similar to: Outgoing call to IVR not being "answered"

Hi All Has anything been done to fix the issue where the * box is sat behind a nat firewall? Regards Mark

Hi everyone, Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing. Tomica ---- This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr

Hi, I have a Cisco 7920 that I'm trying to get working with my * box. When the phone boots it requests XMLDefault.cnf.xml and SEP<MACADDRESSHERE>.cnf. I assume I set the line number, etc in the latter of the two. However I cannot find any reference to how this file is structured. Anyone know? I assume this is why I'm getting the errors below: Oct 17 19:47:24

Might be a stupid question, but is there a default gateway set on the 7960? -----Original Message----- From: Paul Mahler [mailto:pmahler@signate.com] Sent: Thursday, December 18, 2003 7:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 - can't traverse NAT? I have a 7960 running behind a firewall running NAT. From a telnet session to the 7960, I can't ping

My unit's firmware: library 05.03, tape d22h, shows the device as set to random mode. But mtx -f /dev/st0 status gives an error that google says the device is in sequential mode. dmesg|grep -i hp does reflect CentOS thinks the device is a sequential unit. I've tried this with Fedora 8, too, and both show the same, so it is either an issue with CentOS/Fedora RPMs, or the version of

Does anyone in Asterisk land know how to turn on the message light on the back of the earpiece of a cisco 7960 when a message is waiting? Thanks! Paul Paul Mahler mail:pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040130/0efacc79/attachment.htm

Hi, Is there is some form of VAD on * for SIP channels, cause I have a problem with MOH. I made an extension which simply plays MOH, when I dial that extension with my ATA188 MOH sounds choppy if I talk on the phone the MOH keeps playing. I saw the sip channel (show channel SIP/*) and I see no packets going in/out when I talk then packets shows going in/out. I don?t have this kind of problem

Hi All, I'm using Asterisk 1.8.2 with outbound calls using Google Voice. I've been having issues calling several toll free numbers where the call 'is ringing' but never transitions to 'answered'. These are toll free numbers which are typically answered by an ivrs where you enter eg. a conference bridge number. I searched google and the closest reported issues I found are

I have a cisco 7940 with the following sip.conf config: [Desk1.1] type=friend secret=****** defaultip=192.168.1.14 insecure=no mailbox=102 callerid="Desk1.1" qualify=500 canreinvite=no context=extensions host=dynamic group=2 I do not get message waiting indicator (mwi) on this phone. Is the another .conf file invilved in configuring this function other than the mailbox=xxx in the

Hi All, I am not sure if this is the same thing as the hang on exit bug, so sorry if this is a duplication of previous stuff. Essetntially I am experiencing ssh hangs with about .5% - 1% of my connections. I am running 2.9p2, on Solaris 7. I actually have empirical data on the hangings, as I wrote a script to create these connections in an endless loop, setting an alarm so I could recover

Why does voicemail prompt me for an extension instead of just asking my password? [voice-mail] exten => 99,1,VoicemailMain(${EXTEN}@inside) exten => 99,2,Hangup Paul Mahler pmahler@signate.com <mailto:pmahler@signate.com> <http://www.signate.com/> Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training & Consulting

hi, I saw that Mike Dransfield tried to port 'state' plugin from beryl. What does it do? It should be able to place windows, based on name, class etc., to specific viewports. I recently converted to compiz window manager from WindowMaker (used it for 8 years), and I miss automatic 'pinning' of specific windows to particular workspace (or viewport in this case). Mike's

I have one SIP extension that can't logon to voicemail. The log file says -- Incorrect password '3213' for user '4035' (context=other) even though the context in voicemail.cnf says 4035 => 3213,Bill Smith Thanks! Paul Mahler mail:pmahler@signate.com phone: 650.207.9855 fax: 877.408.0105 -------------- next part -------------- An HTML attachment was

Slash-dot is pointing to this article on Asterisk and Pingtel. http://www.theregister.co.uk/2005/05/22/pingtel_voip/ Paul Paul Mahler www.signate.com

I need to dial an extension that tells me what extension I'm dialing from. I'm running a bunch of analog phones off a channel bank to * over a T1. I have the following in extensions.conf. exten => 98,1,SayDigits(${EXTEN}) This says the digits the caller enters on the keypad, not the extension they are calling from. Thanks Guys!!!!!!!! Paul Paul Mahler pmahler@signate.com

Hi, I?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing

Hi list. I'm new to IVRs and trying to set up one that toggles an auto-forward flag on or off for specific accounts. I'd like to have my users dial an extension and then be prompted to enter the account number. (done) Next I'd like it to jump to the appropriate line in the dial plan that corresponds to the entered account number (if it is valid) and have it play back the

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

Some of you might already know that we are releasing a new phone, snom 360. To make the phone well-known and stable, we have made a soft phone version out of it and offer it for trial or private use for free (for more details, see the license conditions). There are only few limitations to the phone. First of all, the audio subsystem will work only work with an acceptable quality if you are using

Hi, I'm using the new SipAddHeader application on Asterisk 1.2.1, here's a snip of my extensions: exten => _9XXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM} exten => _9XXXXXXX,2,SipAddHeader(P-Asserted-Identity: tel:${CALLERIDNUM}) exten => _9XXXXXXX,3,Dial(SIP/${EXTEN}@${SIPTRUNK},,tT) exten => _9XXXXXXX,4,Congestion The problems is that Asterisk