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Alex Rudd Segmentation fault after registering and loading g729a codec on x86-64 clean install
I am new to Asterisk and have purchased a g729 codec from Digium for use with a clean install of Asterisk. I am using FreePBX-64bit-6.12.65.iso Distro on a Dell T110 server. Everything installs and runs correctly.
I have followed the steps in the g729 README:
1) Download and execute the 'register' utility to generate a valid license.
2) Download and execute the 'benchg729' utility to determine the optimum build.
3) Use the 'G.729 Selector' web utility to determine the recommended G.729 codec binary download package (barcelona_64)
4) Download and install the 'codec_g729' binary that is built for your platform.
I have extracted the codec and copied it to the /usr/lib64/asterisk/modules/ directory. After doing this, I am loading the codec with the following command:
[root@localhost ~]# asterisk -rx "module load codec_g729a.so"
When I then try to enter the asterisk CLI, I am getting the following error:
[root@localhost ~]# asterisk -r
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
Rebooting the server brings up a looping error refering to /usr/sbin/safe_asterisk: line 158: 17898 Segementaiton fault (core dumped) and a scrolling text.
Can someone suggest what I can do to troubleshoot this further please, and also if this might be related to trying to use a 64-bit version of FreePBX rather than the 32-bit version?
thanks!
Khanh Pham G729 codecs
Hi there,
I'm having issue with VOIP bandwidth, so i'm plaining to change all G711 codecs to G729.
I'm working for Call Center which is using vicidial base on Asterisk 1.4.x.
My question:
1) If I buy G729 codec licenses for Asterisk, do I have to install all softphones that support G729 OR G729 codec on asterisk will translation G711 from softphone to G729 before sending the call to VOIP Provider?
2) I have asked someone in digium, he said we can use softphone with G729 instead of implement G729 on asterisk. So, my concern is the Predictive Dialler Vicidial makes call through G729 codec or not? If it's not, so i will stuck back the the bandwidth issue.
Thanks in advance for reply.
Regards,
KhanhP
I'm having issue with VOIP bandwidth, so i'm plaining to change all G711 codecs to G729.
I'm working for Call Center which is using vicidial base on Asterisk 1.4.x.
My question:
1) If I buy G729 codec licenses for Asterisk, do I have to install all softphones that support G729 OR G729 codec on asterisk will translation G711 from softphone to G729 before sending the call to VOIP Provider?
2) I have asked someone in digium, he said we can use softphone with G729 instead of implement G729 on asterisk. So, my concern is the Predictive Dialler Vicidial makes call through G729 codec or not? If it's not, so i will stuck back the the bandwidth issue.
Thanks in advance for reply.
Regards,
KhanhP
Best Answer chosen by Denis (Sangoma) Denis (Sangoma) > 1) If I buy G729 codec licenses for Asterisk, do I have to
> install all softphones that support G729 OR G729 codec
> on asterisk will translation G711 from softphone to G729
> before sending the call to VOIP Provider?
If you install G729a codec in Asterisk, Asterisk will get the ability to transcode any other (supported) coded into G729a - as long as you have available licenses. Therefore you could have a call to your VoIP provider in which the first leg (Soft-phone and Asterisk) will be in G711 and the second leg (Asterisk and VoIP provider) would be in G729.
Keep in mind that transcoding is a resource intensive operation, you will need to have a significant processor power to preform this operation with a high number of calls.
If you plan to have more than 60 concurrent calls on a system, we suggest to use TC400 card. For more information about this card please check the following link.
http://www1.digium.com/en/products/telephony-cards/voice-compression
> 2) I have asked someone in digium, he said we can use
> softphone with G729 > instead of implement G729 on
> asterisk. So, my concern is the Predictive > Dialler
> Vicidial makes call through G729 codec or not? If it's
> not, so i will stuck back the the bandwidth issue.
This option is possible, but you need to have a very clear understanding of how your Asterisk implementation works. (How calls are being placed and how the system transfers them, which Asterisk applications are being used, what the dial-plan does, call bridging, etc )
As I was mentioned before, installing G729 codec in Asterisk gives Asterisk the ability to transcode any other (supported) coded into G729 and viceversa
If your Asterisk implementation doesn't require audio manipulation (such as, recordings, voice detection, volume changes, play MoH, etc) you could use the pass-through feature in which Asterisk will "just forward" the RTP packets in G729 from your VoIP provider to the extension without having an understanding about what the RTP media contains.
This option is not suggested or it won't work in package solutions or Asterisk distros (such as FreePBX, Elastix, Trixbox, AsteriskNOW, etc) because you are not in control on the call processing (the GUI does it for you).
You normally see this kind on implementation on system that were designed and built from ground up (compile the code, create your own and unique dialplan, etc)
> install all softphones that support G729 OR G729 codec
> on asterisk will translation G711 from softphone to G729
> before sending the call to VOIP Provider?
