Re: [asterisk-users] Best VoIP conferencing phone ?
I have tried EyeBeam and it worked fine with x members audio conference however it need resources (Processing + RAM) per additional line. Regards, Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Wednesday, November 30, 2011 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind Subject: [asterisk-users] Best VoIP conferencing phone ? Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best VoIP conferencing phone ?
Hi Faisal, Thanks for reply but I want hardware wase VoIP device. If know please gussed me. From google I fould the list of below devices but I am not sure that these are best for used or have an issue *1)Polycom SoundStation IP 7000 * *Why it's best: *The Polycom SoundStation IP 7000 is the most advanced conference phone from the Polycom SoundStation lineup and leaves little to be desired. With an amazing 20’ 360 radius, the 7000 is perfect for large conference rooms. The new HD voice quality (22 kHz) allows. * * *2) Polycom Voicestation 500* * * *Why it's a best pick: *The Polycom VoiceStation 500 is one of the best conference phones for a wide variety of reasons. The VoiceStation 500 features amazing call quality, 7’ 360 radius, Bluetooth connectivity, wired connection, background noise reduction, and an attractive design. * * *3)Panasonic - 8-Microphone Speakerphone with Caller ID KX-TS730S* * * *Why it's a best pick: *With a 360 10’ radius and 8 microphones, everyone is sure to be heard with the Panasonic KX-TS730S. The multiple microphones allows for everyone sitting in on the conference to be heard uniformly without distortion. * * *4)Cisco Unified IP Conference Station 7937G Conference VoIP Phone* * * *Why it's a best pick: *The Cisco 7937G works via VoIP connection, has stunning call clarity, and features a simplistic but expensive design that is easy to use. Cisco is an industry leader in IT communication products, and the 7937G is no different. The 360 design allows everyone to be heard. * * *5)Polycom SoundStation VTX 1000* * * *Why it's a best pick: *The SoundStation VTX 1000 is an incredible conference phone, but it is very pricey and not as good as advertised. The VTX 1000 is designed for large conference rooms and features upgradable software (which is a huge benefit since the cost is so high), 20’ 360 radius. 6)Polycom® SoundStation® IP 5000* 7) GXP2120 6-line Executive HD IP Phone* On Wed, Nov 30, 2011 at 1:47 PM, Faisal Hanif fai...@vopium.com wrote: I have tried EyeBeam and it worked fine with x members audio conference however it need resources (Processing + RAM) per additional line. ** ** Regards, ** ** Faisal Hanif ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati *Sent:* Wednesday, November 30, 2011 11:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind *Subject:* [asterisk-users] Best VoIP conferencing phone ? ** ** Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best VoIP conferencing phone ?
In hardware I used some snom phones up to six lines. You can check on http://www.snom.com/ for appropriate model. Regards, Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Wednesday, November 30, 2011 1:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best VoIP conferencing phone ? Hi Faisal, Thanks for reply but I want hardware wase VoIP device. If know please gussed me. From google I fould the list of below devices but I am not sure that these are best for used or have an issue 1)Polycom SoundStation IP 7000 Why it's best: The Polycom SoundStation IP 7000 is the most advanced conference phone from the Polycom SoundStation lineup and leaves little to be desired. With an amazing 20' 360 radius, the 7000 is perfect for large conference rooms. The new HD voice quality (22 kHz) allows. 2) Polycom Voicestation 500 Why it's a best pick: The Polycom VoiceStation 500 is one of the best conference phones for a wide variety of reasons. The VoiceStation 500 features amazing call quality, 7' 360 radius, Bluetooth connectivity, wired connection, background noise reduction, and an attractive design. 3)Panasonic - 8-Microphone Speakerphone with Caller ID KX-TS730S Why it's a best pick: With a 360 10' radius and 8 microphones, everyone is sure to be heard with the Panasonic KX-TS730S. The multiple microphones allows for everyone sitting in on the conference to be heard uniformly without distortion. 4)Cisco Unified IP Conference Station 7937G Conference VoIP Phone Why it's a best pick: The Cisco 7937G works via VoIP connection, has stunning call clarity, and features a simplistic but expensive design that is easy to use. Cisco is an industry leader in IT communication products, and the 7937G is no different. The 360 design allows everyone to be heard. 5)Polycom SoundStation VTX 1000 Why it's a best pick: The SoundStation VTX 1000 is an incredible conference phone, but it is very pricey and not as good as advertised. The VTX 1000 is designed for large conference rooms and features upgradable software (which is a huge benefit since the cost is so high), 20' 360 radius. 6)PolycomR SoundStationR IP 5000 7) GXP2120 6-line Executive HD IP Phone On Wed, Nov 30, 2011 at 1:47 PM, Faisal Hanif fai...@vopium.com wrote: I have tried EyeBeam and it worked fine with x members audio conference however it need resources (Processing + RAM) per additional line. Regards, Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Wednesday, November 30, 2011 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind Subject: [asterisk-users] Best VoIP conferencing phone ? Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 tel:%2B91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best VoIP conferencing phone ?
The Snom 820 handles 5 and that's the highest I've seen in the snom range. The snom MeetingPoint (dedicated conference phone) only does 4! On Wed, 2011-11-30 at 13:55 +0500, Faisal Hanif wrote: In hardware I used some snom phones up to six lines. You can check on http://www.snom.com/ for appropriate model. Regards, Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Wednesday, November 30, 2011 1:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best VoIP conferencing phone ? Hi Faisal, Thanks for reply but I want hardware wase VoIP device. If know please gussed me. From google I fould the list of below devices but I am not sure that these are best for used or have an issue 1)Polycom SoundStation IP 7000 Why it's best: The Polycom SoundStation IP 7000 is the most advanced conference phone from the Polycom SoundStation lineup and leaves little to be desired. With an amazing 20’ 360 radius, the 7000 is perfect for large conference rooms. The new HD voice quality (22 kHz) allows. 2) Polycom Voicestation 500 Why it's a best pick: The Polycom VoiceStation 500 is one of the best conference phones for a wide variety of reasons. The VoiceStation 500 features amazing call quality, 7’ 360 radius, Bluetooth connectivity, wired connection, background noise reduction, and an attractive design. 3)Panasonic - 8-Microphone Speakerphone with Caller ID KX-TS730S Why it's a best pick: With a 360 10’ radius and 8 microphones, everyone is sure to be heard with the Panasonic KX-TS730S. The multiple microphones allows for everyone sitting in on the conference to be heard uniformly without distortion. 4)Cisco Unified IP Conference Station 7937G Conference VoIP Phone Why it's a best pick: The Cisco 7937G works via VoIP connection, has stunning call clarity, and features a simplistic but expensive design that is easy to use. Cisco is an industry leader in IT communication products, and the 7937G is no different. The 360 design allows everyone to be heard. 5)Polycom SoundStation VTX 1000 Why it's a best pick: The SoundStation VTX 1000 is an incredible conference phone, but it is very pricey and not as good as advertised. The VTX 1000 is designed for large conference rooms and features upgradable software (which is a huge benefit since the cost is so high), 20’ 360 radius. 6)Polycom® SoundStation® IP 5000 7) GXP2120 6-line Executive HD IP Phone On Wed, Nov 30, 2011 at 1:47 PM, Faisal Hanif fai...@vopium.com wrote: I have tried EyeBeam and it worked fine with x members audio conference however it need resources (Processing + RAM) per additional line. Regards, Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Wednesday, November 30, 2011 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind Subject: [asterisk-users] Best VoIP conferencing phone ? Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hwo to stok variable wiith menu
thank you so much for you help,i have flowed your email and installed
thesesadd-ons all
works perfectly i can store the phone_number of the Customer ,now i can do
what i want :)
thanks every one for your support J
2011/11/30 Dale Noll dn...@wi.rr.com
On 11/28/2011 08:24 AM, salaheddine elharit wrote:
thank you for your help
You are welcome.
i would to ask you please, i want to store the phone number of the
customer in the option_name column when he press 3 in context menu
i have created a database aheevacss with user aheevaccs and password
aheevaccs and also i have creatd a table in this database name of table
test with two columns:
option_namevarchar(15)
countint
1-how can i check if the app_mysql module compiled and loaded i use
asterisk 1.4 and if not installed how can ido in order to install and
loaded it
I saw in some other message threads, it looks like you are working out
getting the mysql connectivity working in 1.4. In this version, it is an
'add on' that you have to download separately from the Asterisk source
tree. The instructions given by Warren Selby are correct.
