similar to: No Echo Cancellation with echocancel=yes

Hi, I've set up an Asterisk as voip gatway: VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx. Outgoing calls from dect handset to the world are working. Incoming calls don't even ring the handset. I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in nt mode. The msn is set at the dect phone/base station

Please forgive me, if I misunderstand the problem completely. Following instructions in several german blogs, I want to configure Asterisk with a hfc-pci card, an old NTBA and an ISDN phone as a SIP device. It seems that I have to set signalling in zapata.conf to bri_net_ptmp. When I do this, Asterisk will hang if started with -vvvvc, the last lines of output being: [res_features.so] =>

Hi ""noobs"" and others(like me). I've been struggling for hours,days,days,days.... to get my Asterisk work with 2x HFC-S BRI cards. One card for use in NT-mode for connecting my ISDN DECT phone,and one card as gateway to PSTN. Now it'works,and I just want to share some of the experience with others. I'd guess there must be 'Somebody Out There' having

Hi I have a small ISDN PABX , (BRI) at home - Siemens Gigaset 4175 which has cordless DECT extensions. I have set up asterisk on FC3 with two HFC cards and I am using the latest bristuff. I am trying to use * between my ISDN line and my PABX. One card is in TE mode and can receive and make calls OK. I can make and receive external calls to a SIP phone. I cannot get the other card to talk to my

I am running * with a Junghanns quadbri that should allow us to integrate our ISDN house telephone system with VOIP. Preferably I would like to run a setup, so that our internal ISDN phones on an S bus are not aware that * is sitting in between. With the configuration below I run into the following problems: 1. On outbound calls, I get the normal rining call progress tone althought the

I've got a bit trouble with callerid and zaphfc cards. Basically zaphfc doesn't add the "0" in front of national numbers (haven't tried a international call yet). With chan_capi that allways worked fine, however i had to define the national and international prefixes in capi.conf. Is there something similar in zapata.conf ? Here is my zapata.conf: [channels]

Retrying, falling of the list some how :-( -------- Original Message -------- Subject: feeling n00b again Date: 2018-08-20 09:51 From: asterisk at a-domani.nl To: asterisk-users at lists.digium.com Hi all, Long time ago, I followed a Asterisk training, and both at work and at home, was able to deploy Asterisk, make all sorts of internal call (hard/soft voip-phones, incoming/outgoing,

Hello everyone, I try to connect the Asterisk PBX Server on my NTBA (NT1) - actually its "hard"-connected on the box, but I've tried a crosslink-ISDN cable, too. My ISDN-Phone is connected on one port (of two) S0-Ports on NTBA. When I start Asterisk and want to make a call I'll geht a lot of this messages --> 040827-034639 WARNING[213005]: chan_zap.c:6902 zt_pri_error:

Hello list We're trying to get a HFC card running in NT mode. Zaphfc loads fine as module and ztcfg -v is run only once per reboot. Cabling is ok, the ISDN-TE gets ist power fine. The HFC card does not share any interupts. After the TE picks up the console (pri debug enabled) shows: Nov 21 18:42:20 NOTICE[13326]: chan_zap.c:7169 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1

Hi, I solved the Problem: the context defined in zaptel.conf: context = zapincoming was not in extensions.conf, so nothing worked. ( I assumend this context is only necessary for calls from zaptel, not to it) This may be a bug: in the verbose log I found nothing like "[chan_zap] Could not initialize ..., due to ....", this had helped alot and had saved alot of hours (-;

Hi all, I just posted this question before last week. Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to 1.0.8-BRIstuffed-0.2.0-RCh the same problem occurs, but seems to be more seldom. Attached is now the output of "zap show channel" . - I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.

Hey, We have asterisk 1.2.4 (old I know) with a couple of snom phones, a couple of grandstream phones and around 65 philips dect stations. Now the problem: All calls do peer to peer RTP except the calls from dect station to dect station. snom to dect or dect to snom do peer to peer. So the sip config looks fine because all the 'static deskphones' honor the REINVITE and start talking to

G'day. I hope this isn't off-topic for the list. I am looking at an Asterisk setup that includes cordless phones. The three choices I can see, at this stage, are: * wifi phones * an ATA and a cordless analog phone * a DECT to SIP basestation The various wifi phone options don't grab us as suitable -- they are costly, have poor battery life and even the best have pretty mixed

Hi!I've this problem to another asterisk@home machine, without digium cards, but only with a bri isdn card.It doesn't connect in the amportal graphical,(it's stopped), if I make tha amportal start command this is the result: STARTING ASTERISK Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 Asterisk died

Hello, I have been experiencing audio problems when accessing the Voicemail application using DECT phones and the g729 codec. The issue is that whereas the vm-password is always played correctly by the DECT phone, the rest of audio files, randomly, are played or not by the DECT phone. Everything works correctly if another codec (alaw,ulaw) is used. I have noticed that asterisk doesn't send

I've got some weird behavior on my HFC-s cards. asterisk CVS-06/26/04-21:28:35, bristuff 0.02, libpri 20040510, zaptel 20040623 When i pick up my ISDN phone on Zap5-1 ("3987") and call the external number "1901" it will do so, connect me and everything is fine. In the second, where it tries to attempt the native bridge, the audio will disappear. Using another card

I have installed two HFC PCI A-cards running zaphfc from bristuff-0.0.2, which seems to work quite fine, but I continously receive the messages "D-Channel on span 1 up" followed by "D-Channel on span 1 down" with a few seconds interval. Why is that? Bri intense debug log and configuration files below. I don't need ztdummy or zaprtc, do I? I get some frame slips as well,

Anyone here that knows some hardware USB "client" (DECT) that can connect to a DECT base station and act as a "provider" in asterisk? Ergo, asterisk is the DECT handset, and connects to the DECT base station. Its also important that it also works with most asterisk-compatible devices, including for example raspberry's. The reason I ask, is that many providers in

Hi there, I just upgraded a relatively old Asterisk installation (1.2) in our office to a relatively new version (1.6svn from last wednesday) which runs a Junghans QuadBRI card [1]. To get this flying I got dahdi-linux, dahdi-tools and libpri from SVN as well. After a while of juggling it "works". What doesn't work: connected ISDN devices (Gigaset phones connected to QuadBRI

Hi, I'm pulling my hair down and getting bold :-) ..... I have Asterisk between Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff Asterisk).... I'm trying to do just plain transfer of call from pbx to ISDN through Asterisk... It seems like PBX hangsup, when call is progressing with no apparent reason. I'd kindly ask for any advice or some working example for