similar to: RE: 8 channel fxo setup outgoing call problem (cont)

I have an eight channel fxo setup (2 TDM400P cards) and I have them setup. Here are my configs: Zaptel.conf: fxsks=1-8 loadzone=us defaultzone=us Zapata.conf: [trunkgroups] [channels] musiconhold=default rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callerid=xxxxxx callwaiting=yes busydetect=no callprogress=no usecallingpres=yes callwaitingcallerid=yes

I'm not sure what you mean, I had one card with one FXO in it and it worked fine. Now, we have 2 TDM400P cards with all of their slots filled with FOX chips and I can't make out going calls. Those are the only 2 cards on the system, the rest is just motherboard. -----Original Message----- Do you have ATI FXO daughter cards. We had experienced similar problems. After replacing the ATI

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

hi i've configured a TE205P on asterisk at home this is my zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone = it defaultzone = it and my zapata.conf signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes

Hellow everyone, here's an other newby question. I've got a * configured with the card in the subject line. At times Asterisk fails to notice a disconet from the incoming line going into one of the FXO ports. Consequently it just keeps the line off-hook for ever and that causes my provider to mark the line aut of order. Is there any way to "help" Asterisk notice the disconect?

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

Hi everbody this is my first post, I have two test server .(Both of them is idle) db1 : RHEL4 OCFS2 db2 : RHEL3 OCFS I test the IO both of them The result is below. db1(Time Spend)db2(Time Spend)OS Test Command dd (1GB) (Yazma)0m0.796s0m18.420stime dd if=/dev/zero of=./sill.t bs=1M count=1000 dd (1GB) (Okuma)0m0.241s8m16.406stime dd of=/dev/zero if=./sill.t bs=1M count=1000 cp

Hi everbody this is my first post, I have two test server .(Both of them is idle) db1 : RHEL4 OCFS2 db2 : RHEL3 OCFS I test the IO both of them The result is below. db1(Time Spend)db2(Time Spend)OS Test Command dd (1GB) (Yazma)0m0.796s0m18.420stime dd if=/dev/zero of=./sill.t bs=1M count=1000 dd (1GB) (Okuma)0m0.241s8m16.406stime dd of=/dev/zero if=./sill.t bs=1M count=1000 cp

Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the channel, i cannot hear nothing. If I change the configuration and i send the call to an internal sip

Nik, Just curious - what is your telco setup? Do you have PRI with the specified D channels? You need to make sure that your telco is set up to have the D channels on 16 and 47. When you first start Asterisk, or when you log on to the CLI, do you ever see messages stating the B channels are successfully started? Let us know. -MC -----Original Message----- From:

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

I have Asterisk working with one FXO card (clone x100P card PCI card). I am trying to add 2 more cards so my question is - do I just increase the channel count on the zaptel.conf and zapata.conf files? [original] /etc/zaptel.conf fxsks=1 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] language=en context=from-pstn signalling=fxs_ks faxdetect=incoming usecallerid=yes

Hi All, Channels 25-28 on a customers PBX are regular Zaptel FXO cards that are hooked into 4 incomming phone lines. They are all in a group to do automatic rollover for outgoing calls (if channel 25 is being used, dial on channel 26, etc.). Sometimes when a user is dialing a number, instead of bridging to one of the FXO cards it goes and rings to Zap/1-1. This doesnt occur all the time but some

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

I am experiencing strange results using dmvnorm. I define a scaled distance matrix from the coordinates bellow and then calculate a covariance matrix using a spherical correlation function. Then with certain combinations of range and sill parameters dmvnorm is returning values greater than 1. Surely the results of dmvnorm should be in the interval 0:1 (or do I just nead a holiday?). In addition

Hi All, This must have been asked before, but despite my best googling efforts, I can't find a solution or reason why it won't work. Please accept my Apologies. Is it possible for an outgoing call on the FXO port (PSTN line) to not always register in the call logs as being answered? I've tried adding the 'r' option to the end of the Dial command, but that does not appear to

Hello, I have an issue with Digium TDM 400 card series. When I try to make outgoing call (PSTN call) for example, the Zap channel could not be created and busy channel message appeared. Below is the full log : [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [s at macro- dialout-trunk:20] Dial("SIP/213-09648720", "ZAP/g1/08170709XXX|300|") in new stack [Feb

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure