I was just wondering if you know of any site/forum that is informative about Asterisk. I have just configured an Atcom IP04 in which Asterisk is embedded. I need to know how to configure the IP-PABX to receive fax and come up with a Skype Gateway to receive/call out on Skype.
View 2 RepliesI have two asterisk servers each one behind a linux firewall/gw. Linux is Centos 5.4, kernel 2.6.18-164.el5, iptables v1.3.5. Routes on the fws are ok and when iptables is stoped the servers are see each other, all good. But when I run iptables script in any fw, one server (not always the same) goes unreachable. I verify this with asterisk -r, then show sip trunk, and status becomes UNREACHABLE.
Iptables scripts is generated by fwbuilder. The weird part is I put only one rule to de script and it looks like Source=any, Destination=any, Service=any, Interface=any, Direction (Inbound,Outbound)=any, Time=Any, Action=ACCEPT. So as you can see I tried something like "Do not do anything at all". But anyway I run the script in any fw and one server becomes UNREACHABLE. I think the script does something wrong after all or maybe I have some missconfiguration in my asterisk conf files. The point is I am not so expert in iptables or shell scripting so I can't see anything in the iptables script. I have look for some issues like iptables blocking because of ip_conntrack table full, or "dont fragment" bit set in kernel problem, but nothing seems to be the right problem at all.
I am encountering a strange problem on my VOIP setup Basically, I have a asterisk appliance IP04. I have setup all the extensions and everything. I use a Linksys PAP2T as an ATA remotely. Now, my problem is the ATA sometimes is okay can call SIP and PSTN but sometimes I just can't hear anything. I thought it was my ISP blocking the VOIP packets but I have tried both the SIP softphone and IAX2 softphone on my PC. For IAX2, it works perfectly however in the SIP, I can hear the other end but they cannot hear me.
These are the ports I have opened on my router
1.) UDP 5060 - SIP Port
2.) UDP 10000 - 20000 - RTP Port
3.) UDP 4569 - IAX2
Do I need to open both TCP/UDP for these ports or UDP should be enough? These are the test cases:
1.) Using my WiFi Connection and a analog phone connected to ATA --> Sometimes working sometimes not and sometimes you can call SIP but the other end cannot hear you
2.) Using IAX2 in WiFi connection --> This one works perfectly
3.) Using a mobile phone connected to WiFi Network --> The same...but you can call and go out on PSTN but the other end cannot hear you
4.) Using a mobile phone connected via 3G --> Works perfectly but as expected it is quite slow and voice quality is awful
I want to use SIP rather than IAX2 because it is widely used and since my ATA doesn't support IAX2. Are there other ports I need to open or configure?
Today my boss come to me and ask me to get a cote to upgrade our old "Cisco call manager" (2004) now "Cisco Unified Communications Manager". So I was wondering, instead of doing a costly upgrade (over 35 000�), maybe it's time to change... Does anyone of you got some insight with Asterisk in an enterprise environment? Is it reliable? Following you own judgement, what are the + and - ? If Asterix worth it, what argument (apart of the price) could I use to help the management turning on my side? Will the Cisco 7921 VOIP phone will be able to connect to it? (as we do have over 35 of them)
Enterprise environment:
- 3 sites (VPN interconnected)
- ~35 VOIP phones and ~10 landlines phones
Is there another SIPS/VoIP program that can be used to speak to someone on Skype?
View 1 Replies View RelatedI'm using tcpdump and tcptrace to track all incoming and outgoing data packets through my network interfaces. But I fail to monitor the voip data for skype that way, although it works well with http port 80, for example.
I want to track the ip address of the data packets for skype, i.e. know the ip address of the other one speaking at the other end of skype. How can I achieve this?
I've checked the port setting in my skype and I'm sure I'm listening on the right port. But nothing is showing up while I'm in connection with skype.
Does anyone notice a degradation in sound quality with skype and pulseaudio? It seems that when I use pulseaudio there are bugs in the sound quality in skype. I also notice a lag when talking to someone with the VoIP feature.
