[asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

Hi everyone, How can I introduce some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines so the only thing that I can think of is to limit bandwidth but even that requires quite some scripting work. Is

Re: [asterisk-users] Call Center Reporting

Hi Bilal, Probably there is no open source tool or a good ones available. But few of them I worked with provide up to 2 users free of cost license type of reporting. Reporting for Call Centers can get very complicated. Once you explore some of the commercial apps you will notice how extensive they

Re: [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?

PM, Pezhman Lali wrote: > h is hangup extension, and will be executed after hangup > > > On Mon, Apr 11, 2011 at 6:36 PM, Bruce B wrote: > >> Thanks for the input but I am not sure if that answer my question of if >> it's normal behaviour for AGI scrip to termin

Re: [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?

, try it. > > best > > On Sat, Apr 9, 2011 at 7:22 PM, Bruce B wrote: > >> Hi Everyone, >> >> Trying to run a php script after DeadAGI for A2Billing does it's magic. >> This is the dialplan: >> >> [a2billing] >> exten => _X.,1,Syst

Re: [asterisk-users] Occasional call from "asterisk"

f Of *Brian Henning > *Sent:* Monday, April 11, 2011 8:47 AM > > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' > *Subject:* Re: [asterisk-users] Occasional call from "asterisk" > > > > Bruce B said: > > We experience exact same thing o

[asterisk-users] Asterisk as a Condo door opener/intercom

Hi Everyone, Looking to replace a condo intercom system. Apparently the current one taps into the lines and dials phone numbers but needs to be changed as it's faulty. I will probably still use the same analogue dialing and back it up with a VoIP line and use the current cabling that is in place.

[asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?

Hi Everyone, Trying to run a php script after DeadAGI for A2Billing does it's magic. This is the dialplan: [a2billing] exten => _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN} ${UNIQUEID}) exten => _X.,n,AGI(a2billing.php,1) exten => _X.,n,Hangup() *exten => h,1,Wait(5)* *exten => h,n,Sys

Re: [asterisk-users] Documentation for Asterisk AMI Events?

-04-08 02:56 PM, Bruce B wrote: > >> Hi Everyone, >> >> I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is >> specific to 1.6. I am wondering if the developers cared to write about the >> new events that are spit out in Asterisk 1.8 somewhere

[asterisk-users] Documentation for Asterisk AMI Events?

Hi Everyone, I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is specific to 1.6. I am wondering if the developers cared to write about the new events that are spit out in Asterisk 1.8 somewhere on the web? I checked the tar ball for asterisk 1.8 and documentation doesn't inc

[asterisk-users] Any PHP Ming + for Asterisk guides, tutorial, how-to anywhere?

Hi Everyone, Looking to create a flash status update page from Asterisk events using PHP MING but I can't seem to find much documentation other than those on PHP site. Has anyone ever tried or has came across PHP Ming usage for Asterisk? Any hints of much appreciated. ***I am aware of FOP using

Re: [asterisk-users] Occasional call from "asterisk"

We experience exact same thing on DAHDI with Sangoma USB FXO device on short circuited lines. Phantom calls are actually due to a short in the lines that happen occasionally. -Bruce On Thu, Apr 7, 2011 at 7:16 PM, Warren Selby wrote: > On Thu, Apr 7, 2011 at 4:53 PM, Brian Henning wrote: > >> H

[asterisk-users] Any way to temporarily disable a registered SIP PEER in Asterisk?

