Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

hat are > automatically sent to the AGI script from Asterisk. I do not know why > you are getting the channel instead of the extension, you could try > giving the extension as a parameter to the AGI script if you cannot get > that from the included request variable. > > On Mon, 20

[asterisk-users] Small PC for Asterisk appliance to support 2 x Sangoma A200 (2 x PCIe standard cards)

Hi Guys, Looking for a powerful box that is compact, can take two hard drives for Raid-1 (no SSD, too expensive), have at least two Gig ports or two 10/100mbps ports. Fit two PCIe or one PCIe card plus it's daughter card which needs as much room as a PCIe and doesn't need the actual slot. That is

Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

right because if you notice the last three charecters of that line is* "")*. So, when the phpagi path is correct, it looks like: *"415444555")*. -Bruce On Mon, Jun 14, 2010 at 6:09 PM, Steve Edwards wrote: > > On Mon, 2010-06-14 at 14:57 -0400, bruce bruce wrote: >

Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

is always returned as 0. $peer_count = system('asterisk -rx "sip show peer $sip_peer" | grep -c "X-Lite"', $retval); Should $sip_peer be inside another set of parenthesis? Thanks, Bruce * On Mon, Jun 14, 2010 at 6:44 PM, Steve Edwards wrote: > On Mon, 14 Jun 2010,

Re: [asterisk-users] asterisk issue

Nice and colorful tutorial for cronjobs. http://www.linuxconfig.org/Linux_Cron_Guide -Bruce On Fri, Jun 18, 2010 at 1:55 PM, salaheddine elharit < salah.elharit...@gmail.com> wrote: > thanks for your response > > how can i create and execute this cron > > 2010/6/18 Danny Nicholas > >> I do a c

[asterisk-users] Using SetVar with System() is it possible?

Hi Guys, Is it possible to harvest the output of system into a SetVar(variable)? exten => s,n,SetVar(var=system(*asterisk -rx "sip show channels" | grep -c "(ulaw)")* * * *??? any problem with the syntax? * * * * * *Thanks,* * * --

Re: [asterisk-users] Using SetVar with System() is it possible?

with first three digits being 789 (for example) as all of those variables can help me move to next step which is to decide to place a second call through the same trunk or not. Any inputs? Thanks a lot On Sat, Jun 19, 2010 at 1:56 PM, Tzafrir Cohen wrote: > On Sat, Jun 19, 2010 at 10:58:17AM

[asterisk-users] Deleting some of the CDR data - How to do it safely?

Hi Guys, I am looking to delete some of the CDR logged by Asterisk in asteriskcdrdb in a PbxinaFlash system running Asterisk 1.4.x The CDR records to deleted are probably a big chunk and spread out all through the database but I basically want to delete all calls that came in through a specific D

[asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

Hi Everyone, I want to know if a specific codec type is used at least one. For example, I want to know if out of the 100 calls on the system if there is a 1 channel that is running G.729 codec right now. If using dial-plan and I dial in, I can use this to obtain information about CURRENT channel.

[asterisk-users] Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver

Hi Guys, An 8 channel Astribank is connected to Trixbox 2.8 and I ran freepbx-module-zapauto but I get the following when running these commands and can't make calls out: [Trixbox]# dahdi_genconf xpporder /usr/sbin/dahdi_genconf: warning - OLD DRIVER: missing '/sys/bus/xpds/devices/00:0:0/timing_

Re: [asterisk-users] Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver

ttp://www.xorcom.com/downloads/astribank2-trixbox-ce-drivers.html#trixboxce2.8> On Tue, Jun 22, 2010 at 5:25 AM, Tzafrir Cohen wrote: > On Tue, Jun 22, 2010 at 12:58:00AM -0400, bruce bruce wrote: > > Hi Guys, > > > > An 8 channel > > FXO? > > > Astri

Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

gt; > native; > > done > > The overhead of each 'asterisk -rx' command is noticable. If you have 10 > calls or more, this can have an odd effect. > > Not to mention that the fact that it is so slow exposes its raciness[1]. > > > > > On 21 June 2010 16:08,

Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

and rewrite it in C. thoughts> > > > > -Elliot > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce > *Sent:* Tuesday, June 22, 2010 1:32 PM > *To:* Asterisk Users Mailing List - Non-

[asterisk-users] Xorocom Missing files...where to get it? astribank_upgrade

Hi Everyone, I was on Xorocom site but there is no clear and consice place to download drivers and firmware. I am reading their instructions to install Astribank 8 channel FXO on Trixbox 2.8 and I seem to be missing files at this step: [pbx.archology.com dahdi]# /usr/share/doc/astribank_upgrade /

Re: [asterisk-users] Xorocom Missing files...where to get it? astribank_upgrade

arate Trixbox from Elastix and version to version. On Tue, Jun 22, 2010 at 4:53 PM, Steve Edwards wrote: > On Tue, 22 Jun 2010, bruce bruce wrote: > > > I was on Xorocom site but there is no clear and consice place to > > download drivers and firmware. I am reading their inst

Re: [asterisk-users] I look ARI (Asterisk Recording Interface)

It's one of the bad modules that goes with FreePBX anyhow. The moment you go over 3000 recordings you are already in trouble. It's about time someone come up with a better moduel. On Wed, Jun 23, 2010 at 11:05 AM, Mickael Monsieur < mickael.monsi...@gmail.com> wrote: > Hello, > I look ARI (Asteri

[asterisk-users] A lot of : doing dnsmgr_lookup for - Asterisk installed from YUM

Hi Guys, Asterisk 1.6.2.7 install from Yum Repository shows a lot of :> doing dnsmgr_lookup for sip.provider.com Google searches show it was fixed in some version. Is this to be ignored? Thanks -- _ -- Bandwidth and Coloca

[asterisk-users] What are the guts of AsteriskNOW and how it compares to other popular flavors available?

Hi Everyone, I want to know a bit about the guts of the current AsterisNOW system. I know that FreePBX is embraced as the main GUI but is just an install of CentOS 5.4 + (Asterisk/FreePBX from Yum repos)? - Or is there anymore to this? Maybe some security tools? - Or is Asterisk built from the so

[asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

Hi Everyone, I am accustomed to PUTTY and it's very nice as in it allows many many SSH profiles to be saved and allows tunneling etcbut it's not very good when it comes to scrolling up and down, colors, text size, and specially it doesn't give a title to the opened instance. Maybe giving the I

[asterisk-users] Anyone can share their config file for Cisco phone please?

I have an *ipphone 7965G* which has to be connected to Asterisk. It has been flashed with SIP firmware but the config file doesn't seem to work maybe I am missing something in it. I appreciate it if you can share your working sample config file with me. Thanks --

Re: [asterisk-users] Anyone can share their config file for Cisco phone please?

irectory_url: "http://192.168.20.4/xmlservices/phonebook.xml"; > > logo_url: "http://192.168.20.4/images/logo.bmp"; > > SIP_MAC_ADDR.conf > > proxy1_address: 192.168.20.4 > > ; Line 1 phone number > line1_name : 246 > > ; Line 1 name for authent

Re: [asterisk-users] Anyone can share their config file for Cisco phone please?

Thanks a lot. I will look into it. On Wed, Jun 30, 2010 at 11:15 AM, Warren Selby wrote: > On Wed, Jun 30, 2010 at 8:40 AM, bruce bruce wrote: > >> Thanks a lot. >> >> -Bruce >> >> >> On Wed, Jun 30, 2010 at 4:55 AM, Emanuele Carbone wrote: >> &

Re: [asterisk-users] Problem in establish call from a2billing users.

Yes, you are missing a whole bunch of configurations from creating SIP users to making sure they show as peers on Asterisk to making sure you use dnid, etc.You probably might want to search google for some configuration help On Wed, Jun 30, 2010 at 11:24 AM, gokulakrishnan wrote: > Hi All, >

[asterisk-users] Anyway to know when a channel is going to hangup if Dial Timeout option is used?