When you do the 'make menuselect', you are presented with a menu with 5
options. Under 'Applications' you need to check app_addon_sql_mysql. Under
'Call Detail Recording' select cdr_addon_mysql. Under 'Resource Modules'
check res_config_mysql. Exit from menuselect and type 'make'. You
probably do not need the res_config_mysql, but it does not hurt anything to
compile it.
Aslo as mentioned in another thread, you do need to have mysql-devel
package installed.
Then run 'make' and 'make install' and 'make samples'. This will build
the modules app_addon_sql_mysql.so, cdr_addon_mysql.so and
res_config_mysql.so and install them in /usr/lib/asterisk/modules. This
does not change any existing modules, just adds the new ones.
Start an Asterisk cli (asterisk -r) and issue the command 'module load
app_addon_sql_mysql'. This should load the module and the MYSQL app will
be available in your dialplan. To verify it is loaded, you can issue the
command 'module show like sql'
You should also check the /etc/asterisk/modules.conf file. There should
be a line that says 'autoload=yes'. If it says no, you will have to add a
line 'load = app_addon_sql_mysql' (do not include the quotes). Note: If
you want to load cdr_addon_mysql, you will have to add a 'load =
cdr_addon_mysql' line as well. This file is read by asterisk at startup,
so after you restart asterisk for the first time after these changes, make
sure the module is loaded with the module show command.
2- can you please veify the menu below and tell me waht is wrong
thanks and regards
[default]
exten = 529,1,Ringing()
exten = 529,2,Wait(4)
exten = 529,3,Goto(accueil,s,1)
[accueil] ; définition d’un contexte pour l’accueil
exten = s,1,SetGlobalVar(sounds_path=/**var/lib/asterisk/sounds/)
exten = s,2,Background(${sounds_path}**welcome)
exten = s,3,goto(accueil,s,1)
exten = #,1,Goto(menu,s,1)
exten = i,1,Playback(${sounds_path}**erreur-saisie)
exten = i,2,goto(accueil,s,1)
exten = t,1,Goto(accueil,s,1)
[menu]
exten = s,1,Background(${sounds_path}**menu)
exten = 0,1,Goto(menu,s,1)
exten = 1,1,Goto(appel,s,1)
exten = 2,1,Goto(message,s,1)
exten = 3,1,NoOp(User chose support option)
exten = 3,n,MYSQL(Connect connid localhost aheevaccs aheevaccs aheevaccs)
exten = 3,n,MYSQL(Query resultid ${connid} update test set count =
count + 1 where option_name = 'support')
exten = 3,n,MYSQL(Clear ${resultid})
exten = 3,n,MYSQL(Disconnect ${connid})
exten = 3,n,Goto(support,s,1)
exten = s,2,goto(menu,s,1)
exten = i,1,Playback(${sounds_path}**erreur-saisie)
exten = i,2,Goto(menu,s,1)
exten = t,1,Goto(menu,s,1)
[appel] ; définition d’un contexte pour le menu d’appel
exten = s,1,Background(${sounds_path}**appel)
exten = s,2,WaitExten(10)
exten = 0,1,Goto(menu,s,1)
exten = 223,1,Dial(SIP/${EXTEN},20,tr)
exten = i,1,Playback(${sounds_path}**erreur-saisie)
exten = i,2,Goto(appel,s,1)
exten = t,1,Goto(appel,s,1)
[message] ; définition d’un contexte pour la messagerie
exten = s,1,VoiceMailMain(${**CALLERIDNUM})
exten = t,1,Hangup()
[support] ; définition d’un contexte pour le support
exten = s,1,GoToIfTime(09:00-17:00|**mon-fri|*|*?s,4)
exten = s,2,Playback(${sounds_path}no-**relation-support)
exten = s,3,Goto(menu,s,1)
exten = s,4,Playback(${sounds_path}**relation-support)
exten = s,5,Queue(default)
exten = t,1,Hangup()
In the [accueil] context, you call Background with the name of the file to
play, then immediately return to the top and play the message again, and
again and again. It will never stop until the caller hangs up. Also, you
are asking the caller to press the '#' key to get past the welcome greeting
before getting to the main menu. I would recommend playing the welcome
followed immediately by the Background() for the menu. The call the
WaitExten() to give the
[asterisk-users] s/n ratio detection etc...
Hi everybody, I' ve been following this list for a while now. Is there a way to detect the individual and cumulative s/n ratio values for the incoming calls in Asterisk or any other Call Center solution?... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR mysql with asterisk 1.4
** THIS IS NOT THE RIGHT PLACE TO POST A REPLY ** On Tuesday 29 November 2011, salaheddine elharit wrote: i use centos 5.5 if i install mysql-devel i can still use the version of mysql installed now in my server because i use it with a database and Im afraid to install this mysql-devel and i lost the version of mysql reunning actually please advice Yes, it will work fine. mysql-devel doesn't *replace* an existing mysql package; it merely adds some development files. The -devel packages are a throwback to the bad old days, when Internet connectivity was by dial-up modem and hard disk space was scarce. Basically, when you build a package from Source Code, there are some files that get left behind and aren't needed just for day-to-day use of the program, but are essential if you need to build something else that works with the package. To trim down the size of downloadable packages, these development files were abstracted away into separate packages. Nine out of ten Linux users have to compile a package from source at some stage and the tenth one is probably lying. Most people wouldn't notice the extra space taken up by -devel packages, and the ones that would -- perhaps because they're building embedded systems, or have other disk space constraints -- are smart enough to know what to do. Yet distributions continue to separate out these packages and in the process, make the whole process of compiling from source unnecessarily complicated. Even an option in the installer to select always install -devel package if it exists would be a good beginning. Distro maintainers -- are you listening? Please? -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on PAP2 linksys showing off-hook
I am using my first PAP2 device from linksys. Used many polycom phones... I configured the PAP2 device with asterisk. I have the registration, thought I was good to go. Plugged in my Valcom 2924 public address analog connection, called the extension and I got busy... very strange I thought. I then looked at the status page of the PAP2 and it says the following Reg online and hook state OFF. How do I get the hookstate to be ON so that I can call into the device? I have power cycled it and it always shows hookstate OFF. Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A new hack?
On Tue, Nov 29, 2011 at 4:44 PM, john Millican j...@millican.us wrote: Maybe I am misunderstanding the gist of the comment OP offered an invalid comparison of how iptables is better than Fail2Ban. Whether or not OP knew that Fail2Ban simply feeds rules to iptables is unclear from his comments. Log scraping is a time honored and effective method to correlate bad behavior. Log scraping can see things that no iptables rule would ever find. Think SSL. If Fail2Ban is a bad log scraper framework, then criticize it with a clear understanding of its role. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best VoIP conferencing phone ?