If you install G729a codec in Asterisk, Asterisk will get the ability to transcode any other (supported) coded into G729a - as long as you have available licenses. Therefore you could have a call to your VoIP provider in which the first leg (Soft-phone and Asterisk) will be in G711 and the second leg (Asterisk and VoIP provider) would be in G729.
Keep in mind that transcoding is a resource intensive operation, you will need to have a significant processor power to preform this operation with a high number of calls.
If you plan to have more than 60 concurrent calls on a system, we suggest to use TC400 card. For more information about this card please check the following link.
http://www1.digium.com/en/products/telephony-cards/voice-compression
> 2) I have asked someone in digium, he said we can use
> softphone with G729 > instead of implement G729 on
> asterisk. So, my concern is the Predictive > Dialler
> Vicidial makes call through G729 codec or not? If it's
> not, so i will stuck back the the bandwidth issue.
This option is possible, but you need to have a very clear understanding of how your Asterisk implementation works. (How calls are being placed and how the system transfers them, which Asterisk applications are being used, what the dial-plan does, call bridging, etc )
As I was mentioned before, installing G729 codec in Asterisk gives Asterisk the ability to transcode any other (supported) coded into G729 and viceversa
If your Asterisk implementation doesn't require audio manipulation (such as, recordings, voice detection, volume changes, play MoH, etc) you could use the pass-through feature in which Asterisk will "just forward" the RTP packets in G729 from your VoIP provider to the extension without having an understanding about what the RTP media contains.
This option is not suggested or it won't work in package solutions or Asterisk distros (such as FreePBX, Elastix, Trixbox, AsteriskNOW, etc) because you are not in control on the call processing (the GUI does it for you).
You normally see this kind on implementation on system that were designed and built from ground up (compile the code, create your own and unique dialplan, etc)
Shoaib Khan G.729 Codec
Hello Sir,
May I switch G.729 Codec from one server to another? We have some hardware problem on existing server so we want to switch on another server.
Regards,
May I switch G.729 Codec from one server to another? We have some hardware problem on existing server so we want to switch on another server.
Regards,
Best Answer chosen by Denis (Sangoma) Denis (Sangoma) Shuaib,
if you are using a Switchvox system, please contact Digium Tech Support, It's my understanding that they should be able to assist you on this issue.
In the event that you have Open Source Asterisk, it's possible to move / migrate the license to another system, but first you will need to remove it from the original system.
Please execute the following command on your Linix Command Line:
#rm -v /var/lib/asterisk/licenses/<License-key>.lic
Example:
# rm -v /var/lib/asterisk/licenses/G729-1234567890.lic
removed `/var/lib/asterisk/licenses/G729-1234567890.lic'
...then restart your Asterisk system.
Once you removed the license in the original system, Please register it to your new equipment according to the G.729 README:
http://downloads.digium.com/pub/telephony/codec_g729/README
In you have any problems registering your license, please contact Digium Tech Support for assistance.
if you are using a Switchvox system, please contact Digium Tech Support, It's my understanding that they should be able to assist you on this issue.
In the event that you have Open Source Asterisk, it's possible to move / migrate the license to another system, but first you will need to remove it from the original system.
Please execute the following command on your Linix Command Line:
#rm -v /var/lib/asterisk/licenses/<License-key>.lic
Example:
# rm -v /var/lib/asterisk/licenses/G729-1234567890.lic
removed `/var/lib/asterisk/licenses/G729-1234567890.lic'
...then restart your Asterisk system.
Once you removed the license in the original system, Please register it to your new equipment according to the G.729 README:
http://downloads.digium.com/pub/telephony/codec_g729/README
In you have any problems registering your license, please contact Digium Tech Support for assistance.
Mettichi Bassem Asterisk loose routing
Hello,
Please could you tell me how to configure asterisk to use loose routing or strict routing?
Best Regards
Mettichi Bassem
Please could you tell me how to configure asterisk to use loose routing or strict routing?
Best Regards
Mettichi Bassem
Best Answer chosen by Denis (Sangoma) Denis (Sangoma) Mettichi,
Unfortunately, I don't think I will be able to provide a solution or workaround on this case. Our Engineering team has confirmed that Asterisk can't be configured to operate either way particularly; it reacts to what the other side is doing, whether it's strict or loose.
I suggest to seek assistance with A2Billing, perhaps they have seen this issue before and could be able to provide a solution.
Unfortunately, I don't think I will be able to provide a solution or workaround on this case. Our Engineering team has confirmed that Asterisk can't be configured to operate either way particularly; it reacts to what the other side is doing, whether it's strict or loose.
I suggest to seek assistance with A2Billing, perhaps they have seen this issue before and could be able to provide a solution.