View 2 Replies View RelatedI compiled and installed the Asterisk ztdummy package because there is no rpm for it, unfortunately, and i even reinstalled asterisk, but i still get the "No application 'Meetme' for extension..." error when trying to conference. I do a "module show", and it lists other modules that were compiled with the zt source, but not ztdummy.Does anyone know how to fix this? This is more than a passing interest or hobby, because i need to conference about 3 to 5 people to help me test a new Website Content Management System and User Forums Management System i am about to launch as a service.
View 3 Replies View RelatedI want to use asterisk as a PBX and SIP server using Skype. I have everything planned out but I am having a problem loading the "Skype for Asterisk" module provided by Digium. This module costs roughly $60 and Digium provides its customers with free support. After calling Diguim, the technician was unable to find the problem and said that I "should use a different Linux." I found this advice very hard to swallow as I have grown very fond of my openSUSE.I searched the Digium forums and found that another user had the very same problem with Ubuntu. We compared notes in this forum thread:
[URL]...
We were unable to figure out the problem until another member pointed to a Kernel bug in Ubuntu as the source of the problem. The Kernel where this bug exists is 2.6.31-11. Here is a link with some information on the bug:
My webcam works. It works in Cheese and apparently the person I connect with on skype can see my feed. They claim their webcam works fine, but I can't get their feed: instead, I just get a white box. When I try to test my own webcam, I get a similar white box. I think there's something wrong with the way skype is configured on my laptop, but I can't figure it out.
I'm running ubuntu 9.04 on an acer laptop with an acer crystal eye webcam. I installed skype by downloading the binary directly from the website.
I installed Skype 2.1.0.81 using Software manager on Ubuntu 10.10. Everything is working fine with exception of incoming video. All incoming video is displayed as a black screen. From Google I learned that this is a typical issue with Skype on Ubuntu. I tried the instructions but none of them are working for me.
View 1 Replies View Related1) I made some tests and the webcams works, both other people and me are not able to see my image (any library for video decoding missing?). The cam works fine with cheese.2) Integrated microphone doesn't work, whereas an external one works. The integrated microphone works with "sound recorder" application.My is a Packard Bell netbookCPU Atom N270 @1.6GHz, 1GB RAM, 160GB HDO.S. Ubuntu Karmic Koala 9.10
View 9 Replies View Relatedi installed asterisk with yum, the command i used is that yum install asterisk*
Now i want ro remove only asterisk, not its depencies. how i do this?
hJust installed brand new Centos 5.6 and skype, according to centos wiki (http://wiki.centos.org/HowTos/Skype) Getting these errors: skype: /usr/lib/libstdc++.so.6: version `GLIBCXX_3.4.9' not found (required by skype)
[root@thinkpad opt]# rpm -q libstdc++ --provides
libstdc++ = 4.1.1-52.el5
libstdc++.so.6()(64bit)
libstdc++.so.6(CXXABI_1.3)(64bit)
libstdc++.so.6(CXXABI_1.3.1)(64bit)
[Code]...
I've read and execute the info write in this guide :
[URL]
When I type skype , I receive this message :
bash: skype: command not found.
There are 3 versions of Skype for Ubuntu on the Skype download section. Does anybody know which is the correct one for Ubuntu Karmic?
View 5 Replies View RelatedSince there is no notification area any more in 11.04, where is the skype icon of the running skype instance?
View 2 Replies View RelatedI have a USB 3g modem, which also supports voice calling, I want to implement IVR ( Interactive Voice response) with it on linux, Can I do it with Asterisk? If yes, then please help me on how to do this ( references) ?
Is there example or howtos available to do this?
Can I develop IVR system in linux without Asterisk?
Note: I am very new to Asterisk and IVR.
What are the standard procedures to monitor asterisk with nagios? I downloaded check_asterisk from URL...
View 1 Replies View RelatedI just installed asteriks on my remote server ,and set up 2 account for test purpose.i added these lines to sip.conf
Code:
[benchabane]
type=friend
username=benchabane
secret=tcpip13
host=dynamic
context=mygroup
[Code]...