Hi Guys, I have a channel that is dialed with *Timeout* option. So, there is definite time to it. Only thing is that I don't have control of that channel. I only know that it's using g729 codec and that there is only one channel that is using g729 at any given time. So, my question is: >From with

Re: [asterisk-users] What are the guts of AsteriskNOW and how it compares to other popular flavors available?

Anything guys? Thanks On Mon, Jun 28, 2010 at 10:20 PM, bruce bruce wrote: > Hi Everyone, > > I want to know a bit about the guts of the current AsterisNOW system. I > know that FreePBX is embraced as the main GUI but is just an install of > CentOS 5.4 + (Asterisk/FreePBX

[asterisk-users] Why does my IAX2 trunk between two office hangup a channel after 30 seconds? Can you share your IAX2 trunking configuration? URGENT HELP much appreciated

Hi guys, I have two Asterisk servers (with FreePBX) connected together with IAX2 trunking. When I call from server A->B call connects but hangs up after 30 seconds. What could be cause? Can anyone please share working configuration between two asterisk server in IAX2 trunking for FreePBX? Thanks

Re: [asterisk-users] Anyway to know when a channel is going to hangup if Dial Timeout option is used?

I find out it's remaining EPOCH time? Thanks On Sun, Jul 4, 2010 at 12:03 PM, Steve Edwards wrote: > On Sun, 4 Jul 2010, bruce bruce wrote: > > I have a channel that is dialed with Timeout option. So, there is definite >> time to it. Only thing is that I don't have cont

Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

or MS > Windows platform and WHY? > > On 06/29/2010 06:53 AM, bruce bruce wrote: > > Hi Everyone, > > > > I am accustomed to PUTTY and it's very nice as in it allows many many > > SSH profiles to be saved and allows tunneling etcbut it's not very > &g

Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

Just downloaded PrivateSHELL and it seems to be what everyone is looking for in Putty. It's much better than putty in terms of not being sluggish and scrolling is fine. Plus the window and the text doesn't hurt your eyes. It has One click SFTP as well. So, good bye to WinSCP. I think I found what

[asterisk-users] Y-cords - What are they ?

Good Afternoon, Can someone please explain what Y-cords are available out there and how they can be used with Aastra or other VoIP phones? Maybe with or WITHOUT headsets? Isn't a Y-cord traded for soft Barge in these days? Thanks, Bruce --

Re: [asterisk-users] Y-cords - What are they ?

Thanks for the input guys. My client is looking for Y-cords to train people. So, set beside them take a call and let them listen on the other call. They currently use wireless Plantronic headset with Aastra phones. Can you suggest any specific vendors for Y-cords? Thanks On Tue, Jul 6, 2010 at 4:

[asterisk-users] GURUs - How to monitor all MySQL actions on an Asterisk/FreePBX server?

Hi Guys, This is something related and yet un-related to Asterisk. I have a FreePBX/Asterisk server running and I want to trace everything that FreePBX does to MySQL. Is there a verbose CLI to MySQL that I can pull up on terminal and make configuration change to FreePBX and see it in real-time on

Re: [asterisk-users] GURUs - How to monitor all MySQL actions on an Asterisk/FreePBX server?

tever the name of the mysql log file is. > > Zeeshan A Zakaria > > -- > www.ilovetovoip.com > > On 2010-07-07 9:43 PM, "Carlos Chavez" wrote: > > *On Wed, 7 Jul 2010 19:19:28 -0400, bruce bruce wrote* > > > Hi Guys, > > > > This is someth

[asterisk-users] How to know which party hangup() first when using analogue cards? - Dahdi and Asterisk 1.4.x

Hi Everyone, I am trying to find the issue of dropped calls in the middle of the conversation. The system is Elastix. Anyway to know which party hangup the channel in case of Asterisk 1.4 and Sangoma analogue cards? (this is not PRI) Thanks, Bruce --

Re: [asterisk-users] GURUs - How to monitor all MySQL actions on an Asterisk/FreePBX server?

