Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

Ahh here is the catch i was still using app_cbmysql for this. now i had removed and just followed the README of 4.0 for WMM and m getting following on ,my asterisk console. Verbosity is at least 3 == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] MeetMe("SIP/492-",

Re: [asterisk-users] False answer() being sent by cellphone providers

> From: asterisk-users-boun...@lists.digium.com on behalf of Steve Edwards > > Continuing to veer off-topic... > > Maybe you should re-read Atlas Shrugged. On Fri, 9 Jul 2010, Mike Ely wrote: > And no thanks: I've already read that execrable book, and found it to be > nothing more than overwro

[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] False answer() being sent by cellphone providers

-Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Steve Edwards Sent: Fri 7/9/2010 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:Re: [asterisk-users] False answer() being sent by cellphone providers On Fri,

Re: [asterisk-users] False answer() being sent by cellphone providers

On Fri, 9 Jul 2010, Mike Ely wrote: > (off list) Continuing to veer off-topic... > Yes indeed we do. The telcos here are absolutely abhorrent, to the > point that much could be written about how horrible they are but nobody > would want to read such depressing material. And consumer protecti

Re: [asterisk-users] False answer() being sent by cellphone providers

On 7/9/10 3:20 PM, "Gordon Henderson" wrote: > On Fri, 9 Jul 2010, Mike Ely wrote: > >> On 7/9/10 9:57 AM, "Mike Ely" wrote: >> >>> Hello, list. >>> >>> I've set up an outbound alerting system to play a recording when systems go >>> down, etc. and I'm noticing that cellphones tend to answer()

Re: [asterisk-users] False answer() being sent by cellphone providers

On Fri, 9 Jul 2010, Mike Ely wrote: On 7/9/10 9:57 AM, "Mike Ely" wrote: Hello, list. I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified thi

Re: [asterisk-users] Pbx_för_Windows?_-_Email_f ound_ in_subject

Hi all, Many thanks for your replies! Will tell my friend and see what he will be interested in. Many thanks! Christian -Ursprungligt meddelande- Från: mgra...@mstvp.com Till: "Asterisk Users Mailing List - Non-Commercial Discussion" Skickat: 10-07-09 15:29 Ämne: Re: [asterisk-users] Pbx

Re: [asterisk-users] Logging codec used in CDR

Hi! > Is there a way to log the negotiated codec that was used for each call > in CDR or in a separate log file? Use CHANNEL(audionativeformat) - and do the same with the help of the M option to Dial() for the remote call leg. Store that info in the CDR "userfield", or create your own field if

Re: [asterisk-users] Problem with call-limit

I have the same problem, I have asterisk 1.4.21.2. I have limitonpeer = yes in context general, call-limit=10 in all peers, but still have this message in Cli. 2010/7/8 Jonas Kellens > Hello list, > > asterisk 1.4.30 > > 2 situations in which call-limit should work, but it does not : > > [Ju

[asterisk-users] Logging codec used in CDR

Happy Friday everyone, Is there a way to log the negotiated codec that was used for each call in CDR or in a separate log file? This is for SIP-based calls, if that matters. Perhaps there is some variable that can be queried as part of the dialing script; Or is it possible to grab the codec name

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

I guess it has to be on the Trunk and one of the either user or peer and the opposing party shouldn't have it as no. But, to full proof urself, put it on the trunk and both users. Basically put it anywhere that takes it. http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite

Re: [asterisk-users] False answer() being sent by cellphone providers

Thanks for the tip! On 7/9/10 11:35 AM, "Faisal Hanif" wrote: >Do some R & D with asterisk function AMD (Answering Machine Detection) if > that can help you. > > Signatures fai...@vopium.com > > Regards, > > Faisal Hanif > > > > On 7/9/2010 11:24 PM, Danny Nicholas wrote: >>

Re: [asterisk-users] [Dahdi 2.3.0.1] Does it need all those modules?

On 07/09/2010 01:41 PM, Gilles wrote: > 3. rebooted, and checked that netjet was gone and wctdm was in: > == > # lsmod | grep -i wc > wctc4xxp 32414 0 > dahdi_transcode 5751 1 wctc4xxp > wcb4xxp33905 0 > wcfxo 8968 0 > wctdm

Re: [asterisk-users] [Dahdi 2.3.0.1] Does it need all those modules?