Hi Everyone, We want to be able to momentarily or temporarily provide CONGESTION or DE-REGISTER a SIP PEER to asterisk through WEB GUI. I don't want to indulge into dial-plan and write changes to .conf file every-time. Is there any way that a SIP PEER can be de-registered for an amount of time or

Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

Use Count chan_ooh323.so Objective Systems H323 Channel 0 1 modules loaded Regards, On Mon, Mar 14, 2011 at 9:40 AM, Patrick Lists < asterisk-l...@puzzled.xs4all.nl> wrote: > On 03/13/2011 05:27 PM, Bruce B wrote: > >> Indeed ooh323 is a

Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

Patric, Sounds you have limited knowledge of the Asterisk RPMs or maybe never used. Indeed ooh323 is available as part of the RPMs and the correct URL is: http://packages.asterisk.org/centos/5/current/i386/RPMS/ For the commands, I was following the previous post which presented those command an

[asterisk-users] Anyway to monitor SIP debug from originator and terminator separate of each other on two screens?

Hi Everyone, In order to make life easier and to do debugging easier I want to observe "sip set debug originator" and "sip set debug terminator" on two different putty screens. Trick is that originator calls the terminator. I can of course put two separate calls and get sip debugs at different tim

Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

be loaded in some config file after the rpm install? Thanks, On Fri, Mar 11, 2011 at 12:09 AM, Vladimir Mikhelson wrote: > Bruce, > > Forgot to mention. > > ooh323.conf -- configuration file. > > -Vladimir > > > > On 3/10/2011 9:32 PM, Bruce B wrote: > > But

Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

But even with *asterisk16-addons-ooh323.x86_64* I don't see any of the command for h323 in CLI to work. So, I am missing something still. On Thu, Mar 10, 2011 at 10:23 PM, Bruce B wrote: > I see this in the Digium repository: > > *asterisk16-addons-ooh323.x86_64* > > Woul

Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

I see this in the Digium repository: *asterisk16-addons-ooh323.x86_64* Wouldn't that just do it without having to re-install from the source? Thanks On Thu, Mar 10, 2011 at 10:17 PM, Bruce B wrote: > Can you please provide link to the RPM? > > Thanks > > > On Thu,

Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

1.8 related adjustments to OOH323, > should be available in 1.8.4. > > -Vladimir > > > > On 3/10/2011 2:29 PM, Jose P. Espinal wrote: > > Bruce B wrote: > >> Hi everyone, > >> > >> Installed asterisk from yum repository but I think H.323 is no

Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

I was hoping to get a link to h.323 or oh323 rpm maybe But if I have at the end of the day going either an rpm or not MUST I re-install Asterisk? if so, I may as well start doing that now. Thanks again guys. On Thu, Mar 10, 2011 at 3:29 PM, Jose P. Espinal wrote: > Bruce B wrote: > >

[asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

Hi everyone, Installed asterisk from yum repository but I think H.323 is not supported as I tried commands like this and they don't work: - *h.323 debug*: Enable chan_h323 debug - *h.323 gk cycle*: Manually re-register with the Gatekeper - *h.323 hangup*: Manually try to hang up a call

Re: [asterisk-users] Any good tutorials for setting up Asterisk SNMP and Cacti for remote monitoring?

it's install complications) Thanks again, On Mon, Mar 7, 2011 at 2:14 AM, Faisal Hanif wrote: > http://www.danielaliaman.com/blog/files/AsteriskSNMPtutorial.pdf > > > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digiu

[asterisk-users] Any good tutorials for setting up Asterisk SNMP and Cacti for remote monitoring?

Hi Everyone, I have been searching the web and I don't know if SNMP is just that complex to setup or that not many people use SNMP to monitor Asterisk but the information is scattered all over. I have got to the point to configure SNMP with Asterisk and then it's all confusing from there on to ac

[asterisk-users] PRI "wanrouter status" shows disconnected - system problem or Telco?