We've been happy with the polycom IP 7000. Darren Wiebe On Nov 30, 2011 1:40 AM, virendra bhati virbh...@gmail.com wrote: Hi Faisal, Thanks for reply but I want hardware wase VoIP device. If know please gussed me. From google I fould the list of below devices but I am not sure that these are best for used or have an issue *1)Polycom SoundStation IP 7000 * *Why it's best: *The Polycom SoundStation IP 7000 is the most advanced conference phone from the Polycom SoundStation lineup and leaves little to be desired. With an amazing 20’ 360 radius, the 7000 is perfect for large conference rooms. The new HD voice quality (22 kHz) allows. * * *2) Polycom Voicestation 500* * * *Why it's a best pick: *The Polycom VoiceStation 500 is one of the best conference phones for a wide variety of reasons. The VoiceStation 500 features amazing call quality, 7’ 360 radius, Bluetooth connectivity, wired connection, background noise reduction, and an attractive design. * * *3)Panasonic - 8-Microphone Speakerphone with Caller ID KX-TS730S* * * *Why it's a best pick: *With a 360 10’ radius and 8 microphones, everyone is sure to be heard with the Panasonic KX-TS730S. The multiple microphones allows for everyone sitting in on the conference to be heard uniformly without distortion. * * *4)Cisco Unified IP Conference Station 7937G Conference VoIP Phone* * * *Why it's a best pick: *The Cisco 7937G works via VoIP connection, has stunning call clarity, and features a simplistic but expensive design that is easy to use. Cisco is an industry leader in IT communication products, and the 7937G is no different. The 360 design allows everyone to be heard. * * *5)Polycom SoundStation VTX 1000* * * *Why it's a best pick: *The SoundStation VTX 1000 is an incredible conference phone, but it is very pricey and not as good as advertised. The VTX 1000 is designed for large conference rooms and features upgradable software (which is a huge benefit since the cost is so high), 20’ 360 radius. 6)Polycom® SoundStation® IP 5000* 7) GXP2120 6-line Executive HD IP Phone* On Wed, Nov 30, 2011 at 1:47 PM, Faisal Hanif fai...@vopium.com wrote: I have tried EyeBeam and it worked fine with x members audio conference however it need resources (Processing + RAM) per additional line. ** ** Regards, ** ** Faisal Hanif ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati *Sent:* Wednesday, November 30, 2011 11:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind *Subject:* [asterisk-users] Best VoIP conferencing phone ? ** ** Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A new hack?
On 11/30/2011 09:01 AM, Tom Browning wrote: I agree - its a bad comparison of 2 different things meant for different purposes. iptables is enforcement, fail2ban is detection. if you have time to sit and make up iptables rules by hand during every hack attempt 1) you have too much time on your hands 2) you have too much time on your hands On Tue, Nov 29, 2011 at 4:44 PM, john Millicanj...@millican.us wrote: Maybe I am misunderstanding the gist of the comment OP offered an invalid comparison of how iptables is better than Fail2Ban. Whether or not OP knew that Fail2Ban simply feeds rules to iptables is unclear from his comments. Log scraping is a time honored and effective method to correlate bad behavior. Log scraping can see things that no iptables rule would ever find. Think SSL. If Fail2Ban is a bad log scraper framework, then criticize it with a clear understanding of its role. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] s/n ratio detection etc...
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yasin SULUHAN Sent: Wednesday, November 30, 2011 6:25 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] s/n ratio detection etc... Hi everybody, I' ve been following this list for a while now. Is there a way to detect the individual and cumulative s/n ratio values for the incoming calls in Asterisk or any other Call Center solution?... Either I need to finish my coffee or this should be worded better: Is there a way the detect the individual and cumulative signal-to-noise ratio values for incoming calls to Asterisk (or any other Call Center solution)? This depends on 1. How are the calls delivered to Asterisk (we will ignore the other call center since this is an Asterisk discussion board)? SIP/DAHDI(PSTN/PRI/E1/ETC)? 2. What version of Asterisk? 3. Do you want built-in methods or could other methods such as daemons be used? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] s/n ratio detection etc...
On Wed, Nov 30, 2011 at 4:27 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Yasin SULUHAN *Sent:* Wednesday, November 30, 2011 6:25 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] s/n ratio detection etc... ** ** Hi everybody, I' ve been following this list for a while now. Is there a way to detect the individual and cumulative s/n ratio values for the incoming calls in Asterisk or any other Call Center solution?...** ** ** ** Either I need to finish my coffee or this should be worded better: Sorry about this. This request just came in from a client and we need an answer very quickly. Is there a way the detect the individual and cumulative signal-to-noise ratio values for incoming calls to Asterisk (or any other Call Center solution)? ** ** This depends on **1. **How are the calls delivered to Asterisk (we will ignore the “other call center” since this is an Asterisk discussion board)? SIP/DAHDI(PSTN/PRI/E1/ETC)? DAHDI **2. **What version of Asterisk? 1.8.7 3. Do you want “built-in” methods or could other methods such as daemons be used? either way would be ok. Thank you for your quick response -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] s/n ratio detection etc...
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yasin SULUHAN Sent: Wednesday, November 30, 2011 8:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] s/n ratio detection etc... On Wed, Nov 30, 2011 at 4:27 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yasin SULUHAN Sent: Wednesday, November 30, 2011 6:25 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] s/n ratio detection etc... Hi everybody, I' ve been following this list for a while now. Is there a way to detect the individual and cumulative s/n ratio values for the incoming calls in Asterisk or any other Call Center solution?... Either I need to finish my coffee or this should be worded better: Sorry about this. This request just came in from a client and we need an answer very quickly. Is there a way the detect the individual and cumulative signal-to-noise ratio values for incoming calls to Asterisk (or any other Call Center solution)? This depends on 1. How are the calls delivered to Asterisk (we will ignore the other call center since this is an Asterisk discussion board)? SIP/DAHDI(PSTN/PRI/E1/ETC)? DAHDI 2. What version of Asterisk? 1.8.7 3. Do you want built-in methods or could other methods such as daemons be used? either way would be ok. Your best bet as I understand it would be to use dahdi_tools to monitor your lines or to use mixmonitor to record the calls so you can review and tune problems as needed. Either of these options would cost you some overhead in processor usage and disk space. Thank you for your quick response -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with Polycom SPIP650 and its sidecar
Hello, On one location, I've got from time to time (let say one a week) the following issue : the phone SoundPoint 650 works ok (can call or answer, display and sound are ok), the sidecar looses its display : entries on sidecar's LCD screen are not displayed anymore, or names are truncated, or BLF are not shown or updated. I only have one SPIP650 on this system so I can't compare with others. What could be the root cause of this ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Wednesday, November 30, 2011 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Issue with Polycom SPIP650 and its sidecar Hello, On one location, I've got from time to time (let say one a week) the following issue : the phone SoundPoint 650 works ok (can call or answer, display and sound are ok), the sidecar looses its display : entries on sidecar's LCD screen are not displayed anymore, or names are truncated, or BLF are not shown or updated. I only have one SPIP650 on this system so I can't compare with others. What could be the root cause of this ? Regards I doubt it is an Asterisk related issue, but what version of Asterisk is the phone running on? The most likely probabilities as I see them are (1) network or power issue (2) hardware problem with the phone (3) provisioning issue with the phone (bad sip.ld or such). You could investigate /var/log/asterisk/full to see what kind of call activity occurs around these failures. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] vall directly extensions from E1-PRI line
Hi all, As a new on asterisk i have some silly questions. I'll try to connect an asterisk PBX between Telephone provider and an AVAYA Definity PBX. I've already install elastix-2.2.0 i386 version on a PC with a DE210 ISDN PRI card. In previous status we can dial from external directly to 4digits extensions (i.e. the call NX7721 ring the extension 7721). At the moment on IP phones i can call only 3digits (i.e. the call NX7721 ring the extension 721 and never the extension 7721). Has anyone any idea how can i fix it. Thanks in advantage D. Sidirokastritis NOC HCMR-Crete tel. 2810-337709 This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar
When the side car looses it entries, what does the config file show for the entries. This happened to me one time but that was only because for some reason, the contacts file was deleted by accident and I had to recreate it. ( I have a backup now too! ) It probably as Dan said, check the firmware and get the latest. Make sure your sidecar is pushed down all the way. Possible get a new IP650 and see if it exhibits the same behavior. I am pretty positive its not asterisk related. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, November 30, 2011 10:32 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Wednesday, November 30, 2011 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Issue with Polycom SPIP650 and its sidecar Hello, On one location, I've got from time to time (let say one a week) the following issue : the phone SoundPoint 650 works ok (can call or answer, display and sound are ok), the sidecar looses its display : entries on sidecar's LCD screen are not displayed anymore, or names are truncated, or BLF are not shown or updated. I only have one SPIP650 on this system so I can't compare with others. What could be the root cause of this ? Regards I doubt it is an Asterisk related issue, but what version of Asterisk is the phone running on? The most likely probabilities as I see them are (1) network or power issue (2) hardware problem with the phone (3) provisioning issue with the phone (bad sip.ld or such). You could investigate /var/log/asterisk/full to see what kind of call activity occurs around these failures. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar
2011/11/30 Danny Nicholas da...@debsinc.com *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Wednesday, November 30, 2011 9:27 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Issue with Polycom SPIP650 and its sidecar ** ** Hello, On one location, I've got from time to time (let say one a week) the following issue : the phone SoundPoint 650 works ok (can call or answer, display and sound are ok), the sidecar looses its display : entries on sidecar's LCD screen are not displayed anymore, or names are truncated, or BLF are not shown or updated. I only have one SPIP650 on this system so I can't compare with others. What could be the root cause of this ? Regards ** ** I doubt it is an Asterisk related issue, but what version of Asterisk is the phone running on? 1.6.1.18 The most likely probabilities as I see them are (1) network or power issue (2) hardware problem with the phone (3) provisioning issue with the phone (bad sip.ld or such). You could investigate /var/log/asterisk/full to see what kind of call activity occurs around these failures. I agree with your list of probabilities. The trouble is at the moment, I need phone user help to detect when sidecar display is stopping or restarting : if I could read by myself somewhere something meaning the sidecar is doing wrong, that would be perfect :-( If someone could testify about a successfull (or unsuccessful) experience with the same hardware, that would also help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar
Hi Olivier, It if occurs only on the sidecar, I would imagine this is either a defective sidecar/Polycom phone, or a defective PoE switch not giving enough power. Changing PoE port would eliminate of confirm the PoE port being the issue, but I'm betting on a Polycom defect. Make sure the PoE port is configured (if it`s a smart switch) to send maximum power to the port, with a sidecar I think the phone requires 12W. Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Wednesday, November 30, 2011 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Issue with Polycom SPIP650 and its sidecar Hello, On one location, I've got from time to time (let say one a week) the following issue : the phone SoundPoint 650 works ok (can call or answer, display and sound are ok), the sidecar looses its display : entries on sidecar's LCD screen are not displayed anymore, or names are truncated, or BLF are not shown or updated. I only have one SPIP650 on this system so I can't compare with others. What could be the root cause of this ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hwo to stok variable wiith menu
i have last question regarding this thread
with exten = 3,n,MYSQL(Query resultid ${connid} insert into test (
option_name ) values ('${CALLERID(num)}'))
i can store the phone number without issue
i need also the date and hour fo call in the count coulum
could you please give me the syntex
best regards
2011/11/30 salaheddine elharit salah.elharit...@gmail.com
thank you so much for you help,i have flowed your email and installed
theses add-ons all works perfectly i can store the phone_number of the
Customer ,now i can do what i want :)
thanks every one for your support J
2011/11/30 Dale Noll dn...@wi.rr.com
On 11/28/2011 08:24 AM, salaheddine elharit wrote:
thank you for your help
You are welcome.
i would to ask you please, i want to store the phone number of the
customer in the option_name column when he press 3 in context menu
i have created a database aheevacss with user aheevaccs and password
aheevaccs and also i have creatd a table in this database name of table
test with two columns:
option_namevarchar(15)
countint
1-how can i check if the app_mysql module compiled and loaded i use
asterisk 1.4 and if not installed how can ido in order to install and
loaded it
I saw in some other message threads, it looks like you are working out
getting the mysql connectivity working in 1.4. In this version, it is an
'add on' that you have to download separately from the Asterisk source
tree. The instructions given by Warren Selby are correct.
When you do the 'make menuselect', you are presented with a menu with 5
options. Under 'Applications' you need to check app_addon_sql_mysql. Under
'Call Detail Recording' select cdr_addon_mysql. Under 'Resource Modules'
check res_config_mysql. Exit from menuselect and type 'make'. You
probably do not need the res_config_mysql, but it does not hurt anything to
compile it.
Aslo as mentioned in another thread, you do need to have mysql-devel
package installed.
Then run 'make' and 'make install' and 'make samples'. This will build
the modules app_addon_sql_mysql.so, cdr_addon_mysql.so and
res_config_mysql.so and install them in /usr/lib/asterisk/modules. This
does not change any existing modules, just adds the new ones.
Start an Asterisk cli (asterisk -r) and issue the command 'module load
app_addon_sql_mysql'. This should load the module and the MYSQL app will
be available in your dialplan. To verify it is loaded, you can issue the
command 'module show like sql'
You should also check the /etc/asterisk/modules.conf file. There should
be a line that says 'autoload=yes'. If it says no, you will have to add a
line 'load = app_addon_sql_mysql' (do not include the quotes). Note: If
you want to load cdr_addon_mysql, you will have to add a 'load =
cdr_addon_mysql' line as well. This file is read by asterisk at startup,
so after you restart asterisk for the first time after these changes, make
sure the module is loaded with the module show command.
2- can you please veify the menu below and tell me waht is wrong
thanks and regards
[default]
exten = 529,1,Ringing()
exten = 529,2,Wait(4)
exten = 529,3,Goto(accueil,s,1)
[accueil] ; définition d’un contexte pour l’accueil
exten = s,1,SetGlobalVar(sounds_path=/**var/lib/asterisk/sounds/)
exten = s,2,Background(${sounds_path}**welcome)
exten = s,3,goto(accueil,s,1)
exten = #,1,Goto(menu,s,1)
exten = i,1,Playback(${sounds_path}**erreur-saisie)
exten = i,2,goto(accueil,s,1)
exten = t,1,Goto(accueil,s,1)
[menu]
exten = s,1,Background(${sounds_path}**menu)
exten = 0,1,Goto(menu,s,1)
exten = 1,1,Goto(appel,s,1)
exten = 2,1,Goto(message,s,1)
exten = 3,1,NoOp(User chose support option)
exten = 3,n,MYSQL(Connect connid localhost aheevaccs aheevaccs
aheevaccs)
exten = 3,n,MYSQL(Query resultid ${connid} update test set count =
count + 1 where option_name = 'support')
exten = 3,n,MYSQL(Clear ${resultid})
exten = 3,n,MYSQL(Disconnect ${connid})
exten = 3,n,Goto(support,s,1)
exten = s,2,goto(menu,s,1)
exten = i,1,Playback(${sounds_path}**erreur-saisie)
exten = i,2,Goto(menu,s,1)
exten = t,1,Goto(menu,s,1)
[appel] ; définition d’un contexte pour le menu d’appel
exten = s,1,Background(${sounds_path}**appel)
exten = s,2,WaitExten(10)
exten = 0,1,Goto(menu,s,1)
exten = 223,1,Dial(SIP/${EXTEN},20,tr)
exten = i,1,Playback(${sounds_path}**erreur-saisie)
exten = i,2,Goto(appel,s,1)
exten = t,1,Goto(appel,s,1)
[message] ; définition d’un contexte pour la messagerie
exten = s,1,VoiceMailMain(${**CALLERIDNUM})
exten = t,1,Hangup()
[support] ; définition d’un contexte pour le support
exten = s,1,GoToIfTime(09:00-17:00|**mon-fri|*|*?s,4)
exten = s,2,Playback(${sounds_path}no-**relation-support)
exten = s,3,Goto(menu,s,1)
exten = s,4,Playback(${sounds_path}**relation-support)
exten = s,5,Queue(default)
exten = t,1,Hangup()
In the [accueil] context, you call Background with the name of the file
to play, then
[asterisk-users] # of Polycoms on a DSL line?