Best Answer chosen by Denis (Sangoma) Denis (Sangoma) Mario,
Yes. Your Asterisk / Switchvox system should also have a G729 license.
Yes. Your Asterisk / Switchvox system should also have a G729 license.
Mettichi Bassem Asterisk IAX relatime
Hello,
please how can i resolve this problem:
realtime_peer: Failed to parse sockaddr '(null)' for ipaddr of realtime peer 'ipaddr'
Best Regards
Mettichi Bassem
please how can i resolve this problem:
realtime_peer: Failed to parse sockaddr '(null)' for ipaddr of realtime peer 'ipaddr'
Best Regards
Mettichi Bassem
Best Answer chosen by Denis (Sangoma) Denis (Sangoma) Mettichi,
That mesasge means that the realtime peer has no IP address or hostname associated with it. Its NULL in the database.
Please check your database / table configuration in order to fix the issue.
That mesasge means that the realtime peer has no IP address or hostname associated with it. Its NULL in the database.
Please check your database / table configuration in order to fix the issue.
Dongyu He Asterisk 11 calendar problem
Recently I was working on astersik integrating with the calendar,but i meet some problems. myOS:ubuntu 12.04 ,asterisk 11.The CLI messages:can't read the status line.
*CLI> calendar show calendars
Calendar Type Status
-------- ---- ------
myGoogleCal caldav free
*CLI> calendar show calendar myGoogleCal
Name : myGoogleCal
Notify channel :
Notify context :
Notify extension :
Notify applicatio :
Notify appdata :
Refresh time : 1
Timeframe : 30
Autoreminder : 0
Events
------
*CLI>
The config file:
[myGoogleCal]
type=caldav
url=https://www.google.com/calendar/dav/xxxxxxxx/events/
user=xxxxxxx
secret=xxxxx
please help me
*CLI> calendar show calendars
Calendar Type Status
-------- ---- ------
myGoogleCal caldav free
*CLI> calendar show calendar myGoogleCal
Name : myGoogleCal
Notify channel :
Notify context :
Notify extension :
Notify applicatio :
Notify appdata :
Refresh time : 1
Timeframe : 30
Autoreminder : 0
Events
------
*CLI>
The config file:
[myGoogleCal]
type=caldav
url=https://www.google.com/calendar/dav/xxxxxxxx/events/
user=xxxxxxx
secret=xxxxx
please help me
Best Answer chosen by Denis (Sangoma) Denis (Sangoma) Dongyu,
According to the information that you have provided, it seems that you are dealing with a connection issue, I don't think that the problem is caused by a miss-configuration.
According to our Engineering Department, the module (under the hood) is making an HTTP request to the Google calendar, therefore it should be very easy to check by using the following URL on your web browser.
https://www.google.com/calendar/dav/username@gmail.com/events/
Change username@gmail with whatever gmail account you want, go to it, enter your credentials, and make sure you get a calendar blob back
According to the information that you have provided, it seems that you are dealing with a connection issue, I don't think that the problem is caused by a miss-configuration.
According to our Engineering Department, the module (under the hood) is making an HTTP request to the Google calendar, therefore it should be very easy to check by using the following URL on your web browser.
https://www.google.com/calendar/dav/username@gmail.com/events/
Change username@gmail with whatever gmail account you want, go to it, enter your credentials, and make sure you get a calendar blob back
Best Answer chosen by Denis (Sangoma) Denis (Sangoma) Although is technically is possible, Digium does not support this configuration becase is creates problem when you are trying to use telephony card hardware or software licences.
If you are looking information of how to size your asterisk implementation, please read Asterisk: The Definitive Guide ( http://ofps.oreilly.com/titles/9780596517342/ ) On the first chapters talks about the CPU/Memory/Storage that Asterisk could use.
If you are looking information of how to size your asterisk implementation, please read Asterisk: The Definitive Guide ( http://ofps.oreilly.com/titles/9780596517342/ ) On the first chapters talks about the CPU/Memory/Storage that Asterisk could use.
Problem fixed: I was following the README files provided by email when I purchased my license without thinking them through properly. While I had correctly identified that I needed to modify the wget commands for the x86-64 version of the software I was using, I had not modified the link for the version of Asterisk I was using from 1.8.4 to 11.0 therefore
wget http://downloads.digium.com/pub/telephony/codec_g729/\asterisk-1.8.4/x86-64/codec_g729a-1.8.4_3.1.7-barcelona_64.tar.gz should actually have been wget http://downloads.digium.com/pub/telephony/codec_g729/\asterisk-11.0/x86-64/codec_g729a-11.0_3.1.7-barcelona_64.tar.gz for me (note the path to the different Asterisk version).
Once I corrected this, the codec loaded correctly and license shows on my server.