I just went through a Dahdi rebuild, and I seem to recall a message that all modules will be loaded until you set up the dahdi configuration files. regards Scott On 7/9/2010 11:41 AM, Gilles wrote: > Hello > > To use Dahdi + Asterisk with a PCI card with a single FXO port, I > just... > > 1. co

Re: [asterisk-users] [NAT] * + private IP + locked-down firewalls?

On Fri, 9 Jul 2010 08:06:04 -0400, Ryan Wagoner wrote: >I have around 50 Snom 370s configured this way. They work great for >remote workers. However the Snom speakerphone is terrible compared to >Aastra and Polycom. If there is any background noise it will cut in >and out the other party. Thanks

[asterisk-users] [Dahdi 2.3.0.1] Does it need all those modules?

Hello To use Dahdi + Asterisk with a PCI card with a single FXO port, I just... 1. compiled and installed Dahdi 2. edited /etc/modprobe.d/dahdi.blacklist.conf to blacklist "netjet" and unblacklist "wctdm": == # cat /etc/modprobe.d/dahdi.blacklist.conf blacklist wct4xxp blacklist wct

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

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 To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Fri, Jul 9, 2010 8:13 pm Subject: Re: [asterisk-users] General network ques

Re: [asterisk-users] False answer() being sent by cellphone providers

Do some R & D with asterisk function AMD (Answering Machine Detection) if that can help you. Regards, Faisal Hanif On 7/9/2010 11:24 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf

Re: [asterisk-users] False answer() being sent by cellphone providers

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, July 09, 2010 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] False answer() being

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] power outage

On Fri, Jul 9, 2010 at 12:42 PM, Jerry Geis wrote: > and show status gives me condition RED of course. > Physical problem, check cables / telco -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _

Re: [asterisk-users] False answer() being sent by cellphone providers

On 07/09/2010 12:33 PM, Mike Ely wrote: > On 7/9/10 10:29 AM, "Mike Ely" wrote: > >> Some of the systems blokes might just figure that¹s another collections agent >> and hang up then ;) >> >> >> On 7/9/10 10:09 AM, "Steve Edwards" wrote: >> >>> On Fri, 9 Jul 2010, Mike Ely wrote: >>> > I've

Re: [asterisk-users]

On Fri, Jul 9, 2010 at 12:57 PM, Mike Ely wrote: > Has anyone figured out how to detect the actual cellphone answer rather than > the bogus one sent by the cell carrier? > *CLI> core show application AMD -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC:

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

Hi all, i have a beginners question. How are SIP calls and IAX2 calls processed by Asterisk over the network? What i mean is, is there a permanent connection required between the Asterisk Server and the clients or is the Asterisk Server only involved for lets call it the "routing"? >From my un

Re: [asterisk-users] False answer() being sent by cellphone providers

On 7/9/10 10:29 AM, "Mike Ely" wrote: > Some of the systems blokes might just figure that¹s another collections agent > and hang up then ;) > > > On 7/9/10 10:09 AM, "Steve Edwards" wrote: > >> On Fri, 9 Jul 2010, Mike Ely wrote: >> I've set up an outbound alerting system to play a reco

Re: [asterisk-users] False answer() being sent by cellphone providers

Some of the systems blokes might just figure that¹s another collections agent and hang up then ;) On 7/9/10 10:09 AM, "Steve Edwards" wrote: > On Fri, 9 Jul 2010, Mike Ely wrote: > >>> >> I've set up an outbound alerting system to play a recording when systems >>> go >>> >> down, etc. and I'm

Re: [asterisk-users] power outage

Jerry Geis wrote: > and show status gives me condition RED of course. > What's the output of pri show span 1? Check your cable. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- __

[asterisk-users] chan_iax2: I should never be called!

Hi, Recently, one of my Asterisk servers stopped connecting calls and required a reboot to "fix it" (did not try to restart or reload). The log showed loads of this message: NOTICE[302] chan_iax2.c: I should never be called! This highly repeated message seems to be preceded by something like:

Re: [asterisk-users] False answer() being sent by cellphone providers

On Fri, 9 Jul 2010, Mike Ely wrote: I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (t

[asterisk-users] False answer() being sent by cellphone providers

On 7/9/10 9:57 AM, "Mike Ely" wrote: > Hello, list. > > I've set up an outbound alerting system to play a recording when systems go > down, etc. and I'm noticing that cellphones tend to answer() and then start > ringing the actual handset. So far, I've verified this behavior with > Verizon, T-M

[asterisk-users]

Hello, list. I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOU

[asterisk-users] Delay between answer and pickup ?