Hi everyone, I am reading through Sangoma Wiki right now. But someone may already and quickly notice this. I have a system that is down since the morning (maybe power intruptions). All seems fine except for "wanrouter status" shows disconnected. Following are the alarms raised. Should I call telco

Re: [asterisk-users] No ring tone on inbound call - but channelconnects fine

..@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B > *Sent:* Wednesday, February 16, 2011 2:33 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] No ring tone on inbound call - but > channe

[asterisk-users] No ring tone on inbound call - but channel connects fine

Hi Everyone, I have a SIP turnk which works fine with both inbound and outbound calling. However, the only issue is that there is no Ring Tone if someone calls us. The phones used are Aastra and Polycom connected to the PBX via VPN (SIP). I do get an outbound ring tone, so it's not that there is

Re: [asterisk-users] Asterisk Call File using Local Channel not passing Variable back to Dialplan

Hope that helps. > > Mike. > > > "Bruce B" wrote: > > Hi Everyone, > > > > I am trying to pass a variable using the .call files but it turns out > blank. > > Can s

[asterisk-users] Asterisk Call File using Local Channel not passing Variable back to Dialplan

Hi Everyone, I am trying to pass a variable using the .call files but it turns out blank. Can someone please point out what might be wrong here: */tmp/spool-file.sh* *--* echo "Channel: Local/s@callback_leg*1*/n CallerID: \"Call-

[asterisk-users] IP ban list by country

Hi everyone, I know it's off topic from Asterisk directly but yet related. What sources do you use to limit SIP connecting customers to specific countries by IP (e.g. allowing USA and not China). It would help me a lot of you can note the sources you trust that are complete and up to date. Thank

Re: [asterisk-users] Can a duration limit be specified in spool call file?

Thanks Faisal. That is it. I was confused by the fact that there is also the Context, Extension, and Priority in the .call file that should be filled along with the Channle: local. I found out that the call file first calls the local channel context and once that is connected then it moves onto

Re: [asterisk-users] Callback through extensions.conf?

> > > > In my (1.4.X) experience, the file just stays in > /var/spool/asterisk/outgoing and gets “little tags” added until you get the > problem resolved or delete the file. > > > That is absolutely true if the file is not processed. I guess he can again do a "ls -la" in that folder to check permis

Re: [asterisk-users] Callback through extensions.conf?

Asterisk runs as root but what about the bash script or the php file that creates the file? Maybe comment the "mv" command and check the file permissions by *"ls -la call-filename.call"* to be sure. *chown root.root call-filename* (if root is really the user running Asterisk) and then the "mv" com

Re: [asterisk-users] Can a duration limit be specified in spool call file?

On Mon, Feb 7, 2011 at 12:40 PM, Sherwood McGowan < sherwood.mcgo...@gmail.com> wrote: > oh and didn't you guys already have your little histrionics sessin about > trimming the goddamned emails, mailing list etiquette about top posting > versus bottom, etc../.. > > My complaint is not something as

Re: [asterisk-users] Any voice changer applications for Asterisk?

On Mon, Feb 7, 2011 at 8:39 AM, Steve Underwood wrote: > On 02/06/2011 05:05 PM, Sherwood McGowan wrote: > >> AAhem. >> >> https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT >> >> Granted, it's in 1.8, but it's in the documentation ;-) >> >> Cheers >> > That seems to do e

Re: [asterisk-users] Can a duration limit be specified in spool call file?

earch that up on the magical Google search engine you pointed me to. Cheers bud, Sorry, feeling a bit honest today; had to blur it out On Sun, Feb 6, 2011 at 9:42 PM, Sherwood McGowan wrote: > > > On Sun, Feb 6, 2011 at 5:35 PM, Steve Edwards > wrote: > >> On Sun,

Re: [asterisk-users] Can a duration limit be specified in spool call file?

On Fri, Jan 28, 2011 at 7:49 PM, Tilghman Lesher wrote: > On Friday 28 January 2011 18:27:15 Bruce B wrote: > > Hi Everyone, > > > > I don't see any parameter for limiting duration of a call in the .call > > file for Asterisk spool outgoing directory. > > &

Re: [asterisk-users] Any voice changer applications for Asterisk?

