search for: 229390

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:6991 handle_request: Check for res for 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:1605 update_user_counter: is not a local user 2004-08...

...dont get asterisk to register with my sip provider (FWD)... when I start asterisk using the following command I get the following messages (first, a lot of messages show up immediatly after starting up: I'read this is normal, then the CLI console comes out and this messages appear): NOTICE[229390]: chan_sip.c:3922 sip_reg_timeout: Registration for 485445@fwd.pulver.com has timed out, trying again WARNING[229390]: chan_sip.c: 590 __sip_xmit: sip_xmit of 0x81445dc (len 366) to 192.246.69.223 returned -1: Bad file descriptor WARNING[229390]: chan_sip.c: 590 __sip_xmit: sip_xmit of 0x81445dc (l...

Hi all, There is a test version for the next release of spandsp at ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.2pre2 Only use this if you wish to experiment. In general, using spandsp-0.0.1 is probably more stable right now. If you have had problems with Canon FAX machines not accepting FAXes from spandsp-0.0.1 you might like to try this version. I have had at least one report from

Hi. I have installed a x100p (THE x100p for those who have seen my former post). Now I just want to connect a "normal" phone (not an IP phone) to the card and use it as a sip extension (I have a FWD account)... more clearly: I want to be able to pick up the phone and call any FWD user using my FWD account... receive the FWD calls in that phone, and also to be able to make normal

...t startup asterisk shows in the log : Nov 17 20:02:13 NOTICE[16384]: SCCP channel driver up and running on 10.10.0.1:2000 Nov 17 20:02:13 NOTICE[16384]: SCCP listening on 10.10.0.1:2000 Which hopefully means that SCCP is working There are no messages in the log other than: Nov 17 20:10:20 NOTICE[229390]: Alarm Message: Severity: 2, Name=SEP000770735049 [0/1711278602] Nov 17 20:10:20 WARNING[229390]: Client sent IpPortMessage without first registering. Is this an error in my asterisk config or is there something wrong with my config on the Kirk IP600?? Any help / input greatly appreciated.

...ve seen the fax detected messages coming for every call that comes in over my E&M T1[B8ZS ESF]. This isn't a problem from a functionality standpoint(I have no fax devices), but I see these two lines show up in the DEBUG logs about every second of a live incoming call: Jan 12 11:32:18 DEBUG[229390]: DTMF digit: f on Zap/1-1 Jan 12 11:32:18 DEBUG[229390]: Fax already handled I have tried uncommenting OLD_DSP_ROUTINES in dsp.c and recompiling, restarting, but then the E&M lines are useless(the lines are there and are loaded, but no inbound/outbound calls go through) Any ideas? Thanks,...

Anyone having problems with Stanaphone registration today? I'm getting the following.. Nov 25 11:35:58 NOTICE[229390]: chan_sip.c:4053 sip_reg_timeout: Registration for 'xxxxxxxxxx@216.128.82.18' timed out, trying again -- Got SIP response 500 "Internal Server Error" back from 216.128.82.18

...ple dial the PBX S0 with Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b] -- Accepting AUTHENTICATED call from 192.168.255.1, requested format = 8, actual format = 8 -- Executing Dial("IAX2/ogo@call_saar/1", "Zap/g1/61|60") in new stack -- Called g1/61 Sep 23 10:27:50 NOTICE[229390]: chan_zap.c:8725 pri_dchannel: SETUP_ACK for 'Zap/1-1' -- Accepting overlap call from '' to '<unspecified>' on channel 0/2, span 1 -- Starting simple switch on 'Zap/2-1' ^^^^^^^^ here I get a dialtone...

Hi list, Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? Thanks Eduardo