similar to: X100P installed OK, after added TDM400P Asterisk would no longer start

Hello I am (attempting) to run the astlinux version of Asterisk on a VIA embedded platform. I have a TDM04B and pretty sure zaptel.conf and zapata.conf setup OK. They worked fine with same card in traditional PC anyway. I think need the module wcfxs for a Digium TDM04B card. Is this module not part of astlinux? Do I need to download it? Or is it in opt? I see wctdm - but think that is

Hello, After installing on Ubuntu 10.04 using the tutorial on http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html I have a running instance of Asterisk. PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): MG2 Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2)

Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http:// sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or

I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get: sip show peers Name/username Host Dyn Nat ACL Mask Port Status 202/202 192.168.0.6 D 255.255.255.255 5060 Unmonitored 201/201 (Unspecified) D 255.255.255.255 5060 Unmonitored 200/200 192.168.0.3 D 255.255.255.255 5060

Hello I see it is possible to buy Flash Disks up to 4GB now. Has anyone any experience of building an Asterisk system with a flash disk as the only storage device? Any brands you recommend? Is 2 or 4GB enough for an Asterisk installation? Typically how many MB is required for voicemail recording files for say a 10 user system? What about voicemail - I suppose files could be emailed and

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add

I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4, and Zaptel 1.4 The Digium cards installed are TDM2400 and TE110P. Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9 Now when I run ztcfg I get the following error message: (CAS signalling on span 2 conflicts with Clear channel on channel 40) --NOTE: signaling was spelled wrong in the error

Hello I am using a VIA Epia ME6000 with a 600MHz Eden Fanless CPU. Is this likely to be enough power for a 8 extension system with 6 external pstn lines? How important is cpu? Is there some measure, eg xMHz CPU per extension or something benchmark? I have installed 512MB memory - again any benchmark for asterisk memory usage? Angus

Hello, I have another little problem with my ZAPs channels, in fact, when I received a call, I heard no sound while in the CLI, sound is played: -- Starting simple switch on 'Zap/4-1' -- Executing [s at from-zaptel:1] Answer("Zap/4-1", "") in new stack -- Executing [s at from-zaptel:2] BackGround("Zap/4-1", "hello-world") in new stack --

Hello I have played about with a TDM400 card and plugged in some standard analog phones. I am using the card in FXS mode - for analog extensions. I did notice that one of my phones did not ring and I wondered why. I later read in Paul Mahler's book VoIP Telephony with Asterisk that in his section on the TDM400 on page 127 he says "In the UK, you may need an adapter that provides a

Hello I have downloaded asterisk-addons but when I make install get: cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error:

Hello, AFAIK the feature CD (call deflection) is only possible on point-to-multipoint links, is this correct? I've heard about the feature "partial rerouting" which should do the same on point-to-point-links. Is this implemented in either bristuff or chan-capi(-cm)? Thanks in advance, Karsten

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?

Hello In my extensions.conf file: [frompstnisdn] exten => s,1,Dial(SIP/200&SIP/202,20) exten => s,2,Voicemail(su200) exten => s,3,Hangup I use the s, start, extension to handle incoming calls. In my zapata.conf: context=frompstnisdn This works ok on another asterisk box I setup. But on incoming calls I get: -- Extension '787367' in context 'frompstnisdn'

Hi there! I've set up an asterisk@home with AVM C2 P2P ... Everything works fine ... BUT ... If someone is dialing the PBX head number without any extension, asterisk can't handle this ... the DID in this case is empty .... Any ideas how to handle this? Regards, Marcus Hofbauer -- |** realit?t ist da wo der pizzamann herkommt **|

Hello I am running Asterisk on SUSE Linux Professional 9.3 on a VIA Epia M Series motherboard - CPU runs at 1GHz. There is no fan - just a large heatsink. Currently system is running off standard IDE hard drive - because I couldn't get astlinux to run with my Digium TDM04B card (only PCI card in system). Strangely I also have the same system also running SUSE Linux running as a file

Hello I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more

Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. In the SPA2000 I have set dtmfmode to be inband. I notice that with the asterisk you dial a number and then it waits for a timeout