> AAhem. > > https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT > > Granted, it's in 1.8, but it's in the documentation ;-) > > Cheers > > Thanks for the pointer. Unfortunately, I am using 1.6 for all my servers now. But I would like to know if anyone tested the new pitch c

[asterisk-users] Any voice changer applications for Asterisk?

Hello, Are there any other other voice changer applications to Asterisk other than the one from Lobstertech? (http://lobstertech.com/voice_changer.html) Specifically interested in open-source but can have a look at economical commercial alternatives as well. Thanks -- ___

[asterisk-users] Can a duration limit be specified in spool call file?

Hi Everyone, I don't see any parameter for limiting duration of a call in the .call file for Asterisk spool outgoing directory. I'd rather not use a MeetMe to drop the call in a conference room and to then limit the call duration as that complicates things unnecessarily. I am wondering if there

[asterisk-users] SIP channel status - Why is it different when calling an internal extension rather than an outside line over SIP?

Hi Everyone, I want to call first party using a .callfile and a second party using a context and then bridge the two calls. I MUST make sure that first party picks up first and then the second party should be dialed. Trying the following using an internal extension works nicely and the playback fi

Re: [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?

Yes, it does. Bell provides the same as well and it works with Asterisk. -Bruce On Fri, Jan 21, 2011 at 7:11 AM, Zeeshan Zakaria wrote: > Hi list, > > For a client I am setting up a system which will use T1 PRI from Primus, > who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I

Re: [asterisk-users] Do I need a sip proxy?

simple setups, its default configuration will not need to be altered much to > get it working. Its logic is VERY different to Asterisk, though. I know that > Kamailio would be a very good choice for this role. I believe the > alternatives would be as well. > > > With kind regards, > P

Re: [asterisk-users] Bruce B

LOL what a looser. Are you a fat admin behind a desk who is going to loose his job due to recession and is pissed off? Here is your first response to one of my first posts: "I was going to respond with some very insightful and helpful information but I'm not a "PRI Guru". Sorry, maybe next time."

Re: [asterisk-users] Top Posting

Since I don't want anyone bitch at my spelling again: news up = nose up :-) -Bruce On Fri, Jan 14, 2011 at 8:55 PM, Bruce B wrote: > It was only the people who ONLY asked in a response to go to Google to find > answers that annoyed me but slowly posting preference adds up as w

Re: [asterisk-users] Top Posting

It was only the people who ONLY asked in a response to go to Google to find answers that annoyed me but slowly posting preference adds up as well. As long as the subject header is not changed all e-mail clients (no matter how stupid they are), now-a-days, create a nice tree. Even so does Hotmail,

Re: [asterisk-users] Why are 4 ports used for a single call?

On Fri, Jan 14, 2011 at 6:53 PM, Tom Rymes wrote: > On Jan 14, 2011, at 6:45 PM, Bruce B wrote: > > > You really want to read the LONG LONG signature from some people before > you read the actual latest message? I don't know about thatI guess it's > a preference.

[asterisk-users] Tools to Monitor Asterisk Servers and VMs

Hi Everyone, Are there any generally accepted and widely used tools made and tailored to be used for purpose of monitoring Asterisk servers? I am wondering if there is anything that the Asterisk community mostly uses or are there lots of manual scripts written and nothing really exists that every

Re: [asterisk-users] Why are 4 ports used for a single call?

are there recommendations of opening 5000-5082 UDP for SIP along with 5060 TCP? Are there any "niceties" to that as well? maybe video transmission stuff? Thanks On Fri, Jan 14, 2011 at 6:32 PM, Tom Rymes wrote: > > On Jan 14, 2011, at 5:24 PM, Bruce B wrote: > > > So, simply pressing

Re: [asterisk-users] Why are 4 ports used for a single call?