Out of curiosity, how many concurrent phone calls for an office that uses Polycoms could be sustained on a DSL ( 3meg down, 768 up ) line using g711? Not sure if its 64kbps or 87kbps. I would say roughly 8 but I don't know if the polycoms add any more payload to the network for presence and all that jazz. Thanks, --E -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on PAP2 linksys showing off-hook
Il 30/11/2011 14.38, Jerry Geis ha scritto: I configured the PAP2 device with asterisk. I have the registration, thought I was good to go. Plugged in my Valcom 2924 public address analog connection, called the extension and I got busy... very strange I thought Not so strange ;-) According to the Valcom 2924 manual it is designed to connect to an analog phone or a pbx analog c.o. port. So you're connecting the PAP2 FXS port to another pbx port... the one on Valcom, not only can't work but may damage one or both devices. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar
2011/11/30 eherr email.eherr9...@gmail.com When the side car looses it entries, what does the config file show for the entries. ** ** This happened to me one time but that was only because for some reason, the contacts file was deleted by accident and I had to recreate it. ( I have a backup now too! ) I didn't investigate much in this direction, as in my case, sometimes I have several days of operation without any issue. So to match the problem you met, the phone should get corrupted config for several hours, then correct config for several days, then corrupted config again. ** ** It probably as Dan said, check the firmware and get the latest. You're right, I should do that. Make sure your sidecar is pushed down all the way. The way a Polycom sidecar is plugged into a phone seems fragile to me. Possible get a new IP650 and see if it exhibits the same behavior. I already changed the sidecar once ... ** ** I am pretty positive its not asterisk related. I agree ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 30, 2011 10:32 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Wednesday, November 30, 2011 9:27 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Issue with Polycom SPIP650 and its sidecar ** ** Hello, On one location, I've got from time to time (let say one a week) the following issue : the phone SoundPoint 650 works ok (can call or answer, display and sound are ok), the sidecar looses its display : entries on sidecar's LCD screen are not displayed anymore, or names are truncated, or BLF are not shown or updated. I only have one SPIP650 on this system so I can't compare with others. What could be the root cause of this ? Regards ** ** I doubt it is an Asterisk related issue, but what version of Asterisk is the phone running on? The most likely probabilities as I see them are (1) network or power issue (2) hardware problem with the phone (3) provisioning issue with the phone (bad sip.ld or such). You could investigate /var/log/asterisk/full to see what kind of call activity occurs around these failures. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar
2011/11/30 Mike l...@net-wall.com Hi Olivier, ** ** It if occurs only on the sidecar, I would imagine this is either a defective sidecar/Polycom phone, or a defective PoE switch not giving enough power. Changing PoE port would eliminate of confirm the PoE port being the issue, but I’m betting on a Polycom defect. ** ** Make sure the PoE port is configured (if it`s a smart switch) to send maximum power to the port, with a sidecar I think the phone requires 12W. This info is very interesting. I wouldn't be too surprised that a PoE switch not supplying its theorical 15W output on a long period. I'll try to use work around this possible cause by not using PoE. In any case, I'll report my findings here. ** ** Regards, ** ** Mike ** ** ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Wednesday, November 30, 2011 10:27 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Issue with Polycom SPIP650 and its sidecar ** ** Hello, On one location, I've got from time to time (let say one a week) the following issue : the phone SoundPoint 650 works ok (can call or answer, display and sound are ok), the sidecar looses its display : entries on sidecar's LCD screen are not displayed anymore, or names are truncated, or BLF are not shown or updated. I only have one SPIP650 on this system so I can't compare with others. What could be the root cause of this ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar
Maybe use a power supply instead of PoE, see if problem still occurs. Marco. Op 30 nov. 2011 18:46 schreef Olivier oza_4...@yahoo.fr het volgende: 2011/11/30 Mike l...@net-wall.com Hi Olivier, ** ** It if occurs only on the sidecar, I would imagine this is either a defective sidecar/Polycom phone, or a defective PoE switch not giving enough power. Changing PoE port would eliminate of confirm the PoE port being the issue, but I’m betting on a Polycom defect. ** ** Make sure the PoE port is configured (if it`s a smart switch) to send maximum power to the port, with a sidecar I think the phone requires 12W. This info is very interesting. I wouldn't be too surprised that a PoE switch not supplying its theorical 15W output on a long period. I'll try to use work around this possible cause by not using PoE. In any case, I'll report my findings here. ** ** Regards, ** ** Mike ** ** ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Wednesday, November 30, 2011 10:27 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Issue with Polycom SPIP650 and its sidecar*** * ** ** Hello, On one location, I've got from time to time (let say one a week) the following issue : the phone SoundPoint 650 works ok (can call or answer, display and sound are ok), the sidecar looses its display : entries on sidecar's LCD screen are not displayed anymore, or names are truncated, or BLF are not shown or updated. I only have one SPIP650 on this system so I can't compare with others. What could be the root cause of this ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sound files with MixMonitor not playable with Media Player
Hello, the wav sound files that are created by using MixMonitor()-command are not playable with Windows Media Player. I can play them with vlc-player and on my Fedora with Totem. This is one of the files : /var/ftp/104/2011-11-30_11:54:39_89000404.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz What would be missing on the system (Centos 5.7) that makes wav-files difficult for Windows Media Player ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound files with MixMonitor not playable with Media Player
Since the other data seems kosher, have you tried just renaming the file without the -, _ and : ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, November 30, 2011 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Sound files with MixMonitor not playable with Media Player Hello, the wav sound files that are created by using MixMonitor()-command are not playable with Windows Media Player. I can play them with vlc-player and on my Fedora with Totem. This is one of the files : /var/ftp/104/2011-11-30_11:54:39_89000404.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz What would be missing on the system (Centos 5.7) that makes wav-files difficult for Windows Media Player ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound files with MixMonitor not playable with Media Player
Hello, it worked perfectly before... I just did a clean install of my Asterisk server and changed nothing but Centos 5.6 to CentOS 5.7 Therefore I ask if it should be something that I'm missing on my system ? Jonas. On 11/30/2011 08:59 PM, Danny Nicholas wrote: Since the other data seems kosher, have you tried just renaming the file without the -, _ and : ? *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, November 30, 2011 1:55 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Sound files with MixMonitor not playable with Media Player Hello, the wav sound files that are created by using MixMonitor()-command are not playable with Windows Media Player. I can play them with vlc-player and on my Fedora with Totem. This is one of the files : /var/ftp/104/2011-11-30_11:54:39_89000404.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz What would be missing on the system (Centos 5.7) that makes wav-files difficult for Windows Media Player ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound files with MixMonitor not playable with Media Player
