similar to: S100U / wcusb Zaptel driver / Crash / Kernel problem maybe?

Hi, I have not been successful in working out how to dial a # within a phone number. EG: exten => _12345,1,Dial(Zap/1/0868563823#,5,t) or exten => _08XXXXXXXX,1,Dial(Zap/1/${EXTEN}#) I'm trying to append a # character so that I can use a cellsocket (mobile phone to pots adapter) connected to an x100p. I think that asterisk is simply ignoring the # character. The docs on

Hi everyone, This is off topic and is for GS technical support really but it seems that there are a lot of Budge Tone 100/101/102 users out there. I've got a Budge Tone-100 (101 - without the extra 10base ethernet connetion?) here. I changed the configuration through its web based interface and I clicked the reboot link. But then something went wrong and ever since then it doesn't

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

Hi All, Today I got a couple of ZyXEL Prestige 2000W WiFi phones. I'm having a problem making SIP calls although I can receive calls just fine. When I try to make a call the phone makes some sound (like "bup bup bup bup bup bup beep beep") and then I just hear hissing background noise (not too loud - like comfort noise). I upgraded to the latest firmware on the phone - Wj.00.10

Would anyone know why Voicemail in * doesn't get the DTML keypresses from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do with "dtmf_avt_payload: 101" setting in SIPDefault.cnf in the tftp server? Thanks for any help! Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823

Hi everyone, I'd say this question has come up and been answered before but I haven't been able to find it. I have a Cisco 7940 that I've upgraded to SIP firmware (currently P0S-3-06-3-00 - for some reason there was a failure when trying to upgrade to V7 so I left it at V6). The problem I'm having is that when I connect to voicemail the DTMF key presses dont seem to work

Hi Derek, I'm not sure your recording will match with my needs. I wanna be able to do this myself with our currency here. Can you just tell me what to use and how to use it ?? Thanks. Daniel. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Derek Conniffe Sent: lundi 7 f?vrier 2005 11:59 To: Asterisk

Has anyone worked out a way to transfer the Calling Party's Category codes to Asterisk through E1 / T1 connections? I know this is normally available on SS7 interconnects but is it also available to asterisk on the ISDN signalling channels? (I kind of doubt that it is......) Thanks, Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201

Hi all, I have a 7940 running the latest SIP firmware (V7 - thanks Doug Lytle for the tip on the V7 firmware upgrade!). Its almost working perfectly - I can make calls either though my local PSTN or over VOIP but for some reason if I dial my voicemail (which is mapped fine to the VM button on the telephone) it doesn't detect my DTML keypresses so when I press 1 for new messages it just

Hi Ray, I was wondering if the "qualify" option is used [in sip.conf] to keep a connection (from the SIP phone inside the firewall to the Asterisk server outside the firewall) open then would the firewall not allow two way communication without incoming port mapping/NAT (providing that the SIP phone started "talking" first)? I'm not sure about that - I'm being

Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy",

Hi Everyone, I've got a weird problem with both Firefly & iaxLite (both IAX softphones). They don't seem to stop ringing when an incoming call is make to them. If the call is answered the conversation starts both ways but the ringing sound still keeps going and the softphones keep displaying that a call is coming in (but they do not display that the call is answered). I read

Hi all, I've had a strange problem this morning and I know someone who has reported exactly this problem to me too last week: - I've setup a new server running Asterisk 1.2.4. Currently there is no Zaptel hardware install (but there will be soon). This server is behind a NAT router on an DSL line. The remote IAX server on the Internet (which handles the call termination / origin)

Hi everyone, I've just upgraded from Asterisk 1.0.X to 1.2.5 (and the matching latest libpri & zaptel from www.asterisk.org). All compiled fine but now I've got a weird problem with my EuroISDN lines connected to a Digium quad E1 card: - Asterisk suggests its answering calls and playing voice prompts to the incoming call but, in fact, its not really - the caller hears a couple

Hi, I'm trying to get an S100U that came as part of the dev kit light working. I have tried compiled EHCI as modules or directly into the kernel but the same problem happens both times: I connect the S100U and run modprobe wcusb - initially nothing happens (great - no errors, etc). Fairly soon the server will die. It seem to be related to communications with the S100U. For example if I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've tested the S100U unit in two different computers both with different USB subsystems. The one it worked the best in used the usb-uhci USB driver. The other system uses the usb-ohci USB driver. To make sure it was not the installation, I used the same hard drive that had asterisk in both systems. And I moved the X101P and S100U between both

Hi everyone, RE: Bandwidth. We have an asterisk server sharing bandwidth with other [web] servers in cabinets that we rent in a large data-center and all is working fine. But I'm concerned that web traffic could affect the VoIP quality (my tests so far haven't showed this [yet!]. Currently I'm running a server with Netfilter (iptables) between all the servers and the Internet

i'm having problems with s100u (wcusb) on Via EPIA-V-8000 running RedHat Linux 9 (2.4.20-31.9). when i try to dial, asterisk won't respond (still hear dialtone) or get a fast busy. seldom asterisk would process the dialed number. i was able to dial to echo test and i noticed that the line is noisy (like chirping). anyone encounter this? i have no other s100u to check if this is a

Hi all, this is another post about this problem. I installed from scratch a new Suse Linux 10.0, with latest stable asterisk. Moreover I add the lines to /etc/udev/rules.d/50-udev.rules, in order to let the driver create the /dev/zap....... When I plug into usb port my TigerJet adapter, I see on /var/log/messages Feb 24 14:55:02 srvlnx05 kernel: usb 1-2: new full speed USB device using

Hi, I have been trying to figure out why the S100U is not performing very well on RH9.. Here is my thinking..( may be totally wide of the mark but here goes anyway) I remember reading somwhere that the sound system used by RH has changed... Does the S100U not depend on the sound subsystem?? So what I think is that the sound subsystem in RH9 and the S100U are not happy working together.. Does