So, simply pressing Reply and typing in the first line (using gmail webmail without any clients) is a sin here? How is that top posting??? probably your clients reading that way? On Fri, Jan 14, 2011 at 5:13 PM, Tilghman Lesher wrote: > On Friday 14 January 2011 15:12:29 Bruce B wr

Re: [asterisk-users] Why are 4 ports used for a single call?

..@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B > *Sent:* Friday, January 14, 2011 2:15 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Why are 4 ports used for a single call? > > >

Re: [asterisk-users] Why are 4 ports used for a single call?

Also, to get the SIP very well as well, SIP uses both TCP/UDP 5060 right? and why are there recommendations of opening 5000-5082 UDP for SIP along with 5060 TCP? Are there any "niceties" to that as well? maybe video transmission stuff? Thanks again, On Fri, Jan 14, 2011 at 4:12 P

Re: [asterisk-users] Why are 4 ports used for a single call?

> > > > On Fri, Jan 14, 2011 at 12:44 PM, Bruce B wrote: > >> I mean part of RTP RFC? >> >> >> On Fri, Jan 14, 2011 at 2:41 PM, Bruce B wrote: >> >>> Hi Everyone, >>> >>> I am just tweaking a pfSense router and learning lo

Re: [asterisk-users] Why are 4 ports used for a single call?

I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B wrote: > Hi Everyone, > > I am just tweaking a pfSense router and learning lots about NAT etcI > noticed that each call uses four UDP port for RTP. Here is an example of > port for a call I made: > >

[asterisk-users] Why are 4 ports used for a single call?

Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so

Re: [asterisk-users] CallerID and URL pop up for windows...

works with Inbound and pulls up Outlook contact. Haven't tried outbound. On Fri, Jan 14, 2011 at 9:19 AM, Gilles wrote: > On Thu, 13 Jan 2011 17:59:10 -0500, Bruce B > wrote: > ><http://bestof.nerdvittles.com/applications/screenpop/>But better thing > >would be t

Re: [asterisk-users] CallerID and URL pop up for windows...

What you need already exists: http://bestof.nerdvittles.com/applications/screenpop/ But better thing would be to a have TAPI for outlook to query Outlook contact as well because it allows for making notes on the contact. I am willing to pay f

Re: [asterisk-users] OpenVPN + SIP configuration?

As I said, your tunnel address should be part of localnet. Otherwise you experience what you did. -Bruce On Thu, Jan 13, 2011 at 10:00 AM, Gilles wrote: > On Thu, 13 Jan 2011 15:55:10 +0100, Gilles > wrote: > >The only issue I notice, is that Asterisk doesn't tell the other end > >when the loc

Re: [asterisk-users] OpenVPN + SIP configuration?

In sip_nat.conf you need to specify 10.8.0.1/24 as your localnet and also make sure you have your externip setup as well. Else you will notice one way audio or cut off after 30 seconds. Rest of your work is all good. For security reasons the workstation that creates the keys is not connected to any

[asterisk-users] Paid or Free software that would do pop-up from Outlook 2007 via Asterisk AMI

Hi Everyone, I am looking for a paid version of a program that has proven to work with Outlook 2007 and Asterisk 1.6 on Windows Vista, XP, and maybe even Windows 7. Outcall is not the answer as it has lots of bugs and doesn't work. Something simple with very simple interface would be preferred.

Re: [asterisk-users] OpenVPN + SIP configuration?

Tue, 11 Jan 2011 10:23:18 -0500, Bruce B > wrote: > >I have OpenVPN and Asterisk working nicely. However, I do use > certificates. > >Though, it shouldn't matter. Can you explain what doesn't work for you? Is > >the connection not established or is the Ast

Re: [asterisk-users] Do I need a sip proxy?

/VoIP_Configuration > http://en.wikipedia.org/wiki/Application-level_gateway > > With kind regards, > Pan > > *From:* Bruce B > *Sent:* Tuesday, January 11, 2011 8:58 AM > *To:* Asterisk Users Mailing List - Non-Commercial > Discussion > *Subject:* [asterisk-users

Re: [asterisk-users] OpenVPN + SIP configuration?

