similar to: fax pass-through

Hi, sorry for the long debug output below. I configured Asterisk with AMP to send the whole number including the extensions of the callers to the called party. Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but doesn't seem to work. 033811234451 is the call id i configured, and it seems to use them, but the caller will only see a 0338189040 instead of my

We are having an issue with transferred calls being dropped. Looking at the asterisk 1.2.10 logs, it appears that when it is dropped, the SIP unit send a CANCEL message to the server. On successful transfers this is not seen. The errors logged in the SIP Unit error log, I believe are from the second attempt to transfer the call, after it has actually been disconnected. Nothing is

hi all, i have an asterisk install with a digium 4 port fxo card and cisco 7960 sip phones -- running on a compaq Pentium III (Coppermine) at 800Mhz 256KB cache and 1GB of ram. when a call comes in on zap/1-1 for example, the delay between when zap sees the line going to ring state, and when the desktop telephone rings can be as long as 7000 milliseconds (or about 3 or 4 rings on an ear piece).

Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the

Hi there! Trying to originate and dial a number using Zap-8, used to work, but now it just fails. I enabled all debug I found in the source-code and this is the output from asterisk. Can someone understand something from the debug-output what is wrong and direct me to what the problem might be? The setup is correct, trust me, it worked some hours ago, haven't changed anything. Just dialing

About once a week or so my Asterisk box stops responding to all phones. I can pull up the console, do whatever I want at the CLI but the only way to get things working again is to restart Asterisk altogether. I finally cranked verbose & debugging way up (and watched my log files go from 1mb/day to 100mb/day), but below I believe contains my problem. The next line is 1.5 minutes later where I

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

Hello, my configuration is the following: a TDM400P board with an fxs and fxo daughter boards on it. I thus connect a fax to my FXS port, after having verified that this port was correctly functioning. For this, I had tried before with a simple phone, and with some basic voicemail exten scripts. Here is my simple dialplan for my fax reception: exten => 300,1,Ringing() exten =>

Hi All, I have a rather peculiar problem. Whenever I dial out over ZAP/g0 the phone will just ring and ring, even if I answer the phone on the other end. Whats strange is that the * phone will continue to ring even after I've answered and (sometimes) hung up the dialed phone. If I make an extension to just directly dial out on ZAP/1, its almost the same behavior, it will continue to

I have a very serious problem with calls between PAP2-NA and a TDM2400 (8 FXO). Almost every call dropped after between 20 and 30 seconds with conversation. I disable the sound card, serial and other things on my server, but the problem still continues. I've changed the RPT Packet Size to .20 on PAP2-NA, but nothing. Here a piece of my log: [Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

Hi I am using following say.conf file. Its a default file, which comes with Asterisk installation. When I call SAY DATETIME AGI function, it simply returns without playing date & time. Where as if I use mode=old setting, it works. Is this a bug or mode=new is not implemented for SAY DATETIME AGI function? [general] mode=new ; method for playing numbers and dates ;

Hi, Does any one experience that SIP phone to SIP phone (Polycom phone) calls can't hear each other, but Monitor application records both end's voices. It also happens in group pickup calls. Zap calls to queue (Local channel) also experience this problem (sometimes, our SIP phone can't hear any voice from incoming Zap calls when pickup, sometimes this happens after 10-50

I have the following in my extensions.conf [ext-local] exten => _53XX,1,Wait(2) exten => _53XX,2,NoOp,Dialing ${EXTEN} from ext-local-custom exten => _53XX,3,Macro(dialout-trunk,2,${EXTEN},,) This is used to match inbound caller-id for my legacy PBX. It works fine for inbound calls, but not for internal SIP calls. If I call from a SIP phone that is also in [ext-local], it looks like it

Hi, maybe a dumb question, but it seems that some calls are directed to our central dial in number despite the extensions the callers say they dialled. E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown extension, where it is right, and redirects the call to the central dial in extension 1234-0. This only seems to happen when the numbers are dialled manually. When

We have recently install an asterisk solution with about 60 physical extensions. While the system is running it runs reasonably well (Still have a few teething problems) but twice now they have experienced a degradation in voice quality and dropped calls and then finally asterisk completely crashes out. Restarting asterisk will work for a little while and it will crash again, each time less time

Hi Folks, I'm trying to use my mobile as a trunk via bluetooth - calls done in a softphone go thru GSM network and calls destinated to my mobile are answered at the softphone. I have asterisk configured to do so but I'm facing an issue - Audio is audible but it?s not intelligible. I feel like the audio is breaking. Below is the asterisk log. I also get lots of ?hci_scodata_packet: hci0

Hi all, Anyone know why this happens? Mem: 524288k total, 508120k used, 16168k free, 0k buffers Swap: 0k total, 0k used, 0k free, 0k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1 root 15 0 2152 664 576 S 0.0 0.1 0:49.26 init 7398 root 18 0 10172 2904 2312 S 0.0 0.6 0:00.21 sshd 9856

I have an app which handles a Mitel's command port to change the MWI lights. It detects dial tone, plays some DTMF digits, listens for dialtone-or-busy, does a manager event on what it finds, and returns. Since the Mitel command port does not give answer supervision (looks like it's ringing), and since I run this app via a AMI "originate" command, I set up an extension in

Hi guys, I'm trying to use Asterisk with LDAP integration. I created some schemas and it seems to work fine for sip.conf replacement. When I try to register a softphone to test the service, it seems ok from the softphone point of view (user registred) but when I do a "sip show peers", no one is registered (nor sip show subrscriptions, users...) I put my Asterisk on full debug and I