We are having a situation on our dialler here where our agents are claiming that when they receive a call because it has been answered, it seems as if the call had been answered several seconds earlier - IOW, they are hearing "hello ? Hello ?" and often hear the phone being put down as an initial p

Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

Manmohan wrote: > My Web-MeetMe_v4.0.1, i followed the instructions in the > README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. >> Are you using RealTime enabled app_meetme or app_cbmysql >> from the

[asterisk-users] power outage

I have a TE205P that has been working fine for 2 years. power outage yesterday took out my everything for over an hour. Everything has come back up except the PRI. My provider has checked it to the box and says everything looks good on their end. I get this message: [Jul 9 12:40:32] WARNING[137

Re: [asterisk-users] Call failed: 408 timeout

Hi, Please disable firewall and SElinux. 2010/7/9 Philipp von Klitzing > > SEND >> 0.0.0.100:5060 > > ?! > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live

Re: [asterisk-users] Re : Re : Re : Communication IAX2 >SIP>IAX2

Glad you found the issue, sorry for not being able to help. 2010/7/9 Paul Belanger > On Fri, Jul 9, 2010 at 7:38 AM, Adil Zaaraoui > wrote: > > ok it works i had a problem with a syntax: > > i had to wrire: > > exten =>_!X.,n(external),Dial(SIP/011212664800...@pstn2,,S(20)) > > > Correct, > > D

Re: [asterisk-users] VoIP Users Conference Recordings

/R, It's guest # 3 on the call 2day. Sorry but where exactly are we on the call? I can't seem to find the website you are demoing. Help! On Sat, Jul 3, 2010 at 3:09 AM, Randy R wrote: > Hi, > > Alistair Cunningham of Integrics was our guest yesterday. We talked > about Integrics new product

Re: [asterisk-users] Call failed: 408 timeout

> SEND >> 0.0.0.100:5060 ?! -- _ -- 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 asteris

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 CyberData VoIP amplifier and t

Re: [asterisk-users] Call failed: 408 timeout

Hello: Here is my sip and extentions configuration and the log of x-lite, because i don`t can call inside my LAN with asterisk PBX 1.2 and i don`t have NAT. i hope you can help me. SIP.conf [default] include=>anexos include=>anexos1 include=>anexos2 [anexos] exten=> 100,1,Dial(SIP/100,0) exten=

Re: [asterisk-users] Pbx_för_Windows?_-_Email_f ound_ in_subject

I echo the sentiment that you should just run Asterisk on some small hardwarein an appliance like fashion. In fact, just yesterday I posted an overview of hardware suitable for DIY appliances. I've used many of the platforms mentioned. http://www.mjgraves.com/2010/07/08/d-i-y-asterisk-applianc

Re: [asterisk-users] Pbx för Windows? - Email found in subject

Arjan Kroon | Mobillion wrote: > Mayby Freepbx. > http://www.freepbx.org/ > And, as their page states, "FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk" Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a litt

Re: [asterisk-users] Pbx för Windows?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: Friday, July 09, 2010 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pbx för Windows? O

Re: [asterisk-users] Pbx för Windows?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Friday, July 09, 2010 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pbx för Windows? On Frida

Re: [asterisk-users] Pbx för Windows?

On Friday 09 Jul 2010, Christian wrote: > Hi all, > Yes, this is not the right list for such a question and I am using Asterisk > myself its for a friend who isn't used to Linux. You can write me off list > if you want. He is looking for a Windows based PBX with same functionality > as Asterisk. An

Re: [asterisk-users] Pbx för Windows?

On Fri, 9 Jul 2010, Christian wrote: > Hi all, > Yes, this is not the right list for such a question and I am using > Asterisk myself its for a friend who isn't used to Linux. You can write > me off list if you want. He is looking for a Windows based PBX with same > functionality as Asterisk.

Re: [asterisk-users] Pbx för Windows?

Christian a écrit : > Hi all, > Yes, this is not the right list for such a question and I am using Asterisk > myself its for a friend who isn't used to Linux. You can write me off list if > you want. > He is looking for a Windows based PBX with same functionality as Asterisk. > Any tips? > Many

Re: [asterisk-users] Pbx för Windows? - Email found in subject

Mayby Freepbx. http://www.freepbx.org/ Regards, Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Christian Verzonden: 09-07-2010 14:41 Aan: asterisk-users@lists.digium.com Onderwerp: [asterisk-users

Re: [asterisk-users] asterisk and cisco 2800

If you do back to back, then one end needs to clock. To set it on the Cisco, type "clock source internal" under the controller config. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: F

Re: [asterisk-users] Pbx för Windows?