Hi, I have OpenVPN and Asterisk working nicely. However, I do use certificates. Though, it shouldn't matter. Can you explain what doesn't work for you? Is the connection not established or is the Asterisk and it's client not communicating? -Bruce On Tue, Jan 11, 2011 at 9:20 AM, Gilles wrote:

[asterisk-users] Do I need a sip proxy?

Hi Everyone, I am running multiple instances of Asterisk in Proxmox and so far I had one central Asterisk feeding all others with trunks from one provider. Now, I want to connect each Asterisk server directly to the provider. Based on my understanding, each connection made to the provider port 506

Re: [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?

n Sat, Jan 8, 2011 at 11:27 AM, Steve Edwards wrote: > On Fri, 7 Jan 2011, Bruce B wrote: > > I want to know each and every parameter's detail that can be included in >> the >> >> read= >> write= >> >> in manager.conf >> >> Where can I fi

Re: [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?

Thanks Paul. That is exactly what I was looking for. On Sat, Jan 8, 2011 at 2:07 PM, Paul Belanger wrote: > On 11-01-07 01:33 PM, Bruce B wrote: > > Where can I find this? > > > manager.conf.sample? > > -- > Paul Belanger > Digium, Inc. | Software Develop

[asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?

Hi Everyone, I want to know each and every parameter's detail that can be included in the read= write= in manager.conf Where can I find this? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Ne

Re: [asterisk-users] Asterisk Outlook integration

com> wrote: > Hi BB, > > you could try this: > http://asterisk-outlook-dialer.voip-singapore.qarchive.org/ > > Never tested it deeply but apparently seems to work fine. > > Giorgio Incantalupo > > Bruce B wrote: > >> Hi Guys, >> >> What is ou

[asterisk-users] Asterisk Outlook integration

Hi Guys, What is out there other than OutCall that works with M$ Outlook and Asterisk 1.4/1.6 ? I prefer opensource and free (as in free in price) but can consider low price - working - programs as well. OutCall is giving issues with various versions of Outlook and it always brings up NEW CONTACT

[asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?

Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the

Re: [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?

Thanks a lot for that Sebastian. I will report back my findings when I find the resolution on this. Regards, On Mon, Jan 3, 2011 at 3:28 AM, Sebastian wrote: > Hi Bruce, > > > On 01/03/2011 06:03 AM, Bruce B wrote: > >> Thanks for the input. The errors I pointed out s

Re: [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?

Regards, On Sun, Jan 2, 2011 at 8:18 PM, Sebastian wrote: > Hi, > > > On 01/02/2011 08:08 PM, Bruce B wrote: > >> Thanks for the input. I have the latest drivers but it seems that there >> is some serious incompatibility issue with the kernel as when the >> FLASH

Re: [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?

Regards, Bruce On Sat, Jan 1, 2011 at 6:57 PM, Sebastian wrote: > Hi Bruce, > > > On 12/28/2010 10:51 PM, Bruce B wrote: > >> Thanks for the input. I can not replicate the situation as it happens >> randomely or maybe over the weekend. However I have sent you all the >&

Re: [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?

vo-ar1600/4505-3118_7-33777218.html Looking forward to your analysis. Regards, Bruce On Tue, Dec 28, 2010 at 3:58 PM, Moises Silva wrote: > On Tue, Dec 28, 2010 at 11:33 AM, Bruce B wrote: > >> >> I appreciate your feedback and let me know what info I can post here that >&g

[asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?