Check this link - you might be recording a muted file http://www.centos.org/modules/newbb/print.php?form=1 http://www.centos.org/modules/newbb/print.php?form=1topic_id=34058forum=3 7order=ASCstart=0 topic_id=34058forum=37order=ASCstart=0 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, November 30, 2011 2:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sound files with MixMonitor not playable with Media Player Hello, it worked perfectly before... I just did a clean install of my Asterisk server and changed nothing but Centos 5.6 to CentOS 5.7 Therefore I ask if it should be something that I'm missing on my system ? Jonas. On 11/30/2011 08:59 PM, Danny Nicholas wrote: Since the other data seems kosher, have you tried just renaming the file without the -, _ and : ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, November 30, 2011 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Sound files with MixMonitor not playable with Media Player Hello, the wav sound files that are created by using MixMonitor()-command are not playable with Windows Media Player. I can play them with vlc-player and on my Fedora with Totem. This is one of the files : /var/ftp/104/2011-11-30_11:54:39_89000404.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz What would be missing on the system (Centos 5.7) that makes wav-files difficult for Windows Media Player ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to find out one way latency
Hi All, How can I find out One way latency from my PBX to my SIP Trunk Provider. My SIP provider recommends a One way latency of 100ms for good Voice quality. Ping request to their IP Address gives me a response in approx. 260ms. Will that be good enough for a SIP Trunk. Please help. We are trying to sign up with a new SIP Provider. Thanks, Najim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
Am 30.11.2011 21:47, schrieb NaJIm: Hi All, How can I find out One way latency from my PBX to my SIP Trunk Provider. My SIP provider recommends a One way latency of 100ms for good Voice quality. Ping request to their IP Address gives me a response in approx. 260ms. Will that be good enough for a SIP Trunk. Please help. We are trying to sign up with a new SIP Provider. Thanks, Najim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Najim, a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. regards, Ruben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Walkie talkie to sip phone interface
Hi All, I've been trying to find a solution that would allow our sip phones to communication with walkie talkies. Our setup is that we have sip phones setup in 2 locations, headquarters and dome. We can communication from headquarters and dome through sip phones, but within the dome we have technicians that use walkie talkies to communicate as they go about their work. Our hope is to allow 2 way communications from our sip phones at headquarters (or within the dome) with our technicians using their walkie talkies as they are working throughout the dome. Not sure if this is possible but I would appreciate any suggestions. Thanks, Ferdinand -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Walkie talkie to sip phone interface
On Wed, Nov 30, 2011 at 6:20 PM, Ferdinand Babas ba...@cfht.hawaii.edu wrote: Hi All, I've been trying to find a solution that would allow our sip phones to communication with walkie talkies. Our setup is that we have sip phones setup in 2 locations, headquarters and dome. We can communication from headquarters and dome through sip phones, but within the dome we have technicians that use walkie talkies to communicate as they go about their work. Our hope is to allow 2 way communications from our sip phones at headquarters (or within the dome) with our technicians using their walkie talkies as they are working throughout the dome. Not sure if this is possible but I would appreciate any suggestions. Thanks, Ferdinand Yes there are interfaces for POTS, SIP, H323 and others. People could help if they knew what type of signaling or vendor you are using. A google search gave me this. http://www.motorola.com/web/Business/Products/Two-way%20Radio%20Infrastructure/Gateways/MOTOBRIDGE%20Interoperable%20IP%20Solution/_Documents/MotoBridgeSS_Final.pdf -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
Am 30.11.2011 21:47, schrieb NaJIm: Ping request to their IP Address gives me a response in approx. 260ms. Will that be good enough for a SIP Trunk. On Wed, 30 Nov 2011, Ruben Rögels wrote: a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. 'Ping time' is not an accurate predictor of SIP quality. A 'ping' is an ICMP Echo/reply packet and some routers consider them less important than 'data' packets and service them on an 'as resources permit' basis. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
Thank you Ruben. Is there anything else that I should be concerned about when looking for a SIP provider. ?? Regards, Najim. On Thu, Dec 1, 2011 at 2:34 AM, Ruben Rögels ruben.roeg...@jumping-frog.org wrote: Am 30.11.2011 21:47, schrieb NaJIm: Hi All, How can I find out One way latency from my PBX to my SIP Trunk Provider. My SIP provider recommends a One way latency of 100ms for good Voice quality. Ping request to their IP Address gives me a response in approx. 260ms. Will that be good enough for a SIP Trunk. Please help. We are trying to sign up with a new SIP Provider. Thanks, Najim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Najim, a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. regards, Ruben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing asterisk on a server vs appliance(e.g digium mypbx)
Hi, I am looking into advising a client on the pro's and cons of using Installing asterisk on a server vs appliance(e.g digium mypbx). the appliance seems cheaper initially. From experience, what would be pro and cons for either option? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
Does that mean I can expect lesser delays with my Voice packets ?? That would be even better. Regards, Najim On Thu, Dec 1, 2011 at 3:42 AM, Steve Edwards asterisk@sedwards.comwrote: Am 30.11.2011 21:47, schrieb NaJIm: Ping request to their IP Address gives me a response in approx. 260ms. Will that be good enough for a SIP Trunk. On Wed, 30 Nov 2011, Ruben Rögels wrote: a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. 'Ping time' is not an accurate predictor of SIP quality. A 'ping' is an ICMP Echo/reply packet and some routers consider them less important than 'data' packets and service them on an 'as resources permit' basis. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
Is there anything else that I should be concerned about, when looking to signup for a SIP provider. ?? Regards, Najim On Thu, Dec 1, 2011 at 4:49 AM, NaJIm getna...@gmail.com wrote: Does that mean I can expect lesser delays with my Voice packets ?? That would be even better. Regards, Najim On Thu, Dec 1, 2011 at 3:42 AM, Steve Edwards asterisk@sedwards.comwrote: Am 30.11.2011 21:47, schrieb NaJIm: Ping request to their IP Address gives me a response in approx. 260ms. Will that be good enough for a SIP Trunk. On Wed, 30 Nov 2011, Ruben Rögels wrote: a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. 'Ping time' is not an accurate predictor of SIP quality. A 'ping' is an ICMP Echo/reply packet and some routers consider them less important than 'data' packets and service them on an 'as resources permit' basis. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
On Thu, 2011-12-01 at 04:52 +0530, NaJIm wrote: Is there anything else that I should be concerned about, when looking to signup for a SIP provider. ?? Latency is important, but packet loss also, likewise packet re-ordering. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