[mysqld_safe] log-error=/var/log/mysqld.log pid-file=/var/run/mysqld/mysqld.pid log=/var/log/mysql_query.log *But it doesn't log anything to /var/log/mysql_query.log* What else am i missing? Thanks On Thu, Jul 8, 2010 at 1:20 AM, Steve Edwards wrote: > On Thu, 8 Jul 2010, bruce bru

[asterisk-users] Rx/Tx fine tuning of analogue card to PRI card - Am I right with my theory?

Hi Everyone, I want to fine tune the Rx and Tx gain on an analogue Sangoma card by dialing into another server that is running on Sangoma PRI card (both services on Bell network). [mwatt1004khz] exten => s,1,Answer exten => s,n,PlayTones(1004/1000) exten => s,n,Wait(300) If I match the Rx/Tx num

Re: [asterisk-users] GURUs - How to monitor all MySQL actions on an Asterisk/FreePBX server?

Putting it in /tmp/ just did the job. Sorry, I posted my older my.cnf file. I actaully did have the log under mysqld rather than the safe version but it didn't work. I will put this to privilage problems. On Thu, Jul 8, 2010 at 9:55 PM, Steve Edwards wrote: > On Thu, 8 Jul 2010, bru

[asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

Hi Guys, I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use 2 Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet height. Is that enough? Is there calculator online I can use to determine the number of speakers needed? I guess these speakers go in chain so

Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

Thanks fro the input. The area is a 4 square feet. So, you are saying that if I use four speakers then they would not be as loud as needed? Thanks again 2010/7/9 Massimo Nuvoli > bruce bruce ha scritto: > > Hi Guys, > > > > I am looking to buy a 25 Watt output CyberDa

Re: [asterisk-users] General network question regarding SIP and IAX2

The variable is *canreinvite.* *Please check on voipinfo. If canreinvite is enabled then only SIP signaling is passed through Asterisk and the media is not passed through Asterisk resulting in less bandwidth usage and probably less jitter buffer, etcif you are two phones are closer to each othe

Re: [asterisk-users] General network question regarding SIP and IAX2

<http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite>-Bruce On Fri, Jul 9, 2010 at 2:40 PM, wrote: > Sounds great, thanks for your answer. > Do i need to set this on the trunk, the friend or on both? > > > > > -----Original Message- > From: bruce bruce >

[asterisk-users] PHP can't insert - Can someone please help

Hi Guys, I am making another module for Voicemail. I have three fields in a POST form that have to be connected together to make it a single 10 digit number but there is something wrong in my syntax probably. $npaa = "('$_POST[anpa]')"; $nxxa = "('$_POST[anxx]')"; $blocka = "('$_POST[ablock]')";

Re: [asterisk-users] PHP can't insert - Can someone please help

t seems to be SQL and I'm guessing MySQL at > that. > > Next, you seem to be accepting user input and not sanatizing it. DANGER > WILL ROBINSON!!! > This is bad, because it leaves you open to something known as a "SQL > injection attack". > > Now, as to synta

Re: [asterisk-users] PHP can't insert - Can someone please help

2. Thanks On Sat, Jul 10, 2010 at 10:21 AM, bruce bruce wrote: > Thank you for the amazing reply. First few lines of your e-mail was EXACTLY > getting me to where I made a mistake. I guess I didn't take the () and ' ' > at their face value and was looking somewhere else

Re: [asterisk-users] PHP can't insert - Can someone please help

chools.com/welcome.php?fname=Peter&age=37 Thanks a lot, Bruce On Sat, Jul 10, 2010 at 1:41 PM, Gerald A wrote: > Hi Bruce, > > On Sat, Jul 10, 2010 at 11:12 AM, bruce bruce wrote: > >> Further to my last post, I added this to santize. I also created a new >> mysql user

Re: [asterisk-users] How can get user inputs from called party after dial?