Hi Everyone, We are using two Sangoma U100 (USB FXO) units connected to an Acer Aspire Revo (little PC running on Atom). The units work beautifully except for Monday :-) It maybe a conincedence or maybe the fact that Saturday/Sunday is off and something happens where one of these U100 modules goe

Re: [asterisk-users] OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact

Thanks for feedback. I am looking mainly for pop-up of Outlook and don't need outgoing call at all but it would be nice to have. Regards, On Tue, Dec 28, 2010 at 4:01 AM, Stefan Schmidt wrote: > Am 28.12.10 07:26, schrieb Bruce B: > > Hi Everyone, > > > > I am using

[asterisk-users] OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact

Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and

Re: [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?

Thanks Kai-Uwe and everyone else. I have seen all those examples and I am exploring the sip_notify.conf file now which makes things more clear to me. However, when sending a SIP notify to a phone that is not registered to Asterisk via it's IP address should I expect to receive a success of fail pa

[asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?

Hi Everyone, I use Asterisk for regularPBX use it's made for. But I want to take it a bit further and use it at cmmand level to be able to send SIP notifies to restart a phone or take advantage of a phone's UPnP capabilities. Is Asterisk capable of that? If so, what is a simple SIP reboot message

Re: [asterisk-users] How to install the new cdr-stats?

Thanks for looking into it. Yes, it missed up and not worth looking at it. Unfortuantly, so are a few products from the same company (probably trying to make money of support which I understand)but it seems they released an install script which is here for CentOS: https://github.com/Star2Billi

Re: [asterisk-users] Asterisk hangs up call after 20s

This is a NAT issue like noted before. Try: localnet=192.168.0.0/ 255.255.255.0 instead of: localnet=192.168.0.0/24 Also, make sure you have all your VPN connections as localnet and other side subnet as localnet as well if you are using VPN. Otherwise

[asterisk-users] What is equivalent function to "mv" command in php for Asterisk Spool directory usage?

Hi Everyone, I understand that there are a few warnings about using "cp" to move .call files to /var/spool/asterisk/outgoing as they might acted upon before copy is done. So, using PHP, What is the equivalent of "mv" command? Would it be rename() in php or is there a better method? Thanks, -- ___

[asterisk-users] How to install the new cdr-stats?

Hi Everyone, I am trying to install the new cdr-stats from http://www.cdr-stats.org/ for Asterisk 1.6 but it's installation instructions are all garbled. It mentions both sqlite and mysql and there are no organized documentation. Not to mention that the apache port 8000 and port 9000 are also conf

Re: [asterisk-users] Wireless Desktop VoIP Phone?

Nortel 1535. Does video as well. On Fri, Dec 17, 2010 at 10:40 AM, Matt wrote: > I'm looking for a wireless desktop VoIP phone. Does any exist? > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com --

[asterisk-users] What to check for when there are sound interference using SIP channels only? standard debug methods?

Hi Everyone, I ocassionally hear echo, static, and garbled voice when calling extension to extension between two office (different geographic locations connected using OpenVPN - 1 with DSL and other with T1 - 1500 KM apart). I am guessing it's a bandwidth or jitter issue that is giving me faint pr

Re: [asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address?

010 at 11:45 AM, Bruce B wrote: > > Thanks for the feedback Ryan. > > Siproxd is not installed. I think Siproxd like you said just does the > > reverse meaning if phones are part of pfSense subnet then it connects to > > outside world. But in my case they are comin

Re: [asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address?

: 90 secs ** Regards, Bruce On Sat, Dec 11, 2010 at 10:15 AM, Ryan Wagoner wrote: > On Sat, Dec 11, 2010 at 3:06 AM, Bruce B wrote: > > Hi Everyone, > > I am using pfSense to do firewall and NAT on an Asterisk server. I have > &

Re: [asterisk-users] Why does "sip show peers" show myrouter/gateway address as the client IP address?

Hi Wang, Did you mean to write a feedback? You sent an empty message. Regards, On Sat, Dec 11, 2010 at 11:56 AM, wrote: > > Sent from my “contract free” BlackBerry® smartphone on the WIND network. > > -Original Message----- > From: Bruce B > Sender: a

Re: [asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address?

