similar to: TDM40B hangup on fax or data modem carrier

Hello, I have trouble to enable a sip trunk with a CIRPAK. CIRPAK support answer that's there parameter are unvalid : a=silenceSupp:off - - - - is not standard and not working with cirpak - to be remove m=video 13072 RTP/AVP no video, how to remove it ? my extension.conf : exten => _6X.,1,Dial,SIP/${EXTEN:1}@x.x.x.x Regards, -- Arnaud Pignard (apignard@frontier.fr) Frontier Online -

Hello, We have one of our SIP provider that's is sending incoming sip call without need of registration. Incoming call working fine (as outgoing call), but * still try to register to there sip gateway : chan_sip.c:3159 sip_reg_timeout: Registration for 'phone@50.50.50.50' timed out, trying again -- Got SIP response 404 "User Not Found in data base" back from 50.50.50.50

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --

Hi, Since around a week i have one asterisk server how stop responding randomely. CVS HEAD with RealTime engine used. The debug log only write "Failed to grab lock, trying again..." until i stop Asterisk. No more activity for IAX or SIP channels (no log...). CLI still responding. When i try to stop asterisk (stop now or crtl+C) nothing happen and cli die but asterisk still running.

Hi, I have a partial working installation with zaphfc. Incoming call : For incoming call, seems work fine. But the sound is very bad with bounce short crashing sound. Same sound with echo cancel off or on. SDA work fine. Another problem, it's seems that's zaphfc don't reset correctly the line. I have one of my D channel how was busy even after stop communication. Outgoing call :

Hi everyone, Sorry for forwarding and top-posting this email again but its as if my TDM40b has keeled over yesterday. After a few hours last night and swapping the card to another asterisk server (with exactly the same result) I needed to have the FXS ports working ASAP this morning so I have repaced the functionality of the TDM40b with some Grandstream handytones which I already had in

We are using TDM40B's to connect some devices to Asterisk which depend on caller information arriving as ANI, rather than as Caller ID. I am unsure if the TDM40B supports this in the first place, and if so, I am unsure how to configure it so. I've searched the wiki but couldn't find anything. Can someone please confirm whether or not this is possible? As a fallback I could reconfigure

Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? Kumara

I have problem with a TDM40B installation. When i modprobe wcfxs the error i get is the following: /lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.19-4GB/misc/wcfxs.o: insmod

Hi. I was testing my brand new TDM40B and noticed that when the kernel module is loaded (so the card is powered on), the chip onto the ringer module (the one below the S/N label) is pretty hot (I can't touch it for more than 2 seconds). Is that normal ? Or too much hot could be dangerous ? Matteo Matteo Brancaleoni Espia - Emmegi Sys Admin http://www.espia.it

I have 2 xp100's and one TDM400P. I've plugged a phone into the tdm40b, and when i take it off hook sometimes i get a dialtone, other times i get the message "Power alarm of module2, resetting" spit out to the console from the wcfxs driver does anyone know what this could be? I've tried two different phones, with the same result on both. I changed the signalling to

Hi, I get the following message when I try to run Asterisk. This happened after I changed some of the configuration files: extensions.conf, zaptel.conf and zapata.conf. I have both TDM40B and X100P cards and am trying to use them together. Please let me know if there is something else I should do. Thnx, Gladys [chan_local.so] => (Local Proxy Channel) == Registered channel type

Morning, I've installed asterisk on FC4 and had a few problems with zaptel stuff. Having installed it on SuSE I was able to check a few things that were different. When installed with the RPMS, the udev stuff gets put into the normal 50-udev file under /etc/udev. On SuSE, it worked when the necessary zaptel section was in a separate file name 55-zaptel.udev (the name is arbitrary,

OK, I have a Asterisk @ Home 1.0.7 server with two Digium TDM400 cards, one 4 port FXO and one 4 port FXS. When I plug an analog cordless phone into the TDM40B card and setup a ZAP extension, the phone rings in and you can answer just fine. The weird part is when you try to get dialtone from the cordless handset it rings all of the other extensions... One other thing to mention is that the

Hi list, I'm trying to set up a TDM40B, but modprobe returns the fowling errors: asterisk:~# modprobe wcfxs ZT_CHANCONFIG failed on channel 1: Invalid argument (22) /lib/modules/2.4.18/misc/wcfxs.o: post-install wcfxs failed /lib/modules/2.4.18/misc/wcfxs.o: insmod wcfxs failed Could someone give me a help? Thanks in advance Eduardo

Hi, I'm no phone guy but I've been having quite a bit of fun messing with Asterisk. I just unwrapped my new Wildcard TDM400P (TDM40B) which is the new 4 port FXS PCI card. It has 4 RJ45 connectors and I'd like to hookup 4 regular phones which are RJ11. If I crip RJ45 connectors to the phone cords do I simple use the center two pins (pins 4 and 5)? Thanks, Randy Smith Tiger

Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not

Hello, I've got an Asterisk server with a TDM400P with 2 FXO modules and 2 FXS modules. This server is connected to 2 PSTN lines and 2 analog phones. In my Zaptel configuration, I've defined 2 groups : one for the 2 FXO's and one for the 2 FXS. The asterisk server is just used to add a little IVR and Voicemail service. Eveything works fine, but sometimes the conversation is

Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything "seems" to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in