similar to: PRI TE110P Configuration (Solved)

Your right with this part But as I also have some knowldge on other parts but ms , *nix etc I know it is nowadays possible for almost every email client to correctly display html email. And be honest does it not read more easy if you have a nice font and some markup available? I know mailman is an old package and should be more flexible in handling and distributing html email. For standards:

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

On Sat, Aug 29, 2009 at 10:30 PM, <asterisk-users-request at lists.digium.com>wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help'

So, I have been testing ekiga 3.0 with Asterisk, and sadly, it don't work. I am told by the ekiga devs in http://bugzilla.gnome.org/show_bug.cgi?id=553595 and http://bugzilla.gnome.org/show_bug.cgi?id=553810 that the problem is that Asterisk does not support SIP forking. The issue is that I have multiple addresses on my workstation: 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500

Topic 6: RE:unable to execute command hi there i tried to execute the command as suggest like exten => 1987,1,Playback(posix-restarting) exten => 1987,2,wait(1) exten => 1987,3,System(/usr/bin/python /home/docas/Desktop/mess1.py) exten=> 1987,4,Hangup it still doesn't work,now it does it give unable to execute command but it doesn't reach the system command it just

I'm using it rather successfully. Not perfect, but it works. It is limited to WiFi connectivity... at least here in Spain I cant get either client to work over 3G. I'm using Fring and Truphone. Although I have only configured a SIP to my Asterisk with Fring. Skype works fine. We tested with several Nokia 5800 (EM) using Fring. Call quality is worse. At best, we have a 1+ second delay.

Dear Asterisk users, We would like to announce that Druid, Open Source Unified Communications project has just made a major release: Druid 2.0. It is out!It has a ton of new features and a highly improved interface. Asterisk stability has also been greatly improved. For more info http://forums.voiceroute.org/showthread.php?t=837 Some of the key features - Improved Web GUI, faster and smoother -

On Fri, Sep 16, 2016 at 5:55 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Tried with both softphone (Ekiga) and snom IP phone, contact header > contains the public IP. but from header contains the private IP. after > OPTIONS method sent by the server. client sends an Register with expires 0. > Ok, did setting from_domain work? > > Best

... Activating "sip debug" shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named

tnet.itand SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo < gincantalupo at

Or even: <a class="moz-txt-link-freetext" href="http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946">http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&amp;mid=4946</a> (same thing from the UK site:) <a class="moz-txt-link-freetext"

That sounds like the ideal app for me too. Fring requires we register with Fring and give them user id/password pair. In our case it did not work until we put a public IP on our Asterisk. I just bought WeePhone and I'll give it a try on the iPhone. Cheers, Enrique -----Mensaje original----- De: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] En

ast_waitfordigit that accepts milliseconds as input. Douglas Garstang wrote: > Admittedly I have not used the ExternalIVR app. Is it any good? > > I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, > it can do it, but boy it is UGLY. There's also the fact that you can't > call Backgound() in a macro, which forces you to use Read() which >

<br> Or even:<br> <br> &nbsp;<a href=3D"http://www.blackbox.com/Catalog/Detail.aspx?cid=3D425,1423= ,1424&amp;mid=3D4946" target=3D"_blank">http://www.blackbox.com/Catalog/Det= ail.aspx?cid=3D425,1423,1424&amp;mid=3D4946</a><br> <br> (same thing from the UK site:)<br> <br> <br> <a

Hi, We have recently setup a Live Demo of DRUID our Asterisk management interface product. Also I'd like to thank all of you that took the time to download the trial edition and give us your feedback. WE've tried to incorporate as much of that feedback into our new updated release. Feel free to download the trial, checkout the live demo or buy a copy :)

an external program, which at this stage, is not customizable ... I don't know if alternatives (LiMO, Android, ...) would be more open to this customization but for Symbian, not only Nokia SIP client wouldn't let you autoanswer to SIP calls, but any other SIP client complying to Symbian design wouldn't support autoanswer. PS: Please, note that I'm far from being an expert in GSM

mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? > > However, the MWI does not indicate voice mails for 610 and I keep seeing > this error message: > > ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox > 610 in context a10 > > However, mailbox 610 is clearly defined in voicemail.conf: >

Astricon is only 54 days away! If you're not booked, please take a moment to register for the conference, get your hotel room, and get your plane tickets before things fill up and/or get expensive. This is a great opportunity to meet other developers, users, and members of the Asterisk ecosystem, and I encourage everyone to attend. While there are great things to be said about the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

I get congestion (same error) with exten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r) not dialing 1 exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r) dialing 1 exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r) dialing 9 All the same == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [9544790554 at To_Airspring:1]