similar to: Asterisk in a DMZ

I think I have a bad FXS module on my TDM400P. Channel map: >> >> Channel 01: FXO Kewlstart (Default) (Slaves: 01) >> Channel 02: FXS Kewlstart (Default) (Slaves: 02) >> >> 2 channels configured. >> ZT_CHANCONFIG failed on channel 1: No such device or address (6) The zaptel loads ok, but the wctdm reports No such device or address and stays loaded. The

Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel Release 11. I need to connect both of them. We are using MFC/R2 for this.. The Diagram: [ NORTEL ] ( AMI ) ------------------------------------------------ (DIGIUM) [ ASTERISK] we have green light at the digium card, and at asterisk we see all 31 channels as idle. But when i want to recive a call from the Nortel to the

Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel Release 11. I need to connect both of them. We are using MFC/R2 for this.. The Diagram: [ NORTEL ] ( AMI ) ------------------------------------------------ (DIGIUM) [ ASTERISK] we have green light at the digium card, and at asterisk we see all 31 channels as idle. But when i want to recive a call from the Nortel to the

Hello, I'm searching for a router for our company. Does anybody has a suggestion for a router with a SIP Application Layer Gateway and good working QoS (Upstream AND Downstream). Regards Bastian

Hello to everybody, does anybody knows how to install Asterisk on a Linksys WRT54G(S)? I had read in the Wiki that it is possible. If somebody has a tip, this would help me very much. Regards Bastian

Hi all again, if i make a query with ... exten => _X.,2,PGSQL,"Query resultid ${connid} SELECT getdest('${EXTEN}')"; ... an error like WARNING[32785]: File pbx.c, Line 1126 (pbx_extension_helper): No application 'PGSQL,"Query resultid ${connid} 'SELECT getdest' for extension (tel1, 00905888, 2) occurs. This looks like an parsing bug. As if the brackets be

Hi all, i want lock Zap channels via global var FREE1 if FREE1 = 1 then call should go on with nothing and waiting for digits to go in _X. Otherwise hangup the channel But if the GotoIf goes to s|4 (NoOp) then comes a ringing indication.... !? The "immediate" property in the zapat.conf is "yes" [tel1] exten => s,1,GotoIf($[${FREE1} = 1]?s|4:s|2) exten =>

hi lee. I see your problem with trunk iax, probably i don't have the solution but i don't knew if you can help me to solve mine. i want to connecte two asterisk server: server A and server B. i want make possible calls betwen all asterisk users.: users in server A with sip number 022100 can phone another sip user in server B with number 037100. this is my config:

The Asterisk Development Team has announced the release of Asterisk 1.6.2.9. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.9. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

Does anyone know a good (and stable) voip gateway product with 4 ports (2 fxo and 2 fxs), with the following requirements: * being able to connect analog phones to the FXS ports, and communicate over SIP with an REGISTRAR/PROXY server (SER in our case). * being able to connect the FXO port to local office PSTN network, and dial to that office pstn number and getting an internal dialtone, or

HI ALL got small question i use call-limit=1 on peers but call limit is not working if user is not registered on PBX and making calls so the main question is -- how to Disallow CALLS without registering on PBX -- Best regards Antony tel. +380669197533 tel2. +380636564340 Paypal http://paypal.me/Satskiy

Hi all, I am working for my master's thesis and I need to do a neural network to forecast stock market price, with also external inputs like technical indicators. I would like to know which function and package of R are more suitable for this study. Thanks a lot for your response, Arnaud TREBAOL. -- Arnaud Trébaol T.I.M.E. Student Ecole Centrale de Lille (09) Politecnico di Milano (10)

Got a strange situation [ext-queues] ... exten => h,2,ExecIf($[${CALLERID(num)} = ' ']?Set(var29=${SHELL(curl -X POST --header "Content-Type: application/json" --header "Accept: application/json" -d "{\"Phone\": ${FROMEXTEN}, \"Source\": \"asterisk\"}" "

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I have a working system with two fxo and two fxs channels. I recenlty got an IAX2 account I would like to use also. While I have gotten the IAX2 channel to "register", it remains non functional, as the incoming calls, go nowhere and the outgoing calls attempt to go out over the ZAP channel. I can see this, via the CLI, with debugs on. I strongly suspect this is a dial plan/config

Hello, I've got problems to install zaptel on a SuSE 9.1 System. The System has got a Linux 2.6.9 Kernel. If I try to load zaptel framework (modprobe zaptel) I get this message: FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format How I can fix this. At compile time, there were no Errors. Regards Bastian

Hello Everybody, I'm using the snom Phones together with Asterisk and I already able to see which Peer is used via "hint" priority. Then a LED on the snom phone is blinking. But I don't see who is calling the other phone. I know that the snom phones are already support this feature. But how I can enable this on Asterisk? Regards Bastian

Hello Asterisk fans, I try to connect an analog modem to Asterisk. The modems are connected e.g. to alarm systems or a cash terminals (POS). As PSTN-Interface I'm using a Wildcard TE110P (E1). Is it possible to connect the modems to an ATA? Which ATA I should use for that scenarios? Cheers Bastian ____________ Virus checked by G DATA AntiVirusKit Version: AVK 17.1806 from 04.01.2007 Virus

Hi there, I've got a small problem with the zaphfc channel. No MSN of an any incoming call which comes trough the ISDN card (Acer ISDN, with HFC chipset and zaphfc driver) which will be forwarded to the SIP-Phone will be displayed. Always it will be shown "asterisk" an the Display. --- snip (zapata.conf) --- [channels] language=de switchtype = euroisdn signalling =