similar to: DEBUG Message

Hi all, I have an Astribank connected to Asterisk 1.4. I'm setting up extensions and I have a problem with inbound calls to zap extensions. The phone at 65 rings once and then the line gets hung up. If I pick up the phone really fast, it works. Any suggestions? I have the following setup: [from-pstn] exten => 207582401,1,Dial(Zap/65,30) CLI shows me this: -- Accepting call from

hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten =>

Hi all, I have a connect between a siemens hipath & Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting "The number you have dialed is not in service" In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk)

I noticed a problem this morning with our cdr logging. We have a cron job that places a call file into the spool directory having asterisk call itself to check to make sure its still handling incoming calls correctly, then queries the CDR database in mysql and makes sure that appropriate records exist. I can confirm that the call is happening correctly, but I'm missing records in the

Hi All I am using asterisk version cvs-v1-0-04/15/05 and spandsp 0.0.2pre18. I can receive and then email most faxes without issues, but recently I am having this issue when receiving faxes from a particular person. I can receive the faxes ok, but there are alot of bad rows as indicated by my logs below and the fax is not readable. I have included a good (from another user) and a bad fax. We

I have a user who's system crashes on pri hangup request. Tried 1.4.19.1 and 1.4.20 as well as the latest libpri no change Progress is as follows...... < Supervisory frame: < SAPI: 00 C/R: 0 EA: 0 < TEI: 000 EA: 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 025 P/F: 1 < 0 bytes of data -- ACKing all packets from 24 to (but not including) 25 -- Since

Hi All I am using asterisk version cvs-v1-0-04/15/05 and spandsp 0.0.2pre18. I can receive and then email most faxes without issues, but recently I am having this issue when receiving faxes from a particular person. I can receive the faxes ok, but there are alot of bad rows as indicated by my logs below and the fax is not readable. I have included a good (from another user) and a bad fax. We

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, We are running Asterisk 1.0.0 with a TE410P. Very often we exerience calls dropping in the middle of the call. I enable the full logging and saw a couple of suspicious messages right before the hangup. Thos could happen on a Zap-IAX2 bridge as well as on a Zap-Agent bridge... I see Nov 23 09:08:36 DEBUG[-1274020944]: Bridge stops because we're zombie or need a soft hangup:

Hello, I'm using asterisk-1.0.8 with BRI and spandsp-0.0.2_pre20. Modules app_txfax.so and app_rxfax.so are compiled and loaded sucessfully. It seems that the channel cant't detect the call as a fax-call 7612022801 is the calling faxmaschine 1209259 is my recieving fax extension logs are:

7. Take an Asterisk training course and become a dCAP. As for "we have try to solve it but we're lack of asterisk knowledge" - would you get a plumber to service your car? Why not employ (as in 'pay money') somebody to investigate this further. As Satish pointed out - QoS type issues take a lot of debugging and that usually has to be done on-site. BTW - I doubt any of

Hi! We have some problems with calls that get disconnected in the middle of a call. We are using Asterisk 1.2.1 with a TE410P (2.gen firmware). When the call is disconnected Asterisk writes this to the log: Jan 9 14:56:17 DEBUG[4404] dsp.c: ast_dsp_busydetect detected busy, avgtone: 300, avgsilence 2090 Jan 9 14:56:17 DEBUG[4404] dsp.c: Requesting Hangup because the busy tone was detected on

I'm running * with a bog standard HFC ISDN card using zaphfc. Everything seems to work, including incoming calls, but I simply cannot make outgoing calls. This is very odd since the same card worked with the same configuration in another server. This is what I get from * debug. The only possible difference between the two servers that I can think of is that the HFC card is sharing an IRQ

One of our client facing this issue, we have try to solve it but we're lack of asterisk knowledge. Anybody can help us? Isn't any problem with asterisk configuration or the problem come from PRI E1 itself? [Apr 11 15:32:48] VERBOSE[9231] chan_dahdi.c: -- Requested transfer capability: 0x00 - SPEECH [Apr 11 15:32:48] DEBUG[6888] channel.c: Avoiding initial deadlock for channel

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Making some changes in extensions.conf to test the incoming calls so that these are derived to a SIP extension, I found something that draws attention to me: if I test calling to my PSTN line from a mobile phone, when take the call from the SIP extension (softphone), if the mobile phone releases the call, sofphone do it too without problems,

Hi, all. I've got app_rxfax going and nicely receiving a fax, which I then throw to a script, and have it convert it to a PDF and mail it. Works great... a lot of the time. But a fair bit of the time, rxfax throws errors, and extensions.conf seems never to invoke my script. Here are the pertinent lines: exten => _6403,n,rxfax(${FAXFILE}) exten =>

Hi Everybody, I am trying to use SIP (Sipura 2000) to connect to Asterisk which then dials out a local number using the Digium E100P. We have purchased the G729 codec licenses from Digium and loaded them into Asterisk successfully. However, the call drops immediately after being answered with the debug error message saying something like: "channel.c:2646 ast_channel_bridge: Didn't get a

Hello, I have an Asterisk box with a TE410P connected to a PRI line and agents with X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I make outbound calls it hangs up as soon as other party tries to picks up the call. Does someone ever experienced this situation? On X-Lite, only G711-ulaw is enabled and here is what i put in sip.conf: [4001] type=friend username=4001

Hi I am experiencing Asterisk Crash. Log got stopped when asterisk crashed. Please help me to identify the reason and fix this issue. Asterisk: 1.8.20 I am using AMI and fastAGI to control the call. Some part of dial plan is also defined in extensions.conf I am experiencing this crash when app_meetme conference functionality is used with more than 3 parties. I faced this issue with

Hi, I facing a problem with flash hook. When ever I do a flash hook to place an extsing call on hold, the call gets disconnected. The debugs on Asterisk shows that 'on hook event detected' when I press the flash button on the phone. The setup is like this Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD and configured for ISDN PRI lines. Analog phones come