You need read(): http://www.voip-info.org/wiki/view/Asterisk+cmd+Read It's as easy as: exten => s,n,Read(variable,,11) exten => s,n,NoOp(${variable}) Above will take up to 11 digits input by user and will display it back in NoOP on Asterisk C

Re: [asterisk-users] PHP can't insert - Can someone please help

$nxxa) && is_numeric($blocka) && $npaa>=200 && $nxxa>=200 && $npaa!=900 && $npaa!=911) * * * * {* echo "Number passed sanitization"; } What do you think? :-) -Bruce On Sat, Jul 10, 2010 at 2:17 PM, bruce bruce

Re: [asterisk-users] How can get user inputs from called party after dial?

ed. > > > > Eyal > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce > *Sent:* Saturday, July 10, 2010 9:30 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subj

Re: [asterisk-users] How can get user inputs from called party after dial?

For dial you do this: [first-Dialplan] exten => s,1,Answer exten => s,n,Dial(SIP/provider/111222) exten => s,n,Playback(Welcome) exten => s,n,Read(numb,,10) exten => s,n,NoOp(${numb}) -Bruce On Sat, Jul 10, 2010 at 2:51 PM, bruce bruce wrote: > You need to do some read

Re: [asterisk-users] How can get user inputs from called party after dial?

I was under the impression that he is new to Asterisk. No need to fuss. Hence the ":-)" On Sat, Jul 10, 2010 at 3:35 PM, Steve Edwards wrote: > On Sat, 10 Jul 2010, bruce bruce wrote: > > > You need to do some reading :-) > > Now that is funny -- maybe you could take

[asterisk-users] My own FreePBX FollowME module - Stuck at Reload - Anyone else had experience with this?

Hi Everyone, I have done some php coding to come up with my own FollowME module for FreePBX. The need for this has some security considerations behind it. This is what my code does at core: $sql="REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre, grplist, annmsg_id,postdest, dring, need

Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

ing 5 ft > has never killed anyone, but this really depends on the power of the > speaker, I usually deal with 70v speakers tapped at 16 or 8 watts > depending on how many speakers I put on one amplifier and the output > wattage of that amplifier. > > > > On Fri, Jul 9, 2010

Re: [asterisk-users] My own FreePBX FollowME module - Stuck at Reload - Anyone else had experience with this?

Tim Nelson wrote: > - "bruce bruce" wrote: > > Hi Everyone, > > > > I have done some php coding to come up with my own FollowME module for > FreePBX. The need for this has some security considerations behind it. > > > This is what my code does at

[asterisk-users] How to install speex codec for Asterisk that is downloaded from Digium Yum Repository?

Hi Everyone, I have done "yum install speex libspeex-devel speex-devel" and it was succesful on CentOS. I then tried "yum install asterisk16 asterisk16-addons asterisk16-configs" but "core show translation" doesn't show speex loaded. Is there a way to or an option that I can append to the asterisk

Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

which ended up 3 facing one way and 2 the other. You > can get double horn speakers which will face 2 sides. I wouldn't mount > them on the wall specifically not so low as fork lifts and what not > will damage them. > > > On Mon, Jul 12, 2010 at 2:17 PM, bruce bruce wrote:

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

I am stuck with the same problem but I have used asterisk yum repository and it worked by itself without me worrying for kernel stuff. However, I need to install speex codec and now I am stuck as it doesn't get picked up by the yum asterisk install somehow. I have lib speex and speex already insta

[asterisk-users] Where should I look for MWI settings if Aastra phones don't do it?

Hi Guys, Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and 6730i, but none of them indicate the voic-email. Where should I look for trouble to find the root issue for MWI? Thanks, -- _ -- Bandwidth and

Re: [asterisk-users] Where should I look for MWI settings if Aastra phones don't do it?

12:25 PM, Steve Johnson wrote: > On Wed, Jul 14, 2010 at 10:04 AM, bruce bruce wrote: > > Hi Guys, > > Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, > and > > 6730i, but none of them indicate the voic-email. Where should I look for > > t

[asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?

Hi Everyone, Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? Even though FreePBX Inbound has an option for Alert_INFO but that doesn't work when the call comes into an IVR or Queue. The calls has to go d

Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?

