similar to: SpanDSP Noise every 300 ms

Hey All, I'm still (since April) having problems getting RxFax to work over an ISDN4Linux channel. Just wondering if anyone has had any luck getting it to work? I have done a CVS update today (about half hour ago) and made sure I have the latest version of spandsp according to Steve's website (spandsp-0.0.1k). When I was compiling asterisk, I got the following warnings:

Hi, I am trying to recice a fax with * using SpanDSP - but it doesn't create the output file. (See the bottom of log file). * Loads both app_rxfax.so and app_txfax.so fine. Also I can't make * autodetect an incomming fax call (yes I have enabled faxdetect=both in zapata.conf - though it's not a Zap device) Any ideas are welcome :-) Best Regards Michael L?jtnant System Details:

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

Some unsuccessfull attempts to make console calls working. If a sip phone is called, the other side will hear nothing. If I try to record some sound the application will not finish. There is a sound file, but it is empty (0 bytes). "Record(${FILE}:gsm|10|30|skip)" is used in the dial plan. After hangup the following error messages show up: NOTICE[]: channel.c:1683 ast_set_read_format:

Hi all, I am struggling to make my asterisk server work. The problem is I can not place a call from a phone outside, but I can call out from a phone in the local network where the asterisk server sits. I turn the debug on, and the log are shown below. I can see "REGISTER" method is OK. ( SIP/2.0 200 OK) But Later, in the "INVITE" method, the SIP addresses become

Well, it has been a long time since I used my NetJet S cards with asterisk (moved to the quad PRI card) but I am trying to get this working again for home use. Basically, what I am using is this: Linux 2.6.8.1: config lines: # ISDN subsystem CONFIG_ISDN=y # Old ISDN4Linux CONFIG_ISDN_I4L=y # CONFIG_ISDN_PPP is not set CONFIG_ISDN_AUDIO=y # CONFIG_ISDN_TTY_FAX is not set CONFIG_ISDN_DRV_HISAX=y

Hi Manfred, I applied the patch and recompiled and reinstalled and I got the folowing warning during my first test call: Aug 19 12:26:51 WARNING[294927]: dsp.c:1234 __ast_dsp_silence: zero length packet It looks like that could be the problem... and the fix! I'll let you know if the problem reoccurs. Might it be an idea to submit the patch to the bugtracker? Thanks, Gary ----- Original

Hi all, this is a rather "newbie-oriented" question, so please bear with me... The system running Asterisk has been provided with an AVM FRITZ!Card PnP. SuSE Linux 9.0 recognizes it right after booting the system and it seems to be configured (MSN) correctly... The hwinfo looks like this: --- pbx:/etc/asterisk # hwinfo --isapnp 11: ISA(PnP) 01.0: 10300 ISDN Adapter [Created at

Hello! Im pretty new of asterisk world: my goal would be using iaxcomm to call ppl over POTN. Yesterday i configured asterisk to be able to hear a song (with Playback() command) with iaxcomm, and it was wonderful. Now, today i tried to configure asterisk to use my ISDN4Linux supported card (a plain HFC card) with asterisk. i configured modem.conf this way, after reading

Hello, I just got my isdn-card working together with i4l and asterisk. Everything seems to be working fine: I can accept calls coming from the outside and I can dial out. Even setting the msn works like charm but my problem is that I cannot hear a word. There's complete silence in both directions. Any idea what could be the cause? Thanks for your help, Gunther Lspci: 0000:01:07.0 Network

Hello everybody, I have a strange comportment with oh323 and asterisk, I'start testing asterisk but with this I can't understant plesae help me ! Thanks Eltorio ---------------------------------------------------------- 1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a Modem[i4l] line ---------------------------------------------------------- Nothing happens

Hello all, I currently have an Eicon Diva Client isdn card using i4l. Outbound dtmf doesn't work (and never has), but there has been an annoying problem with false dtmf detection in calls (that could be triggered easily by blowing into the receiver on the remote end). I looked through the list and found two patches that need to be applied - 1 to isdn_tty.c in the kernel, and another to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I'm having some problems using the asterisk modem driver with an ISDN4Linux card (AVM Fritz). (I realise that it's better to use CAPI, but unfortunately this card doesn't have any CAPI drivers). Every so often the ISDN just stops working (it neither dials out nor accepts incoming calls). Trying to dial out just logs: --

I tested SpanDSP as an internal extension, and it worked like a charm. Now I am trying to receive faxes from a toll-free nufone DID. I am running g.711uLaw in on this line, so no to cause too many problems. However I receive the following errors after the fax is finished receiving: so the fax comes in Executing RxFAX("IAX2[NuFone@198.22.67.70:4569]/5",

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Hi, will spandsp work with i4l, or should I use capi? tia mazek -- http://www.marcinmazurek.com/ ::: nic-hdl: MM3380-RIPE GnuPG 6687 E661 98B0 AEE6 DA8B 7F48 AEE4 776F 5688 DC89

Hi to everyone! I am new to this list and i want to greet you all! I am trying to install Asterisk on a Fedora Core 3 Distribution with a Digi Datafire Micro V ISDN Card. I know that I have to use the ISDN4Linux so in modem.conf I set the I4l driver ,the msn and device. Modem.conf [interfaces] context=remote ;driver=aopen ; modem by AOpen driver=i4l ; isdn4linux - an

Greetings, today I'm recompiling asterisk on a new machine with the same isdn card (not the same model, the *SAME* card) using CVS downloaded yesterday night. When I try to start *, it gives me the error: [chan_modem.so] => (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated Modem

Hi all, i have noticed that i can't hear a ring tone if i make a call from my TDM40B to a chan_modem_i4l endpoint. I had a look in the code from chan_modem_i4l and there is a function calling "i4l_handle_escape" that gives a AST_CONTROL_RINGING frame back. But this seems not work ...(or i4l is not signaling it ?) Til now i have used the Dail app like Dial, Zap/g1:XXXXXX|60|r so it

more detailed output Sep 21 15:54:31 DEBUG[1120357296]: pbx.c:1255 pbx_extension_helper: Launching 'RxFAX' Sep 21 15:54:31 DEBUG[1120357296]: channel.c:1699 ast_set_read_format: Set channel Zap/1-1 to read format SLINR Sep 21 15:54:31 DEBUG[1120357296]: channel.c:1666 ast_set_write_format: Set channel Zap/1-1 to write format SLINR Changed from phase 0 to 1 Slow carrier up Slow carrier