I recently bought a IOgear USB bluetooth adapter and a Sony DRBT101/BLK bluetooth headphones to use with my computer.Whenever I try to make a call using these headphones with skype, I can speak for say 10 seconds and then the headset disconnects completely from the system. I then have to reconnect the headset by switching it off and on. I do not have this problem listening to music using A2DP. It works fine with my mobile phone.When I change the profile to Telephony Duplex (HSP/HFP), it shows up as hands free, I thought it should be headset in blueman manager. My friend uses Plantronics headset with is Dell laptop with built in bluetooth and he says his comes up as headset not hands free. Could this be causing the problem?
View 1 Replies View Relatedget Skype to work on 64 bit have downloaded and installed the 32 bit which installed fine but just wont load
View 2 Replies View RelatedDoes anybody know a good option for a Skype based solution that works on Linux? I have a real problem with the current Linux version as it is a crappy, buggy and downgraded version of the Windows equivalent.
View 7 Replies View RelatedI just recently purchased a 3g dongle for my netbook. Everything seems to be working fine until I discovered this strange problem. I cant seem to text via skype, it doesnt go through. I testet the skype test call service and voice works fine, audio and mic. I have tried to google and search forums but no luck. I have before used skype on my netbook with my cellphone as a hot spot and it worked without problem but it doesnt have the same provider - different sim cards in phone and dongle.
View 2 Replies View RelatedI want to authenticate and authorize for SiP user of my * box, it is my project, but i did not finish it, hic hic! My asterisk box is version 1.6.0, fedora 10.the first, i downed and installed freeradius to my computer, and run "radiusd -X", it's run normally. Then, I installed radiusclient-ng0.5.6, it's rpm file. All is perfect. But the problem is i can't run asterisk box as radiusclient.
View 2 Replies View RelatedI have been reading about asterisk, I did some basic configuration, a small ivr, record messages.. but I was wondering how and what/where should I modify to use a database to save sip users and voicemail user..then we can add user/voicemail with php-myadmin I always search but I cant find about this configuration..
View 1 Replies View RelatedMy asterisk version is 1.6 and Web-MeetMe is 4.0.2 I get the following upon issuing "make" in web-meetme/cbmysql/ :
Quote:
cc -pipe -I/usr/include/mysql -L/usr/lib/mysql -fPIC -I/usr/src/asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_cbmysql.o app_cbmysql.c
app_cbmysql.c:14:22: error: asterisk.h: No such file or directory
app_cbmysql.c:25:33: error: asterisk/autoconfig.h: No such file or directory
app_cbmysql.c:26:27: error: asterisk/lock.h: No such file or directory
[Code]....
Has anyone ever had any luck with htpasswd and htaccess on asterisk, I set it up on a test apache server in VMWare just to make sure I knew what I was doing, so It was a very basic html page that I used, however, when I go to implement it on one of my Asterisk Servers, It comes up with the following page after I type a user name and password credentials in:
***************************************************************************
Internal Server Error
The server encountered an internal error or misconfiguration and was unable to complete your request.
Please contact the server administrator, webmaster@localhost and inform them of the time the error occurred, and anything you might have done that may have caused the error.
More information about this error may be available in the server error log.
Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny4 with Suhosin-Patch Server at fred Port 80
***************************************************************************
Anyone know if its compatible? and if so, is there any tricks around this?
I want to install Asterisk gui using svn command, but I have the following error:
# svn co http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui
svn: PROPFIND request failed on '/svn/asterisk-gui/trunk'
svn: PROPFIND of '/svn/asterisk-gui/trunk': Could not resolve hostname `svn.digium.com': Temporary failure in name resolution (http://svn.digium.com)
We would like to install Asterisk with a minimal centos build. We are writing kick-starts to install packages, does anyone know or have to hand a packages list which installs minimum system requirements for asterisk? Preferably this will not exceed the contents of disk 1
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