similar to: Faxing - machines stop talking, line stays up

Sorry, I didn't mention we're using spandsp txfax (0.0.2pre25) to send the faxes. -------------- next part -------------- An embedded message was scrubbed... From: Warrick Zedi <wzedi@noojeeit.com.au> Subject: [Asterisk-Users] Faxing - machines stop talking, line stays up Date: Tue, 23 May 2006 17:23:08 +1000 Size: 4568 Url:

I went to put in the new sound files over the weekend, but forgot to backup the custom folder and lost my custom digital receptionist files. I then had to copy the old files back from a duplicate machine. The problem is now though that voicemail just hangs up when I dial it. Other apps work - *70 for example gives me 'call waiting...activated' so I know it's accessing the files

I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting slips and ask me to update the clock source. I have my system.conf set as the following but when I run dahdi_scan only the ports on Card 1 are showing up with syncsrc=1 system.conf : span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,2,0,esf,b8zs bchan=26-48 mtp2=25 span=3,3,0,esf,b8zs bchan=49-72

Hi All, I am facing problem with E1 line. I have installed Asterisk (1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10) But every now and then I face problem of down E1's. The log show lot of entries like "pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2" This happens on a regular basis and the E1 becomes up after some time. My

Hi, Running 1.4.26.1 here. I have installed TE420B card in my server, and followed the appropriate steps (as far as I know to configure it). This TE420B is connected to a CLEC (T1s), so I am using pri_cpe as singalling type. When I dial out, I get this message: Dec 4 11:37:31] WARNING[27983]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0

Hi all, I'm having some problems getting TDMoE setup for the 1st time. I have a TE405P installed in the main server with an ethernet cross-connection to the secondary machine. (Yes, I know about IAX2 but I want to use TDMoE to simulate using T1s.) I'm using -HEAD from yesterday. On the main machine /etc/zaptel.conf: loadzone = us defaultzone=us

Guys, I'm trying to configure a TE406P with MFC/R2. here goes my zaptel.conf: span=1,0,0,ccs,hdb3,crc4 cas=1-15:1101 dchan=16 cas=17-31:1101 span=2,0,0,ccs,hdb3,crc4 cas=32-46:1101 dchan=47 cas=48-62:1101 The first strange behavior is that the: zap show status shows this: Description Alarms IRQ bpviol CRC4 T4XXP (PCI) Card 0 Span 1

We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are getting frequent restarts on the spans which lead to dropped calls. I have pasted some hopefully pertinent information below -- anyone have any clues that might help? Thanks Next line is repeated throughout messages, going through every channel in every connected span. asterisk/full.1:Apr 24 01:15:25 VERBOSE[4196]

The value of 14800 is correct. I had issues with my TDM400p with 2x FXO's installed and using the Xlite client. I could not get the echo stable at the initial call. Changing to a hard phone made everything work correctly. I still had problems with the off location I called, but mostly worked great. To see the TX gain I created an extension with the Milliwatt command attached to it and called

To start ztmonitor in quantitative mode you do the following. Assuming you are running Asterisk V1.0.9 you need to edit one line to ztmonitor.c in /usr/src/zaptel as per patch 2783 http://bugs.digium.com/bug_view_page.php?bug_id=0002783 Change line 261 fprintf(stderr, "Usage: ztmonitor <channel num> [-v] [-f FILE]\n"); to fprintf(stderr, "Usage: ztmonitor <channel

Hello, I have a busy problem with Asterisk when I try to transfer a call from PRI directly to IVR. This problem appear sometime after 2 hours or 2 minutes. The log file contain : Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) When this problem appear I must restart Asterisk to solve it. Another thing, I don't know why the alarm is set to NOP on SPAN

hello list i have installed 2 diguim card in my server using asterisk 1.4 (i use the old version with zapata.conf and zaptel.conf) i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i want to active the round-robin for span 2 and 6) in order to activate the WIMAX and FH please see the configuration below and tell me if there is anything wrong question 2: what is

getting some errors on my test system. this is 1.4 (Asterisk SVN-branch-1.4-r214194) with a 4 port T412p card. Three of the ports are connected: Span 1 to the PSTN on a 10 channel bearer line, ports 2 and 3 are cross-overed (!) to each other. Port 4 is not plugged in. This has been working fine for several months. I updated a few days ago to the latest 1.4 branch. However, now I cannot dial into

hello list, i have installed 2 diguim cards in my server using asterisk 1.4 (i use the old version with zapata.conf and zaptel.conf) i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i want to active the round-robin for span 2 and 6) in order to activate the WIMAX and FH please see the configuration below and tell me if there is anything wrong question 2: what is

Hi All, I have a production server that's running our corporate mail (Cyrus-IMAP). I used to use cpio to archive the mail store, but that's just getting silly now that there's about 40GB that it needs to run through. I tried running rsync in order to backup the data, but as soon as I started running the script the machine came to a grinding halt. I left it running for about 30

Good day list, I am feeling extra stupid this Monday morning and am hoping someone can come to the rescue. I am trying to use the ztmonitor utility on my wildfire 4 FXO card. and have read the following from the wiki. *********Wiki start******** If you set this to yes, use ztmonitor to adjust the rxgain and txgain. Ztmonitor isn't installed by default; but it is included with the Zaptel

Folks, I've been following recent discussions regarding echo and echo training with much interest, since it's a problem we've never been able to eliminate here. We're facing two challenges presently, and they may be related (or not): a) Cisco 7960s in the office here echo back to our staff, but the customer hears decent sound. b) We've yet to be able to pass faxes through

Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall-->E1 Digium port 1--->(Loop)E1 port 2---->sip extension local. kindly can any can help me to

Hi, I'm having problems with a TE420P card, in which I cannot make calls using spans 2 through 4. After a couple of days of working correctly, spans 2, 3 and 4 start failing (can not make calls). The system is configured to work with SS7. After the ACM message goes out, immediately a REL message is returned. I searched for error messages in /var/log/messages and could not find a clue

Hi, I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm using an OpenVox D410E and the drivers are loaded. My system.conf looks like this: # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED span=1,2,0,esf,b8zs bchan=1-24 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED span=2,1,0,esf,b8zs bchan=25-47 dchan=48 These