similar to: DAHDI channel congested busy

Hi All, I have a TE121P card installed and since connected it to the PRI I keep getting the Current Alarm as continually changing from Blue Alarm/Recovering and Recovering. The config I have is: /etc/dahdi/system.conf bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone = cn defaultzone = cn /etc/asterisk/chan_dahdi.cfg [channels] Context=telco

I've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending "reset all channels"

Hi, I running into the following problem on my Asterisk setup: --snip-- [Sep 3 01:40:59] NOTICE[9170] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 3 [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep

I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make calls from the Toshiba to Asterisk and internal calls from Asterisk to the Toshiba. What I can't do is make an call with an outside destination from Asterisk to the Toshiba. The Toshiba is looking for 9 to grab an outside line then it expects to see the 10 digits. In the FreePBX dial plan I use 9|. which sends 9 plus the 10

OK, I know it's only just out today but this is what I get when compiling dahdi-linux. make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware' make -C /lib/modules/2.6.31.5-0.1-default/build

Hi all, So, we reverted the LibPRI version and tested it, and then tried with the latest version of everything. Still no changes. The BRI line is in PTMP. If we set the configs to PTMP in the genconf_parameters and try it, we get the following: [Jan 21 17:32:20] ERROR[20341]: chan_dahdi.c:12645 dahdi_pri_error: Unable to receive TEI from network! If we set it to PTP (which it is not) we

Aloha, I'm not sure why I'm getting this error, but I can't seem to get chan_dahdi to load. SIP & IAX2 are working fine. Debian 4 w/ 2.6.28 kernel. Asterisk 1.6.1.5, dahdi-linux 2.2.0.2, dahdi-tools-2.2.0 CLI> module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Sep 1 10:57:51] WARNING[31696]: pbx.c:4550

Hey list, I'm getting the following warning when installing dahdi-linux-complete-2.2.0.2+2.2.0 : make[2]: Entering directory `/usr/src/kernels/2.6.18-128.1.6.el5-i686' Building modules, stage 2. MODPOST WARNING: could not find /usr/src/dahdi-linux-complete-2.2.0.2 +2.2.0/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.o.cmd for /usr/src/dahdi-linux-complete-2.2.0.2

hello, all of users: there are header files missed when you compile dahdi with kernel-2.6.29 or 2.6.33. i believe that few files are affected: wctdm.c dahdi-base.c wcb4xxp/base.c, opvxa1200.c... the errors look like these: ================================================ from?/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:61:

I trying to receive a SIP call and have ring a analog phone. I have a TDM11B card with FXS(green) module in line 1. I have Sip server "SER" setup to accept a SIP call, add a 970 extension to uri and set to asterisk SIP server on port 5065. When I send a SIP call from "kphone a soft SIP phone" running to sip://wally.world@cci.net "SER" picks call ok and changes uri

Hi all, yesterday afternoon, I called through my provider (teliax). but from the evening, I get this error. (below). then I checked in My Account page ans support page in teliax. and I saw that they have given new setting (to another proxy sever). I followed new settings. my Asterisk server is connecting to the teliax. but still I con not make called. it shows this error. If somebody had this

hi all i have a TE110P connected to my PBX when i try to call a extension number in other location 3525 the asterisk give me a error -- User entered '3525' -- Executing [450 at lacnicuy:4] GotoIf("Zap/31-1", "0?6:5") in new stack -- Goto (lacnicuy,450,5) -- Executing [450 at lacnicuy:5] Dial("Zap/31-1", "IAX2/lacnic:splacnic at

On Sat, Mar 14, 2020 at 2:02 PM John Roman <john at dev1ce.com> wrote: > greetings asterisk users :) > ive just deployed version 17 and migrated as best I can to pjsip. I can > receive calls, and get to my mailbox prompt, however placing calls seems > impossible with the following error on dial: > > Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel

On Wed, Mar 18, 2020 at 2:37 PM John Roman <john at dev1ce.com> wrote: > ive enabled logging. aside from a realm error i see on my endpoint, im > still not sure whats up Did you selectively enable logging? I don't see any SIP request for the trunk. If you did enable it for everything, then I'd suggest checking "pjsip show endpoints" and seeing the status of the

Hi, I'm having a problem with the queue behaviour in my place: I have two ISDN channels to the outside (Zap/1) and two channels two a Siemens Gigaset (Zap/4). I also use a SIP gateway to call outside and have a couple of IP phones around as well (SIP). The Gigaset has about 5 phones connected to it (+base station). Whenever two people are using those, I always am blocking two internal

Hi, we are still trying to properly connect a Tenovis PBX to an Asterisk server (asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this time with QSIG. Calling from a Tenovis phone to a SIP phone (i.e. traditional phone -> Tenovis PBX -> QSIG -> Asterisk -> SIP phone) works with the following messages: --- Don't know what to do if second ROSE component is of

Hello, The company I work for recently purchased 2 Rhino CB24s and a Rhino PCI-E R4T1. The channel banks are plugged into the R4T1, as well as 2 PRIs from our telco. The CB24s are for all internal analog phones. Most of the phones are setup in "batphone mode", which is "immediate=on" in the DAHDI config. They are set up this way because we are an outgoing call

Hello guys, i have a question about a voip gateway we use. I saw those values typing in cli: core show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - - - - - - - - - - - - gsm - - 2001 2001 6000 2001 2000 16000 - 34002 - 6000

Dear all, i have a problem with a system running - Ubuntu 10.04 ( all updates done ) - ii asterisk 1:1.8.5.0-1digium1~lucid Open Source Private Branch Exchange (PBX) - ii asterisk-dahdi 1:1.8.5.0-1digium1~lucid DAHDI devices support for the Asterisk PBX I also use freepbx 2.9 for the configuration. Hardware is a Dell R410 and a Digium

Hello! I use asterisk with TE420 as PRI switch for two channels : ;panasonic uplink group=3 context=panasuplink ; relaxdtmf=yes ; immediate=yes rxgain=0.0 txgain=0.0 mohsuggest=default jbenable = no ; jbenable = yes ; jbmaxsize = 200 ; display_send=name_initial display_send=name