Thanks for the input but that won't be good because people are not going to remember two extensions for one person. The sip header should be able to carry alert_info to internal extensions really easily. Anyone else got a thought? Thanks again, On Wed, Jul 14, 2010 at 5:44 PM, Ira wrote: > At

Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

On Wed, Jul 14, 2010 at 12:39 AM, bruce bruce wrote: > > Thanks for the input guys. For other refrence, a CyberData Voip Amplifier > > which supplies 10 Watt to each of the two bogen 30 Watt speakers did the > job > > for a 35, square feet warehouse with environmental nois

[asterisk-users] Does Flash Operator Panel allow for dragging a call into a parking lot?

Hi Everyone, If I receive a call on a ZAP line and pickup the call and drag and drop it (by mouse) into a Parking Lot through FOP, it just hangs up. Is this feature supported by FOP? Thanks, Bruce -- _ -- Bandwidth and Colocatio

Re: [asterisk-users] Does Flash Operator Panel allow for dragging a call into a parking lot?

It's doable with a work around. Create a misc extension with followme set to ##70# which point to your parking lots and failed destination to Misc parking extension. Regards, Bruce On Sun, Jul 18, 2010 at 3:38 PM, Doug Lytle wrote: > bruce bruce wrote: > > Hi Everyone, > >

Re: [asterisk-users] POE Splitters

The Aastra 53i draws only 2 Watts from a Linksys 24 port POE switch. 25 phones is around 55 Watts. -Bruce On Thu, Jul 22, 2010 at 5:16 PM, Andrew Latham wrote: > The Snom 360 phone in front of me draws 4w... > > > ~ > Andrew "lathama" Latham > lath...@gmail.com > > * Learn more about OSS http:/

[asterisk-users] Why does a bridged channel stay open for 4 hours?

Hi Everyone, Using a PRI with Sangoma A101D and Asterisk 1.4.2.x. I notice that occasionally after a call is disconnected and both the phone devices and the the channel is down but the bridge stays open for hours. Channel Location State Application(Data) Local/9054445.

Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

This is running Elastix (FreePBX), so I am pretty sure there is Hangup() requested. At least this doesn't happen ALL THE TIME. So, something is getting stuck. Thanks, Bruce On Fri, Jul 23, 2010 at 9:10 AM, Paul Belanger wrote: > On Fri, Jul 23, 2010 at 1:16 AM, bruce bruce wrote: >

Re: [asterisk-users] POE Splitters

You can also use Ethernet Over Power Lines solution or wireless :-) On Fri, Jul 23, 2010 at 8:55 AM, David Backeberg wrote: > On Fri, Jul 23, 2010 at 8:46 AM, Matt wrote: > > It's not necessarily this simple. There is an approximately 50-75foot > cable > > run through ceilings and walls (CAT5)

Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

Well, what about PRI? Why should this stay on? Isn't the native bridge just a bridge channel that should go down automatically if the actually Dahdi/ZAP channel is down and there are no SIP channels on either? Thanks, Bruce On Fri, Jul 23, 2010 at 5:09 PM, Tzafrir Cohen wrote: > On Fri, Jul 23,

Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

I am having this issue with PRI. But I do not use conference rooms. Our system is a simple queue and extensions. -Bruce On Fri, Jul 23, 2010 at 6:13 PM, Maurizio Faccio adinet < mauf...@adinet.com.uy> wrote: > You're right but it do not detect that I hungs on my side of the line. > I think that

[asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?

Hi Guys, I seem to not be able to find any good open source Asterisk Queue Analyzer and Asterisk Log Analyzer on the web. The Asterisk Queue Analyzer is to serve as the graphic tool for call center or pbx admins. It will pull the info in queue.log and in MySQL asterisk CDR to create a graphic bar

Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?

e that are neat in interface or useful in features. I guess no one else has any thoughts on this? Maybe there is none available? Thanks, Bruce On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg wrote: > On Mon, Jul 26, 2010 at 11:34 PM, bruce bruce wrote: > > I seem to not be able to

[asterisk-users] Why do Zaptel calls drop all of a sudden? Could busy detect be the problem?

Hi Guys, I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: [Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy, avgtone: 75, avgsilence 135 [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [...@macr

Re: [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?

asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce > *Subject:* [asterisk-users] Why do Zaptel calls drop all of a sudden? > Couldbusy detect be the problem? > > > > I am getting a complain that call on analogu

Re: [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?