YATE & FreeSWITCH available on windows. Asterisk can be build for windows using cygwin. There are some PBX software also available on windows but with some limitation. Signatures fai...@vopium.com Regards, Fai

[asterisk-users] Pbx för Windows?

Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Many thanks! -- _

Re: [asterisk-users] Re : Re : Re : Communication IAX2 >SIP>IAX2

On Fri, Jul 9, 2010 at 7:38 AM, Adil Zaaraoui wrote: > ok it works i had a problem with a syntax: > i had to wrire: > exten =>_!X.,n(external),Dial(SIP/011212664800...@pstn2,,S(20)) > Correct, Dial(SIP/lo...@pstn2/011212664800450,,S(20)) Is not valid syntax -- Paul Belanger | dCAP Polybeacon |

[asterisk-users] Sip Proxy

Can i make build Proxy server by asterisk -- Best Regards M.D -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http:/

Re: [asterisk-users] [NAT] * + private IP + locked-down firewalls?

On Fri, Jul 9, 2010 at 4:28 AM, Gilles wrote: > On Mon, 05 Jul 2010 12:45:34 +0200, Gilles > wrote: >>Provided the user doesn't have access to the firewall (eg. corporate >>or hotel), and the firewall doesn't allow dynamic port opening through >>UPnP or NAT-PMP... > > For those interested, I was

[asterisk-users] Re : Re : Re : Communication IAX2 >SIP>IAX2

ok it works i had a problem with a syntax: i had to wrire: exten =>_!X.,n(external),Dial(SIP/011212664800...@pstn2,,S(20)) thanks De : Adil Zaaraoui À : Asterisk Users Mailing List - Non-Commercial Discussion Envoyé le : Jeu 8 juillet 2010, 19h 41min 15s Obj

Re: [asterisk-users] AGI get full variable

Dear All, Please anyone help me to solve the following problem. Thanks, Velusamy On Thu, Jul 8, 2010 at 4:19 PM, velusamy Krishnan wrote: > Dear All, >I have "get full variable" AGI call to get the ANSWEREDTIME channel > variable. I have originated the call to one extension, once answer

Re: [asterisk-users] [NAT] * + private IP + locked-down firewalls?

Hi! > >Provided the user doesn't have access to the firewall (eg. corporate or > >hotel), and the firewall doesn't allow dynamic port opening through UPnP > >or NAT-PMP... > > For those interested, I was tipped through private e-mail about using > OpenVPN to open a steady tunnel between the clien

Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

Hi, Following is what i did. [r...@linuxtest ~]# yum install mysql* Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * addons: centos.skknet.net * base: centos.skknet.net * extras: centos.skknet.net * updates: centos.skknet.net Setting up Install Process Package m

Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

It still crashes and in gdb trace following is what its showing: --More-- warning: .dynamic section for "/usr/lib/mysql/libmysqlclient.so.15" is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations [New Thread 13310] LAST FEW LINES IN GDB TRACE

Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

Hi Install mysql 'n mysql-devel which includes /usr/lib/mysql/libmysqlclient.so.15 library. And also insert /usr/lib/mysql into /etc/ld.so.conf and then execute ldconfig command on terminal. -- Regards, Chandrakant Solanki On Fri, Jul 9, 2010 at 2:32 PM, Manmohan Singh Jandu wrote: > Hi, >

Re: [asterisk-users] asterisk and cisco 2800

Hi Peder, it seems to work, thank you! Now I've got a problem with the cisco 2800 which is resetting every 5 minutes but I do not think it is related to the cable, maybe something about the clock but except for a wiki page (http://www.voip-info.org/wiki/view/Asterisk+legacy+integration) there

Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

Hi, My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. >Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? > i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try

[asterisk-users] Click2call from an OpenOffice document

Hi, What would you suggest to get click2call from an OpenOffice document ? For instance, in OOo Writer, there is a block : M. John Doe Tel: +1 234 567 890 email: j...@example.com Looking at this block, the line +1 234 567 890 is underlined. When clicking on this, a contextual menu pops up allowi

Re: [asterisk-users] [NAT] * + private IP + locked-down firewalls?

On Mon, 05 Jul 2010 12:45:34 +0200, Gilles wrote: >Provided the user doesn't have access to the firewall (eg. corporate >or hotel), and the firewall doesn't allow dynamic port opening through >UPnP or NAT-PMP... For those interested, I was tipped through private e-mail about using OpenVPN to open

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

bruce bruce ha scritto: > 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