similar to: function Echo() doesn't work

Hello, can anyboby tell me, how can I disable the echo cancellation for sip? thx a lot... best regards, Felix -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110216/4e54620f/attachment.htm>

Hello, could I adjust the Rx and Tx gains for SIP and CAPI? If it is possible, how should I do it? Thanks a lot. best regards, Felix -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110215/f36da2eb/attachment.htm>

Hi, I tested the capi_chan with latest cvs of * and I have problems with Early-B3. The following dialstring works for me (without Early B3): exten => _0X.,1,Dial(CAPI/@22715291:${EXTEN:1}|30) But if I add the 'b' for using Early-B3 exten => _0X.,1,Dial(CAPI/@22715291:b${EXTEN:1}|30) nothing changes (no dialtone). If in this example the called party discards the call, there is no

Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was

Hi, I just wonder, if it is possible, to suppress my own number on outbound dials with chan_capi. I took a look into the sources and think it might work with toggeling the "@" in front of the outbound msn in the dialstring. (Dial(CAPI/@msn... vs. Dial(CAPI/msn... But it doesn't work. Maybee I'm wrong and misunderstood the code. Thanks for any answers! Karsten

Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI> -- data = @89930:0107901723168212 -- capi

Hi everyone, we are using custom dfree commands to implement quotas. While these work fine on normal shares, the "dfree command" parameter seems to be ignored in the homes section. Is this correct (and intended)? Best regards Felix IT-Services Telefon 02461 61-9243 E-Mail: f.stolte at fz-juelich.de -------------------------------------------------------------------------------------

Hi, I installed Asterisk with CAPI support. Everything works fine while starting Asterisk, but when a call comes in Asterisk hangsup the call after two times of ringing. The output is like: Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg: CONNECT_IND ID=002 #0x011d LEN=0048 Controller/PLCI/NCCI = 0x101 CIPValue = 0x10

Hallo all, I am trying to get * with chan_capi and a ptp-ISDN with 4 lines on a AVM C4 card to work. But weather inbound nor outbound is working :( My capi.conf: ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] mode=immediate isdnmode=ptp msn=8993 incomingmsn=* mode=immediate controller=1,2,3,4 softdtmf=1 ;accountcode= context=demo

O.K. thanks a lot, Felix and Peer Oliver. But somehow asterisk keeps telling me while startup: [chan_capi.so] => (Common ISDN API for Asterisk) Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr1 Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr2 Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr3 Jan 24 14:30:47

Hello, while testing the following scenario, I ran into trouble: One * box with two AVM active controllers in Point-to-Point-Mode is connected to another * box with ZapHFC/Quad-BRI cards using bristuff in NT-mode. All is working fine, I can call from one box to the other and vice versa. But if I'll cut one line, it is not possible to place an outbound call from chan-capi accross the still

Hi, I am using Asterisk on a DSS1 ISDN-BRI with ISDN4Linux (and a Fritz Card PnP). The ISDN-BRI is in PTP-Mode (Point to Point "german: Anlagenanschluss") which is enabled within I4L with "hisaxctrl fcpcipnp0 7 1". Everything works fine except that I can not see the called number/MSN of incoming calls within Asterisk and because of this I can not route incoming calls

Hello list! I have a Asterisk 1.2.10 running using the package from Backports.org for Debian Sarge. I have setup chan_capi (0.6.5 from Backports) and it seems that I am only able to use on B-Channel. When trying to place the second call I get: CAPI INFO 0x34a2: No circuit / channel available capi info shows: Contr1: 2 B channels total, 1 B channels free And capi.conf is: [ISDN1]

Hi Felix, On 4/18/24 08:33, Stolte, Felix via samba wrote: > Is this a bug or maybe a misconfiguration? In the latter, which parameters could cause this? I guess the nodes end up using different combinations of dev/inode as primary key for the locking.tdb record. Probably because the device numbers differ on the nodes and you didn't fix this known issue with GPFS with the fileid VFS

Has anyone a working /etc/asterisk/capi.conf example for Germany or Switzerland using the AVM C4 - ISDN Card. I try to connect asterisk to 3 wires BRI-ISDN (Swisscom). I appreciate your help and it would save me a lot of time, figuring it out by myself. regards, Claudio

Hi there, i called one asterisk server from another asterisk server. The calling server played back a audio data und the answering server recorded the audio sample using record() function. I tried both ISDN, VoIP connections. Camparing with the original audio data, the recorded samples from both connections were amplified by asterisk, so that the recording were much louder. But I didn't

Hi! I'm trying to enable echo cancelation with my Eicon Diva Server 4 Bri card. I've enabled it from zapata.conf, (as I read from www.voip-info.org) ------------------------------- ; Enable echo cancellation echocancel=64 echocancelwhenbridged=yes echotraining=2000 ---------------------------- but nothing seems to change, echo is still present. Any idea? Should I enable/change something

Hello, I am using chan_capi 0.3.5 and Asterisk CVS-v1-0-12/29/04-15:32:48 on a SuSE Linux 9.0 with Kernel 2.4.21-99-default In the system is a AVM C4 with one port connected to PSTN at PTP BRI and another one to an ISDN PBX with an PMP BRI. The system is running fine, but I have regualary this error, and then chan_capi is not working anymore. Jan 18 15:29:46 WARNING[2919]: Avoided initial

Hello Armin, hello List I'm trying to get chan_capi working with asterisk from debian stable (asterisk 1.0.7, the debian version number is 1:1.0.7.dfsg.1-2). I managed to get it compiled by providing my own version of ast_copy_string. This is an Austrian PTP line. I can do outgoing calls fine (no comprehensive tests yet). For incoming calls, I'm getting "No answer" on the

Hi! I'm looking for someone who has successfuly setup an asterisk in austria with isdn in p2p mode and chan_capi. There is is a special problem in austria with DID. If someone is dialing the phone number of the asterisk pbx like 12345-0, zero is passed as an DID, but in Austria u can dial 12345, and the DID which is passed is empty. It seems that asterisk cant handle this. Any ideas?