similar to: PRI answer too fast

Is there anyway a T1/PRI can ignore the ANSWERED signal and just go straight from a dial command to the call was answered? I have a PBX that when calling a certain analog trunk it is not giving me signaling that the call was answered however I hear the PA system come off hook and give dial tone. So is there something in the dial command that can override the signaling and just go straight to

What is needed to get DAHDI to start up correctly on CentOS 7 and systemd... I am using DAHDI-linux-complete 2.11.1 I saw mention in my search that it has been fixed after 2.11.1 but cannot find what the fix is. Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:54] NOTICE[11305]

I used my cell to call in and create a CDR record here from asterisk 1.4.43: "","317XXXXXXX","s","default","""GEIS JERRY "" <317XXXXXXX>","DAHDI/23-1","","BackGround","SM_ATTENDANT","2012-01-05 18:12:09","2012-01-05 18:12:10","2012-01-05

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000

I am connecting to a siemens hipath 4000 with dahdi 2.1.0.4 and asterisk 1.4.23 using a Te210P card. the phone guy is saying that the lines are reporting always BUSY. however on my end the status shows OK. Anyone seen this? Is there something different about connecting PRI to siemens hipath? system.conf shows: loadzone=us defaultzone=us span=1,1,6,esf,b8zs bchan=1-5 dchan=24

I am getting this: make gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pritest.o -MF .pritest.o.d -MP -c -o pritest.o pritest.c pritest.c:50:28: fatal error: dahdi/tonezone.h: No such file or directory #include <dahdi/tonezone.h> I am using dahdi-linux-complete-2.11.0+2.11.0 I did "find . | grep timezone.h" from the dahdi root and its not

Hi all, We we send a server out to a customer with a PRI card I basically tell the PBX person that we want national, NI2, b8zs, D channel is on 24, and we are the CPE and your the NET and us a cross over cable. Allow extensions, local calls and long distance. on the surface I would have thought this would be enough for them to connect the digium card to the PBX and get calls working. The PBX

Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settings pri show spans Displays PRI

In 1.4.43 I would see things from "core show channels" like DAHDI/18/xxxxx for line 18 in Asterisk 11 its DAHDI/i4/xxxx How do I get the line number back? Jerry

I was running : asterisk 1.2.24 zaptel 1.2.21 libpri 1.2.6 I remove zaptel and compiled asterisk 1.4.22 libpri 1.4.7 dahdi 2.1.0 dahdi_cfg -vvv DAHDI Tools Version - 2.1.0 DAHDI Version: 2.1.0 Echo Canceller(s): Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel

Anyone using 5ess with success? I have a customer that wants to use it. Seems to WORK for a short time then there are errors on the D-channel and their side (nortel switch) shuts down the channel. They were saying something about a slip error. I dont know this term. I am using dahdi 2.1.0.4 and asterisk 1.4.22 and libpri 1.4.7 Any ideas? jerry

Have a new TE133 in a dell R320. dahdi_cfg -vvv shows: DAHDI Tools Version - 2.9.1 DAHDI Version: 2.9.1 Echo Canceller(s): Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear

On Mon, Nov 17, 2008 at 10:20 AM, Tony Mountifield <tony at softins.clara.co.uk > wrote: > > If I do this from an NEC digital extension I get 141496920000, but if I > do > > it from an NEC POTS extension I get 1942124000 > > That looks like when you pick up the analogue phone and dial 9, it > immediately opens the outgoing line and sends the 141 acces code, but >

>>I'm not sure that your question is clear. You'll probably want to be more specific. >> What is pulse? You mention "as a user", are you talking about voicepulse.com ? >> What are you trying to do with pulse? >> What problem are you running into? Sorry Rusty... I am trying to get Asterisk 11 to co-exist with a centos 7 box that has pulse audio running as

I have two servers, each connected to the PTSN via PRI. When I call from site A (951-999-9999) to site B (555-1212) and the phone at site B is on the phone, I hear the normal ring tone for about 20 seconds, then the message "all circuits are busy now. please try your call again latter" followed by the congestion tone. Instead, I want this to busy ring and then hang up without any

Hello all, I've got an intermittent issue with my asterisk set-up, and I'm pulling my hair out! Mostly everything works fine, but I get an error once every few days that sometimes (but not always) causes asterisk to stop accepting new calls and also stop responding to commands on the console (asterisk -r). The usual errors look something like this: [Feb 19 13:09:52] ERROR[18728]

Hi, I tried to use Early Media: exten => 1,1,Playback(demo-thanks,noanswer) same => n,Hangup() But when calling my extension I do not hear the voicefile - I only hear the ring tone. In the Asterisk-Log I can see, that the voicefile is played. I got the same result when using "Progress()" in the first priority. I tried "pri set debug on span 1" and got the

dahdi_cfg -vvv error `cut: '/sys/bus/dahdi_devices/devices/pci:0000:01:00.0/spantype': No such file or directory` PRI show spans --command not found test call not working. -- dahdi_genconf -vvv Default parameters from /etc/dahdi/genconf_parameters Generating /etc/dahdi/assigned-spans.conf cut: '/sys/bus/dahdi_devices/devices/pci:0000:01:00.0/spantype': No such file or

Hi List, I have configured TE121PF card in E1 mode. I am using asterisk 1.6 and freepbx 2.7. My card staus turn in to green and looks like sync with the service provider. My service provider is BSNL - India. I have one toll free number for incoming and one land line number for out going calls. Problem : If i am calling to the toll free number, i am getting the ring but that call is