My ping requests show 0% packet loss. How do we find out packet re-ordering.?? Najim. On Thu, Dec 1, 2011 at 5:18 AM, Hans Witvliet aster...@a-domani.nl wrote: On Thu, 2011-12-01 at 04:52 +0530, NaJIm wrote: Is there anything else that I should be concerned about, when looking to signup for a SIP provider. ?? Latency is important, but packet loss also, likewise packet re-ordering. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing asterisk on a server vs appliance(e.g digium mypbx)
At the most basic level, typically an appliance will have a GUI and be geared towards non-tech installation. Loading bare Asterisk on a server is very different. Do you want a GUI or bare Asterisk? BTW, the MyPBX product is not a Digium product, it's from an oriental company named Yeastar. My experience in talking to them about their phones has been so-so. Historically we've had awful experiences with other Chinese phone vendors and have stopped considering products from Chinese companies. We did not actually try Yeastar products. On Wed, Nov 30, 2011 at 3:39 PM, James Mutuku listmut...@gmail.com wrote: Hi, I am looking into advising a client on the pro's and cons of using Installing asterisk on a server vs appliance(e.g digium mypbx). the appliance seems cheaper initially. From experience, what would be pro and cons for either option? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI script that uses google's text to speech engine
Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first public release ca be obtained here: https://github.com/downloads/zaf/asterisk-googletts/asterisk-googletts-0.2.tar.gz To get a sample of the speech synthesis quality try this link: http://translate.google.com/translate_tts?tl=enq=this+is+a+test+for+google+text+to+speech+engine The code is still very young so suggestions, comments and bug reports are more than welcome. -- Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. I am probably splitting hairs, but that's not always true because there's no guarantee that the reply traveled the same path as the echo request. If you dig into BGP issues you'll see sometimes that traffic one direction takes a different route than traffic the other direction. I don't know of any simple and accurate way to learn the one way latency so I'm surprised they specified anything other than round trip time. 'Ping time' is not an accurate predictor of SIP quality. A 'ping' is an ICMP Echo/reply packet and some routers consider them less important than 'data' packets and service them on an 'as resources permit' basis. That's possibly maybe true if someone's router or connection is overloaded and they are trying to make up for it with CoS policies while they save up for an upgrade. Otherwise it's an apology for a crappy network. That's the brutally honest truth. You can make a pretty good prediction with ping. sudo ping -f -i .02 -s 180 -Q 0xb8 [ip] gives a tolerable simulation of voip traffic. let it run for awhile, then press ctrl+c and see how many packets were dropped and also check the mdev number. If mdev is low and packet loss is almost nothing then you can expect decent voice quality. It may not be a 100% perfect test, but I'll bet you a vast majority of the time I can do that test and tell you whether it's going to suck. latency by itself with low jitter and no packet loss just means delay. It's a matter of opinion and circumstance how tolerable delay is, but I think your 230ms ping is at the upper edge of what most people can live with. Much more than that and you'll be tempted to say 'over' at the end of sentence. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Walkie talkie to sip phone interfacere:
I've been trying to find a solution that would allow our sip phones to communication with walkie talkies. Our setup is that we have sip phones setup in 2 locations, headquarters and dome. We can communication from headquarters and dome through sip phones, but within the dome we have technicians that use walkie talkies to communicate as they go about their work. Our hope is to allow 2 way communications from our sip phones at headquarters (or within the dome) with our technicians using their walkie talkies as they are working throughout the dome. Not sure if this is possible but I would appreciate any suggestions. It's definitely possible. How practical it is, remains to be seen and is probably a how well do the details work out? issue. The simplest approach is probably this: you'll need a walkie talkie base station which will serve as the transmit/receive point for the dome. The most straightforward would be to use one of the actual walkie talkie radios, with a well-filtered DC power supply in place of a battery pack. You would need to hook up the W/T's speaker out and mic in, and perhaps the push to talk line, to suitable audio and control I/Os on some sort of Asterisk end-point. The least expensive way would probably be to use the Asterisk server itself (or some PC running a soft-phone client), and use the PC/server's sound card jacks. You would need some sort of level-adjusting (padding) system for the signal being fed into the walkie talkie's mic input (these generally require much less voltage than a sound card's line-level output), and it would probably be a good idea to have an audio isolation transformer in each audio path to prevent ground loops and hum and RF pickup. You'd need some way of keying the radio's push-to-talk when someone on the phone starts to speak, and then release PTT when the voice stops. Some walkie talkies have a VOX (voice-operated switch) which will do the job. Others do not, and you would need a separate VOX circuit (not difficult). One possible hardware device which might save you trouble is the Tigertronics SignaLink USB - it's primarily designed for and sold to amateur-radio operators but has multiple uses. It consists of a USB sound card, isolation transformers, an adjustable VOX/PTT circuit, and a very flexible radio-interconnect-cable system which you could adapt to the speaker/mic needs of your walkie talkie. You'd simply plug it into the server's USB jack, and it would become a secondary audio interface. On the Asterisk side, you'd want to use the ALSA channel driver, and create an inbound extension which would simply dial the ALSA channel. Or, you might decide to use one of the various Asterisk bridging/conference applications, and have the ALSA walkie-talkie channel be perpetually signed into a conference bridge which one or more other users could phone into. You'll need to select and implement a suitable security policy, to control who can access your dome audio link, and decide whether someone in the dome can use a walkie talkie with DTMF capability to dial calls or otherwise control the Asterisk system via radio. Finally, you need to make sure that all of this is legal in your particular jurisdiction. Here in the U.S., there are quite a few personal and mobile-radio services for which it is *not* legal to create a connection to the wired telephony system. This is probably something you should determine first, rather than at the end. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hwo to stok variable wiith menu
On 11/30/2011 11:13 AM, salaheddine elharit wrote:
i have last question regarding this thread
with exten = 3,n,MYSQL(Query resultid ${connid} insert into test (
option_name ) values ('${CALLERID(num)}'))
i can store the phone number without issue
i need also the date and hour fo call in the count coulum
could you please give me the syntex
best regards
The example table that I gave originally was before I knew what you were
looking to do. I assumed, incorrectly that you simply wanted to track
how many times an option was selected in the menu.
I would recommend that you create a table specifically for this
application.
That table may look like this. Please name the table and columns
appropriately for your application.
create table option_three (
calldatedatetime,
calleridvarchar(40)
)
Then the sql would look something like this...
exten = 3,n,MYSQL(Query resultid ${connid} insert into option_three
( calldate, callerid ) values ( now(), '${CALLERID(num)}'))
Dale
--
The truth speaks for itself. I'm just the messenger.
