search for: loggers

Nik, This definitely helps! Please check your dial command. You've got "Dial(Zap/0/mynumber)" and I think you might possibly want it to be something like this: Dial(Zap/1/mynumber) or Dial(Zap/g0/mynumber) I don't recall there being a zap channel zero, but it is common to have a group zero. I would recommend trying Zap channel 1 - Dial(Zap/1/mynumber) - before trying the

I have a PRI E1 link between Asterisk 1.4.24 and Alcatel-Lucent OmniPCX Enterprise R9.0. As EuroISDN it works fine. However, I need to move to QSIG because of a firmware upgrade on the Alcatel PBX which doesn't support EuroISDN (please don't ask why). Besides, I've read somewhere that 2 B Channel Transfers "should" work with * 1.4, the latest 1.4 libpri and QSIG. So this

Hello, I have a strange h323 issue. After executing command "Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what

Hi Folks, I'm have Asterisk version 1.2.1 with a A101 PRI card. I'm working with the CLEC to bring up the PRI and inbound calls are hanging up at his end after a few seconds. I ran PRI debug but it only gives me minimal insight. " Ext: 1 Cause: Unknown (16), class = Normal Event (1)" He ran a trace and the only difference he is seeing is a "ISDN interface explicitly

I'm seeing both inbound and outgoing call failures on our ISDN-30 lines that only seem to go away when I do a "zap restart" or in extremis restart Asterisk (1.4.25 with a Digium TE205P and zaptel 1.4.12.1). If I don't restart zapata or Asterisk the problem rapidly get worse :-( The lines are from BT with LCR from Cable&Wireless (I've tried using the LCR bypass code and

Hi All, Hope you all are fine and good, Today i have found that Mine all PRI Channels are restating after every interval of one hour, and i have search and psot on fourms and everyone said that this is a normal behaviour. If this is a normal behaviour is there is any way to stop it { i still don't know what is the reson to restart ever hour } . Because this is listed everywhere that

Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Here is the portion of the log: Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)... Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing

Hi I have 3FXO trunks called ZAP-25,ZAP-26 and ZAP-28 and T1 Channnel bank I get this deadlock problem when 2 incoming call from FXO(Here ZAP-28 and then ZAP-26) wants to dial same channel (Here ZAP-1). In this senario ZAP-1 first answer ZAP-28 and thne ZAP-26 wants to call ZAP-1 but it time out and goto voicemail after that ZAP-1 try to reach ZAP-26 call by puting ZAP-28 on HOLD During

I am new to Asterisk and the protocol/language complex world of VoIp and PBX. But I have a dedicated machine running A@H 2.8, a single TDM400P with one FXS module card connected to a standard analog phone. The second card is an X100P connected to my analog PSTN phone line. I also have Grandsteam IP phone plugged into the network and a couple of x-lite SIP softphones. I can make outgoing calls on

I have been trying to sort this out for a while now but with no luck I have isdn <-> asterisk<-> pabx on a te205 incoming calls work fine outgoing calls seem to work fine but the call is dropped when answered I think it is to do with the line [May 8 17:51:55] WARNING[4762] channel.c: Unexpected control subclass '5' that is causing the problem but I don't know how to

Hi, We've implemented a 'page-all' function for some of our customers, and we've noticed that on occasion the page-all will cause asterisk to crash (safe_asterisk then restarts it again). The particular customer has about 20 phones, and also has 5 Linksys 932 to monitor the state of these extensions. I'm not sure whether it is the page-all that causes the crash, or the

Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a

Hi, I have a duplicated DTMF issue with, it appears, bridged IAX channels. I have the following setup: PRI IAX <-------->* PSTN <------->* Dialplan I've configured a number on the dialplan server to make and outbound call to the pstn. This call then comes back into the dialplan server to SayDigits(). I'm seeing that a few of my digits are being duplicated

Hi, all! when my h323 phone dial in Asterisk system, i can hear nothing. and the following is the log slice i picked from /var/log/asterisk/full. ps: i am using red hat AS5 kernel 2.6.18-53.el5,Asterisk-1.4.24.1, pwlib_v1_11_0, openh323_v1_19_0_1. Best Regards! 81948 [May 6 10:07:34] VERBOSE[11579] logger.c: -- Remote UNIX connection 81949 [May 6 10:07:51] VERBOSE[29627] logger.c:

Hi list! I recently upgraded to FreePBX 2.1.3 although I am not sure if this has something to do with it. I do a nightly restart of Asterisk, just in case. This has been working fine months but since a few days asterisk seems to die and I am not able to restart it again, I keep getting a socket in use message. This is on Asterisk 1.2.12.1, Zaptel 1.2.9.1 and Libpri 1.2.3 This is a snippet

I have recently moved our asterisk server from our LAN to a Debian Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our network. Our phones are behind a natted firewall. An ITSP provides a PSTN to SIP termination for incoming calls Public ITSP -->Asterisk server-->Natted firewall-->extension (192.168.1.x) Everything works fine (incoming/outgoing audio etc.) except

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

Hi, I set up a ring group. I would like for people who select a certain voice menu option to ring a list of extensions (I have just one extension in there at the moment) and if it doesn't answer to go to an extension's voice mail. I am using a version of asterisk from CVS, last updated a couple of weeks ago. This line in extensions_addtional.conf sends the call to ringgroup 3 if

Hey, I'm up and running fine with 30 Polycom 500s connected to Asterisk 1.2Beta on Cent OS 4.1 with a Digium TE110 connected to a PRI line. Nearly every hour, almost on the hour I get this: Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/1 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/2 successfully restarted on span 1 Sep 29

Hi. I''m trying to use logger from a rails cron job. It works fine, but I''d like the standard ruby behavior of stamping each log with date and time. How do I do this? I''m trying: # Set up out logging require ''logger'' logger = Logger.new(STDERR) logger.level = Logger::INFO RAILS_DEFAULT_LOGGER = logger # Tell rails to log to our logger also The