10:15 AM, Ryan Wagoner wrote: > On Sat, Dec 11, 2010 at 3:06 AM, Bruce B wrote: > > Hi Everyone, > > I am using pfSense to do firewall and NAT on an Asterisk server. I have > > ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local > IP > > 192

[asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address?

Hi Everyone, I am using pfSense to do firewall and NAT on an Asterisk server. I have ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local IP 192.168.5.5. However, when a user from outside using Linksys WRP400 ata connects to the Asterisk server and registers I see them as 192.

Re: [asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?

Thanks for the input guys. I really appreciate all the input and I am sure they work but I thought there would be a much better way to do this. Sounds like patching things to me. Why doesn't Asterisk take advantage of the qualify values to make sure if the SIP connection is up or not? Shouldn't thi

[asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?

Hi Everyone, There are situations when internet connection is lost, SIP provider fails, or even authentication to SIP provider fails, and we want to use the backup Dahdi channels (PSTN). As simple as it may sound but with the many different situations and error messages it seems like it's not so e

Re: [asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?

To be honest this is the first time I see this wiki mentioned. It doesn't even come up in talks on this list. The wiki should be advertised often and there should be some sort of active monitoring and supervision of the contents as well as some serious ongoing official contributions. All this well

[asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?

Hi Everyone, I am wondering why documentation of some of the vital parts of Asterisk is hosted on voipinfo.org (unreliable is some parts) and not on asterisk.org? For example the list of AMI events are not well documented and one has to guess which version supports which event. The documentation f

[asterisk-users] Using AMI to harvest / record HOLD time - Using FreePBX

Hi Everyone, I am looking into AMI (using PHP) to record every instance of HOLD that is generated by putting a caller on HOLD (press hold button on the phone set). There is no HOLD in Asterisk but the event Music on Hold is generated when HOLD is pressed. The complexity is that all of the the call

Re: [asterisk-users] Scheduled maintenance for various Asterisk community services

Seems like the servers are still not up? or at least the yum repository? Regards, On Fri, Nov 12, 2010 at 4:36 PM, Asterisk Development Team < asteriskt...@digium.com> wrote: > Between 10:00AM and 1:00PM CST on Saturday, November 13, the services > below will experience extended outages as the

[asterisk-users] eSXI and Asterisk?

Hi Everyone, I don't have much experience with eSXI. I can really use some advise on how to run it without any trouble with Asterisk on CentOS VMs. First of all, is it a good option to run multiple hosted Asterisk instances on a VMware eSXI? or would you rather prefer something like Xen, proxmox,

Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?

Thanks for input. Great info. Good to know all this about the router. I see you use a 256MB CF card there. Do you use a USB key stick for storage? Thanks, Bruce On Tue, Nov 9, 2010 at 4:09 AM, Gordon Henderson > wrote: > On Mon, 8 Nov 2010, Bruce B wrote: > > > Yes, it is a sma

Re: [asterisk-users] Is this a DDoS to reach Asterisk?

And that's the problem. There is no such service running or such port is not open. They only keep trying this for no reason. It might cost us bandwidth for no reason. In fact there is no open ports on our network whatsoever. Thanks On Mon, Nov 8, 2010 at 9:50 PM, Lyle Giese wrote: >

[asterisk-users] Is this a DDoS to reach Asterisk?

Hi Everyone, I have pfSense running which supplies Asterisk with DHCP. I had some testing ports opened for a web server which I have totally closed now but when I chose option 10 (filter log) on pfSense I get all of this type of traffic (note that it was only 1 single IP and once I blocked that on

Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?

. Thanks On Mon, Nov 8, 2010 at 7:24 PM, John Novack wrote: > > > Bruce B wrote: > > Thanks. I think I would still need a firewall. Maybe a 1u rack > double enclosure for two Alix boards - one as firewall - and one as PBX > would do better. > > Anyhow, I don't want to

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