Lyta Alexander - Babylon 5
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
WOW.. That is the most complicated Ping I have ever seen.. :) This is the result I got. # ping -f -i .02 -s 180 -Q 0xb8 xx.xx.xx.xx *PING xx.xx.xx.xx (xx.xx.xx.xx) 180(208) bytes of data. . --- xx.xx.xx.xx ping statistics --- 15338 packets transmitted, 15325 received, 0% packet loss, time 352748ms rtt min/avg/max/mdev = 276.499/286.185/310.118/7.248 ms, pipe 15, ipg/ewma 22.999/284.882 ms * The same test with my Present SIP Provider gave me the result below. *10926 packets transmitted, 10913 received, 0% packet loss, time 244048ms rtt min/avg/max/mdev = 289.514/292.668/316.350/2.336 ms, pipe 15, ipg/ewma 22.338/292.941 ms * I suppose the value of mdev is much higher in the first case but 0% packet loss in both the cases. Does this mean that the voice quality is going to be real bad?? Thanks, Najim On Thu, Dec 1, 2011 at 6:33 AM, Adam Moffett adamli...@plexicomm.netwrote: a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. I am probably splitting hairs, but that's not always true because there's no guarantee that the reply traveled the same path as the echo request. If you dig into BGP issues you'll see sometimes that traffic one direction takes a different route than traffic the other direction. I don't know of any simple and accurate way to learn the one way latency so I'm surprised they specified anything other than round trip time. 'Ping time' is not an accurate predictor of SIP quality. A 'ping' is an ICMP Echo/reply packet and some routers consider them less important than 'data' packets and service them on an 'as resources permit' basis. That's possibly maybe true if someone's router or connection is overloaded and they are trying to make up for it with CoS policies while they save up for an upgrade. Otherwise it's an apology for a crappy network. That's the brutally honest truth. You can make a pretty good prediction with ping. sudo ping -f -i .02 -s 180 -Q 0xb8 [ip] gives a tolerable simulation of voip traffic. let it run for awhile, then press ctrl+c and see how many packets were dropped and also check the mdev number. If mdev is low and packet loss is almost nothing then you can expect decent voice quality. It may not be a 100% perfect test, but I'll bet you a vast majority of the time I can do that test and tell you whether it's going to suck. latency by itself with low jitter and no packet loss just means delay. It's a matter of opinion and circumstance how tolerable delay is, but I think your 230ms ping is at the upper edge of what most people can live with. Much more than that and you'll be tempted to say 'over' at the end of sentence. -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
I would bet you get about the same result with the two providers.all else being equal. mdev (mean deviation) is a simple way to measure jitter, and you have to put in context with the min/avg/max numbers. If I had 7ms of deviation and average times of 4ms, that would be an issue because you would be likely to get packets out of order. But 7ms compared to 286ms probably means nothing. Your biggest problem with both providers is delay, but if you can tolerate the delay you have now, then you can probably tolerate the delay with the other provider. Also note that although packet loss is 0%, some packets are still dropped in both cases. One dropped packet means a small amount of audio is lost (depends on codec, but often 20ms). If those handful of dropped packets are scattered evenly then you wouldn't notice it, but it's common for them to occur in a cluster. If the 13 packets dropped in the first example all happened at once you would have lost 260ms of audioand you would certainly hear that. You may be able to tell by watching the periods appear on the screen when you run the ping command. Each period is a dropped packetif they accumulate in a burst then something is happening that you would hear on the phone. WOW.. That is the most complicated Ping I have ever seen.. :) This is the result I got. # ping -f -i .02 -s 180 -Q 0xb8 xx.xx.xx.xx /PING xx.xx.xx.xx (xx.xx.xx.xx) 180(208) bytes of data. . --- xx.xx.xx.xx ping statistics --- 15338 packets transmitted, 15325 received, 0% packet loss, time 352748ms rtt min/avg/max/mdev = 276.499/286.185/310.118/7.248 ms, pipe 15, ipg/ewma 22.999/284.882 ms / The same test with my Present SIP Provider gave me the result below. /10926 packets transmitted, 10913 received, 0% packet loss, time 244048ms rtt min/avg/max/mdev = 289.514/292.668/316.350/2.336 ms, pipe 15, ipg/ewma 22.338/292.941 ms / I suppose the value of mdev is much higher in the first case but 0% packet loss in both the cases. Does this mean that the voice quality is going to be real bad?? Thanks, Najim On Thu, Dec 1, 2011 at 6:33 AM, Adam Moffett adamli...@plexicomm.net mailto:adamli...@plexicomm.net wrote: a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. I am probably splitting hairs, but that's not always true because there's no guarantee that the reply traveled the same path as the echo request. If you dig into BGP issues you'll see sometimes that traffic one direction takes a different route than traffic the other direction. I don't know of any simple and accurate way to learn the one way latency so I'm surprised they specified anything other than round trip time. 'Ping time' is not an accurate predictor of SIP quality. A 'ping' is an ICMP Echo/reply packet and some routers consider them less important than 'data' packets and service them on an 'as resources permit' basis. That's possibly maybe true if someone's router or connection is overloaded and they are trying to make up for it with CoS policies while they save up for an upgrade. Otherwise it's an apology for a crappy network. That's the brutally honest truth. You can make a pretty good prediction with ping. sudo ping -f -i .02 -s 180 -Q 0xb8 [ip] gives a tolerable simulation of voip traffic. let it run for awhile, then press ctrl+c and see how many packets were dropped and also check the mdev number. If mdev is low and packet loss is almost nothing then you can expect decent voice quality. It may not be a 100% perfect test, but I'll bet you a vast majority of the time I can do that test and tell you whether it's going to suck. latency by itself with low jitter and no packet loss just means delay. It's a matter of opinion and circumstance how tolerable delay is, but I think your 230ms ping is at the upper edge of what most people can live with. Much more than that and you'll be tempted to say 'over' at the end of sentence. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Best VoIP conferencing phone ?
Thank you for sharing your exp. with me. On Wed, Nov 30, 2011 at 7:34 PM, Darren Wiebe dar...@aleph-com.net wrote: We've been happy with the polycom IP 7000. Darren Wiebe On Nov 30, 2011 1:40 AM, virendra bhati virbh...@gmail.com wrote: Hi Faisal, Thanks for reply but I want hardware wase VoIP device. If know please gussed me. From google I fould the list of below devices but I am not sure that these are best for used or have an issue *1)Polycom SoundStation IP 7000 * *Why it's best: *The Polycom SoundStation IP 7000 is the most advanced conference phone from the Polycom SoundStation lineup and leaves little to be desired. With an amazing 20’ 360 radius, the 7000 is perfect for large conference rooms. The new HD voice quality (22 kHz) allows. * * *2) Polycom Voicestation 500* * * *Why it's a best pick: *The Polycom VoiceStation 500 is one of the best conference phones for a wide variety of reasons. The VoiceStation 500 features amazing call quality, 7’ 360 radius, Bluetooth connectivity, wired connection, background noise reduction, and an attractive design. * * *3)Panasonic - 8-Microphone Speakerphone with Caller ID KX-TS730S* * * *Why it's a best pick: *With a 360 10’ radius and 8 microphones, everyone is sure to be heard with the Panasonic KX-TS730S. The multiple microphones allows for everyone sitting in on the conference to be heard uniformly without distortion. * * *4)Cisco Unified IP Conference Station 7937G Conference VoIP Phone* * * *Why it's a best pick: *The Cisco 7937G works via VoIP connection, has stunning call clarity, and features a simplistic but expensive design that is easy to use. Cisco is an industry leader in IT communication products, and the 7937G is no different. The 360 design allows everyone to be heard. * * *5)Polycom SoundStation VTX 1000* * * *Why it's a best pick: *The SoundStation VTX 1000 is an incredible conference phone, but it is very pricey and not as good as advertised. The VTX 1000 is designed for large conference rooms and features upgradable software (which is a huge benefit since the cost is so high), 20’ 360 radius. 6)Polycom® SoundStation® IP 5000* 7) GXP2120 6-line Executive HD IP Phone * On Wed, Nov 30, 2011 at 1:47 PM, Faisal Hanif fai...@vopium.com wrote: I have tried EyeBeam and it worked fine with x members audio conference however it need resources (Processing + RAM) per additional line. ** ** Regards, ** ** Faisal Hanif ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati *Sent:* Wednesday, November 30, 2011 11:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind *Subject:* [asterisk-users] Best VoIP conferencing phone ? ** ** Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Sound files with MixMonitor not playable with Media Player
On 11/30/2011 09:45 PM, Danny Nicholas wrote: Check this link -- you might be recording a muted file http://www.centos.org/modules/newbb/print.php?form=1topic_id=34058forum=37order=ASCstart=0 http://www.centos.org/modules/newbb/print.php?form=1topic_id=34058forum=37order=ASCstart=0 Like I said : I can play the sound file with Totem on Linux or VLC-player on Windows. So it's not that the wav-file has no sound... Jonas. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, November 30, 2011 2:19 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Sound files with MixMonitor not playable with Media Player Hello, it worked perfectly before... I just did a clean install of my Asterisk server and changed nothing but Centos 5.6 to CentOS 5.7 Therefore I ask if it should be something that I'm missing on my system ? Jonas. On 11/30/2011 08:59 PM, Danny Nicholas wrote: Since the other data seems kosher, have you tried just renaming the file without the -, _ and : ? *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, November 30, 2011 1:55 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Sound files with MixMonitor not playable with Media Player Hello, the wav sound files that are created by using MixMonitor()-command are not playable with Windows Media Player. I can play them with vlc-player and on my Fedora with Totem. This is one of the files : /var/ftp/104/2011-11-30_11:54:39_89000404.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz What would be missing on the system (Centos 5.7) that makes wav-files difficult for Windows Media Player ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
