similar to: asterisk on the firewall/router

Hi, the app HasNewVoiceMail can't find my voicemail. This is the errormessage: Dec 3 14:24:01 NOTICE[12222481]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 25 at /var/spool/asterisk/voicemail/default/25/(null) does not exist however this is the output of lspbx:~# ls -l /var/spool/asterisk/voicemail/default/25/ total 316 -rwx------ 1 root root 11814 2003-11-22 18:18

Good morning, I am trying to use the Windows iax client. My iax.conf looks like this: [general] port=5036 bindaddr=10.1.3.111 bandwidth=high allow=gsm ; Always allow GSM, it's cool :) tos=lowdelay [pos| type=friend context=default auth=plaintext secret=pos deny=0.0.0.0/0.0.0.0 permit=10.1.3.0/255.255.255.0 host=dynamic defaultip=10.1.3.2 In the registrations dialog

Hello, today I tried a DIAX -> * -> DIAX connection over the internet (768/128 ADSL connection on both sides). The sound quality was great. However, we had some latency problems, and also, if both sides where not talking the first words had some problems getting thru. Is this expected, is there anything that can be done on our setup, any magical iax.conf entry? Thanks and best regards

I have my asterisk box on the public network. I have a winders box on the public network, running diax. I have a winders box, same setup, behind my Linux iptables firewall, on a private network. Both boxes cann register iax2 to asterisk, and dial, but as soon as asterisk tries to do the native bridge, the call fails. I've poked at this way too much today... what ports do I need to open for

After apt-get update && apt-get upgrade -y wget http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.5c.tar.gz tar xfvz chan_capi.02.5c.tar.gz cd chan_capi-0.2.5c make && make install shutdown -r now asterisk seg faults upon calling in via ISDN. Any ideas are greatly appreciated. rgds pos

Is there a Debian package available for asterisk-oh323, or the chan_h323? If yes, where might I find one? Thanks rgds pos

Hello, I've been using * for quite a while using AVM C4 card and a few Grandstream Budgetone 102 hard phones plus couple of HandyTone 286. Echo is a big problem. I am getting used to it, but some users complain. Anyone has experience w/ regards to echo comparing the AVM C4 with two HFC-cards? Before shelling out time and money, maybe someone else already has done so, and could tell me if

Hi, does the hint extension work together with the Snom phones in stable? I don't get an error in the dialplan, but it does not work either. On SIP/26 I want to monitor SIP/22. This is what I do right now: extension.conf [incoming] exten => 955,hint,SIP/22 exten => 955,1,Dial(SIP/22) sip.conf [26] ... subscribecontext=incoming ... Running Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b TIA

Good day, *newbie alert* I try to run * with capi_chan (and later on SIP). System: Debian 3.0 / apt-getted most stuff except chan_capi and b1.t4 HW: AVM B1 ISA / Celeron 400 / 128MB RAM Starting * the following ends up in the messages Jul 22 19:47:03 WARNING[16384]: File chan_mgcp.c, Line 2925 (load_module): Unable to get our IP address, MGCP disabled Jul 22 19:47:03 WARNING[16384]: File

Is it possible to pass the Network port to a virtual host and have a minimal Firewall, IDS/IPS install there and this virtual host to be the Gateway to the Network for the host as well as other virtual hosts? I am talking specifically Centos 6.2 & KVM/Xen. Also, what harware requirements need to be met. And most of all what security implications. Normally minimal software at the perimeter of

Hi, I have asterisk behind my primary PBX connected via ISDN (chan_capi). Calling out and calling in works just fine, however I can't connect to my primary pbxs' extensions. If anyone has an example extensions.conf, I'd be grateful for a copy. I tried (the MSN of the ISDN card is set to 30) exten => 22,1,Dial,CAPI/30:22 but this does not work. Changing it to exten =>

Hi, in a HTML page I can write href="mailto:joe@doe.com" and clicking on the link will open the default mail application. Is anything like that possible with any of the soft phones (SIP or IAX [Windows])? Any and all information is greatly appreciated. rgds pos

Do you mean this occurs when traffic is passed over an IPSec tunnel or that it occurs anytime a tunnel is use on a machine that also is passing VoIP traffic (outside the tunnel)? I assume you must mean over the tunnel but I am curious... Thanks, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tim

Good day, I want to have Asterisk as my gateway to the outside world and use another PBX to connect my existing phones. How do I specify the dial string to forward the original Caller ID to over the ISDN to the second PBX? Right now, my extensions.conf looks like this: exten => ${OUTSIDEMSN},1,Dial,CAPI/${MSN2NDPBX}:${EXTEN} How do I transfer the caller Id information initially coming

Well here's a suggestion - a little crazy - but works... Most equipment is taking the 120vac and converting it into DC voltage. So why not just feed it DC voltage directly??? We had a situation where our field techs needed to test dsl circuits and voip ata from the demarcation point outside a house or business. A UPS might have worked - but the down conversion of 12v dc battery in ups up to

On 3/4/2015 12:45 PM, Dave McGuire wrote: > There is. But I already have a firewall, running on bulletproof > hardware that doesn't depend on spinning disks. I don't want to add > ANOTHER firewall when I already have a perfectly good one. Besides, my > mail server is built for...serving mail. Not being a firewall. You can implement whatever type of security you are

HI everyone, I try to get some bootstrap CIs for coefficients obtained by quantile regression. I have influencial values and thus switched to quantreg.. The data is clustered and within clusters the variance of my DV = 0.. Is this sensible for the below data? And what about the warnings? Thanks in advance for any guidance, Kay > dput(d) structure(list(Porenfläche = c(4990L, 7002L, 7558L,

wondering if anybody knows this......does shady dial work only with a zap interface or can it be configured to be used with SIP or IAX. Nauman -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Thursday, July 08, 2004 5:48 PM To: asterisk-users@lists.digium.com Subject:

Hi All, Our FW appears to be blocking my CVS attempts, does anyone have a tar ball they can send me ? Rgds, Adam ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message

Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro Some of the features are: - registering with Asterisk PBX; - can use any audio device as ring device (including PC speaker), independent of the play device;