search for: 180236

...9;s a problem -- autoconf is clearly not working, or there's some other related issue. So I try to use the demo and do "dial 500". This should connect and play the demo's intro file. I've gotten this to work in the past. Instead I get the following warnings: > WARNING[180236]: File file.c, Line 418 (ast_openstream): File demo-abouttotry does not exist in any format > WARNING[180236]: File file.c, Line 561 (ast_streamfile): Unable to open demo-abouttotry (format 64): Success > WARNING[180236]: File app_playback.c, Line 83 (playback_exec): ast_streamfile failed on...

I have just set up my first Asterisk, and I have the basics up an running. I have set it up with two SIP phones only. I can call between them, and I can call out to FWD phones. However, on receiving calls from FWD, my Asterisk blocks the calls with the following message: May 30 20:19:24 NOTICE[180236]: chan_sip.c:6397 handle_request: Failed to authenticate user <user> <sip:fwduser@fwd.pulver.com>. Obviously, I want FWD users to be able to call me without my registering them first. Any suggestions would be appreciated. Thor

...get a Wildcard T100P working with Asterisk and so far I haven't had any luck. No problems with the card itself from what I can see and the telco says that the problem is on our end (don't they always?). A small sample of what asterisk spews out on the console: Jan 13 15:45:32 WARNING[180236]: chan_zap.c:5834 zt_pri_error: PRI: Read on 43 failed: Unknown error 500 PRI got event: 5 Jan 13 15:45:36 WARNING[180236]: chan_zap.c:5834 zt_pri_error: PRI: Read on 43 failed: Unknown error 500 PRI got event: 6 Jan 13 15:45:36 WARNING[196621]: chan_zap.c:4781 handle_init_event: Detected alarm...

I've built the latest CVS of asterisk -- not the zaptel or libpri directories, just the asterisk directory. asterisk installs successfully, but there are severe problems. I built this system in the past and ran it, but now building it again fails. This is the CVS as of this morning, 2003-06-13, but I had problems on 06-11/12 as well. After make; make install; make samples; make config, I

I must be doing something stupid, but I can't figure it out. I have three PRI lines connected to Asterisk, one from the phone company, and two more connected to PBXs. Asterisk uses the incoming DID information to decide which PBX to route the call to. Should be simple. Asterisk is clearly getting the caller id info from the phone company: -- Accepting call from '512345xxxx'

For some reason I get the "Comedian Mail" prompts, I can move around in mail, "listen" delete save record prompts. BUT. When I try to leave a message, I hear no greeting, it claims to be recording in the debug window, but I get this error. Sep 15 19:10:08 ERROR[180236]: sccp_actions.c:624 sccp_handle_open_recieve_channel_ack: Device SEP000BBE94840A sent OpenChannelAck, but there is no active channel! It never records. The phone is using G711 as it's a Cisco 7960 running SCCP. Thoughts? -------------- next part -------------- An HTML attachment w...

...gnoring loadzone Sep 30 10:51:07 WARNING[16384]: chan_zap.c:9567 setup_zap: Ignoring loadzone Sep 30 10:51:07 WARNING[16384]: chan_zap.c:9567 setup_zap: Ignoring loadzone Sep 30 10:51:07 WARNING[16384]: chan_zap.c:9567 setup_zap: Ignoring defaultzone [New Thread 163851 (LWP 28296)] .[New Thread 180236 (LWP 28297)] [New Thread 196621 (LWP 28298)] .. -- Logging into database with user asterisk, password xxxxxx, and database asteriskvmusers ........[New Thread 213006 (LWP 28300)] ........................................Sep 30 10:51:12 WARNING[98311]: chan_sip.c:681 retrans_pkt: Maximum retries...

Just cross pairs 1-2 and 4-5 1 <--> 4 2 <--> 5 Never done an E1, but I think it's the same for a T1 4 wire cable. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: bam [mailto:bam@cqm.co.uk] Sent: Monday, December 29, 2003 9:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] E100P pinouts

...ICE[98311]: Registration from '<sip:021@192.168.1.92>' failed for '192.168.1.93' > Sep 13 08:33:11 NOTICE[262160]: Unable to create channel of type 'SIP' > Sep 13 08:33:11 NOTICE[262160]: Unable to create channel of type 'SIP' > Sep 13 08:36:39 WARNING[180236]: Detected alarm on channel 1: Red AlarmSep 13 05:40:50 NOTICE[16384]: registered database handle 'mysql1' dsn->[MySQL-asterisk] Also, from time to time something goes wrong (maybe one call in 5) while a call is in progress, and instead of hearing the person's voice, one hears garb...

...IAX Subclass: NEW Timestamp: 00001ms SCall: 00004 DCall: 00000 [146.231.125.65:4569] VERSION : 2 CALLED NUMBER : s LANGUAGE : en FORMAT : 64 CAPABILITY : 2147483647 ADSICPE : 0 DATE TIME : 147935484 Jun 17 10:07:57 WARNING[180236]: chan_iax2.c:4534 socket_read: Call rejected by 146.231.125.65: No compatible Codecs I can see the inconsitency with the FORMAT header of the two setup messages. According to the IAX protocol spec. The FORMAT (0x4) represents G.711 U-LAW, which is exactly what the resulting call uses. However, t...