Furthermore, these are lines in Hunt, so, I am not sure if Call-Waiting is turned ON on these lines at all. But it's definitely an interesting idea. On Wed, Jul 28, 2010 at 5:54 PM, bruce bruce wrote: > Hmmwhat about call waiting? > You mean, when a call comes in on that specif

[asterisk-users] Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1

Hi Everyone, This is probably more related to Linux than to Asterisk. Analogue channels on a system were un-responsive on Monday morning. Apparently something happened over the weekend and the router went off or it lost it's DSL connection. [Jul 23 22:50:01] VERBOSE[12437] logger.c: -- Remote

Re: [asterisk-users] Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1

ng of the 26th at 9:22 the server was restarted because it was un-reachable from outside and hence the restart log but where is the 24th, and 25th? Thanks, Bruce On Thu, Jul 29, 2010 at 9:10 AM, Paul Belanger wrote: > On Wed, Jul 28, 2010 at 9:06 PM, bruce bruce wrote: > > See the jump

[asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

Hi Everyone, I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The phones occasionally go into "No Service" mode. The POE switch doesn't seem to be the problem as it's tested fine. I think the router sometimes gives up and comes back quickly. Or something of that nature. However, t

Re: [asterisk-users] Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1

logs. Even MS Blue Screed Of Death does a better job of logging at instances like this :-( On Thu, Jul 29, 2010 at 10:13 PM, Lyle Giese wrote: > Lyle Giese wrote: > > bruce bruce wrote: > > I am not sure why it would be sleeping. I have never dealt with putting a > linux serv

Re: [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?

end to someone for code patch... > > regards > Martin > > On Wed, Jul 28, 2010 at 4:54 PM, bruce bruce wrote: > > Hmmwhat about call waiting? > > You mean, when a call comes in on that specific line, it generate two > beep > > tones and hence the system hang

Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?

free. >> >> Zeeshan A Zakaria >> >> -- >> www.ilovetovoip.com >> >> On 2010-07-27 6:12 PM, "bruce bruce" wrote: >> >> :-) I knew someone would bring up FreePBX. I have FreePBX installed and >> it's not good for Queues at all

Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

Adria, How can I build a dns cache into my lan? I am using a Linksys 48 port POE switch and running a micro DD-WRT firmware on a linksys router. Gareth, I think the registration time is part of the reason. I have lowered it less than 10 seconds. Thanks On Fri, Jul 30, 2010 at 8:21 AM, Adrià Vi

Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

DNSMasq has always been enabled. It's only one check box and if I didn't have it enabled phones won't work. So, that is set. Any other suggestions? including things regarding DNSMasq? Thanks On Fri, Jul 30, 2010 at 11:04 AM, Dave Cotton wrote: > On 30/07/10 16:15, bruce bruc

Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

. At best, I can think of a cable or two jacked improperly into the patch panel and that's all which MAYBE the cause for failing of DNS. Thanks, Bruce On Fri, Jul 30, 2010 at 12:03 PM, bruce bruce wrote: > DNSMasq has always been enabled. It's only one check box and if I didn't

Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?

2 users. So, it's probably never used as a free version as probably there are no 2 seat call centers that can survive this economy. But, it should great for testing. On Sat, Jul 31, 2010 at 10:28 AM, Leif Madsen wrote: > On 7/30/2010 5:49 AM, Lenz Emilitri wrote: > > QueueMetrics is actually free

[asterisk-users] What do you use for Invoicing?

Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to creat

Re: [asterisk-users] What do you use for Invoicing?

Maybe good but the first look brought me to a Pay version. Doesn't satisfy the opensource condition. thanks, On Mon, Aug 2, 2010 at 2:39 PM, Jeff LaCoursiere wrote: > On Mon, 2010-08-02 at 14:26 -0400, bruce bruce wrote: > > Hi Everyone, > > > > > > Sorry,

Re: [asterisk-users] What do you use for Invoicing?

