is also in the MySQL database, I have a switch statement in the [siptest] context pointing to the database for extension logic. This also works great. All servers are pointing to the same data source with all sip extensions in the database starting with exten => 1234,2,Answer and so on exten => 1235,2,Answer and so on notice the priority 2 starting point in th...

I have a test SIP account set up with WorldCom and I have been trying to have Asterisk register to the WorldCom server with no luck. It appears that the SIP headers are different coming from Asterisk. I have included a packet capture from a successful login with a Windows Messenger client for reference. I have also copied in the SIP packet I captured with sip debug turned on. In my sip.conf file,

Hello I find very little feedback on the following warning/error when trying to connect to Google calendar : [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:118 auth_credentials: Invalid username or password for CalDAV calendar 'cal1' [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar cal1, request REPORT to

...ime= 0h0m0s Context= macro-testing Extension= s Priority= 1 CallGroup= PickupGroup= Application= DumpChan Data= (Empty) Blocking_in= (Not Blocking) Variables: MACRO_DEPTH=1> ARG1=30086 MACRO_PRIORITY=1 MACRO_CONTEXT=siptest ============================================================================ ==== -- Executing AGI("SIP/vaso-e26c", "whopicked.rb") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/whopicked.rb AGI Tx >> agi_request: whopicked.rb AGI Tx >> a...