search for: verbosing

Hello, As you can see are pri is being reset every 60 minutes! Is there a way to stop this?? Is it a Zapata configuration problem? We have a * box with a single port T1/pri card installed. Thanks lach Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 2 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 3 successfully restarted on span 1 Apr 7 09:00:07

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

Hello everyone, I am facing a strange problem on my asterisk box (using isdn lines with pri card installed on it). Normal incoming/outgoing calls are working perfectly fine. When a user dial a wrong/out-of-service number they don't hear back any such message like "The number is wrong or user is switched off" in some cases, and it's just a silence for the user. Now while

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 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

Hello using Asterisk 1.8.32. I notice that there is a spontaneous reboot of the Asterisk system from time to time. When I look in the logs (verbose file) I noticed that every time this occurs it's at a moment that there is a MySQL action, be it a lookup or an insert/update/delete. I must say I do have some MySQL queries that occur in my dialplan when a call comes in, to look up

Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate.

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 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

We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 'resource unavailble'. Most of the time I see this error on the first or second channel on the second span in a trunk group (This is the providers trunk group for hunting, not an Asterisk trunk group). All

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

Hello, I have setup a test server with asterisk/AMP and have several 7960's connected to it. The asterisk server has a public ip and all the 7960's are behind nat'd routers. When I try to call from extension to extension I get directed straight to voicemail. I do not have any cards installed and instead direct everything to an Ondo server. I have been told it's not an AMP

Hello thank you for your answer. If I read your (and others) reaction correctly I can conclude that this is an Asterisk problem and not a problem of MySQL or dialplan logic ? You should know that the MySQL database is heavily questioned : mysql> show status like '%onn%'; +--------------------------+--------+ | Variable_name            | Value  |

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

Don't pass these flags to dozens of functions. --- builder/builder.ml | 47 +++++++++-------- builder/cache.ml | 4 +- builder/cache.mli | 2 +- builder/cmdline.ml | 13 ++--- builder/downloader.ml | 14 +++-- builder/downloader.mli

Using K-9 Mail to IDLE on a test folder "MyIncTmp". When there is a flag change on a message in the folder, which generates an untagged FETCH, K-9 Mail responds with a DONE, but dovecot does not exit IDLE state (LOG 1). However, if a new message is delivered, generating an untagged EXISTS, K-9 Mail responds the same way, with a DONE, and dovecot does exit IDLE state. (LOG 2) You

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