search for: siptester

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 "verbose 10" setting shows [Mar

...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 An...

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

Hi Sorry if this comes twice; i sendt first version from non-member address. I'm learning use Asterisk but cannot solve following problem: i have Asterisk v1.0.7 (DEbian) and 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

Hello, I am searching for a DTMF issue on my setup ( 2 years and counting ), and I am wondering why rtp.c has code to mute DTMF ( the rtp->dtmfmute variable ), but this same mechanism does not exist in dahdi. I am sending a DTMF over SIP w/ RTP & RFC2833 to the asterisk box with the dahdi card. The dahdi card sends it out on the PRI line. Trouble is, the DTMF is echoed back and the

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

Hello I have changed type 'caldav' to 'ical', but still no succes : [Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:117 auth_credentials: Invalid username or password for iCalendar 'cal1' [Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:150 fetch_icalendar: Unable to retrieve iCalendar 'cal1' from

All, I made some progress, but it seems the further I go with clustering the harder things get. Hmmm, I guess if it were easy, it would be documented...... Anyhow, I have 1 * server as the DUNDi peering master with a ttl=1. The only function of this server is to lookup where other sip peers are registered and forward that info on to the requesting * server. I have 4 * servers accepting

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

-- SIP/ vaso -e26c answered Zap/14-1 -- Executing DumpChan("SIP/ vaso -e26c", "") in new stack -- Executing DumpChan("SIP/vaso-e26c", "") in new stack Dumping Info For Channel: SIP/vaso-e26c: ============================================================================ ==== Info: Name= SIP/vaso-e26c Type=