Sorry, I am not familiar with them. Wondering if any full package system out there does the job. Thanks On Mon, Aug 2, 2010 at 2:55 PM, Gordon Henderson > wrote: > On Mon, 2 Aug 2010, bruce bruce wrote: > > > Hi Everyone, > > > > Sorry, if it's not directly rel

Re: [asterisk-users] What do you use for Invoicing?

I agree but the mentioned software is not opensource. My conditions clearly included opensource. On Tue, Aug 3, 2010 at 12:35 AM, Nick Brown wrote: > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce &

Re: [asterisk-users] What do you use for Invoicing?

ICTInvoice > only work with Elastix 1.6 > > Regards > > On Mon, Aug 2, 2010 at 11:26 PM, bruce bruce wrote: > >> Hi Everyone, >> >> Sorry, if it's not directly related to Asterisk. Some of people on this >> list might have PBX deployed for their clients.

Re: [asterisk-users] What do you use for Invoicing?

Yep, I seen that. That is probably the closet thing but looking at he interface it makes me not try to install it. Maybe too complicated. I wouldn't want to send customer the whole CDRs but rather a nice Bill like the telco sends out. I am currently toying with NCH Invoicing. Those guys make a sof

Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

> Mike > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce > *Sent:* Thursday, July 29, 2010 22:36 > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users]

[asterisk-users] How to determine which party hangup th e call? cause of Hang-up needed.‏

Hi Everyone Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell Canada. User claims that call hangup without any interferance to the phone set. Is there ANYWAY to find out which party hang-up the call or if the call was cut-off due to other reasons? I checked the *"asterisk

[asterisk-users] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?

Hello Everyone, I am trying to diagnose issue with my IAX2 extension not working. When I have iax2 set debug on all I see is this: *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 3ms SCall: 00130 DCall: 0 [64.229.229.111:64823]* * USERN

Re: [asterisk-users] How to determine which party hangup th e call? cause of Hang-up needed.‏

* *Thanks,* *Bruce * On Wed, Aug 11, 2010 at 12:33 AM, Faisal Hanif wrote: > read the value of var ${HANGUPCAUSE} next line to dial command. > > Regards, > > Faisal Hanif > *VoIP Manager > ***Vopium A/S > On 8/10/2010 9:51 PM, bruce bruce wrote: > > Hi Everyone > >

[asterisk-users] Asterisk with Motorola Canopy

Hi Everyone, Can anyone share their experience with Motorola Canopy solution deployment and Asterisk? Is this a good fit? Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk?

[asterisk-users] Pfsense and IAX2 - What is the proper firewall NAT setup?

Hi Everyone, Just trying to connect the Zoiper Communicator to connect to Asterisk which is behind Pfsense. Here is what I get at debug and it doesn't register. Error code 16. Can someone please let me know their firewall, NAT, outbound 1-to-1 pfsense settings as it seems to me I am doing somethin

Re: [asterisk-users] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?

your peer configuration > > Regards > > On Wed, Aug 11, 2010 at 4:28 AM, bruce bruce wrote: > >> Hello Everyone, >> >> I am trying to diagnose issue with my IAX2 extension not working. >> >> When I have iax2 set debug on all I see is this: >> >> *Rx-

[asterisk-users] Opensource Speech recognition for Asterisk

Hi Everyone, Has anyone got any opensource speech recognition software to work with Asterisk? Please only list WORKING ones. Not the "theoretically" should work ones! Thanks -- _ -- Bandwidth and Colocation Provided by http://ww

Re: [asterisk-users] Opensource Speech recognition for Asterisk

Thanks guys. A lot of info here :-) I am wondering if anyone followed this and it was working for them: http://scribblej.com/svn/ ??? I am not looking for anything fancy. The basic "yes", "no", dialing a number, asking for agent, etc...out of which probably the hardest is a 10 digit number to b

Re: [asterisk-users] Opensource Speech recognition for Asterisk

Bob, Both ZanziIVR and Speechforge have similar look web pages. I guess you have used one of those to get the speech going as this link: http://scribblej.com/svn/ probably is not the full thing. These seem like practical project. Thanks for pointing out. This is what I was looking for. Now start

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