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

2010-07-08 Thread bruce bruce
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] Not detecting hangup

2010-07-08 Thread Julian Lyndon-Smith
That looks like the option that will help a lot. Thanks. On 8 July 2010 23:21, Steve Edwards wrote: >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian >> Lyndon-Smith >> >> We have had 20 calls over the last month where the SIP channel has not >> identified that the person o

Re: [asterisk-users] This may be a problem.. Answer not working on 1.4.32 over SIP trunk..

2010-07-08 Thread Massimo Nuvoli
Zeeshan Zakaria ha scritto: > I have two test asterisk boxes, both version 1.4.26, on which I do > Answer() followed by MusicOnHold() and it works just fine. I do this all > the time as this is my standard way of testing new contexts. Yesterday i tested another installation and i found the same is

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

2010-07-08 Thread bruce bruce
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, bruce bruce wrote

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

2010-07-08 Thread bruce bruce
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?

2010-07-08 Thread Steve Edwards
On Thu, 8 Jul 2010, bruce bruce wrote: > I have this in /etc/my.cnf: > > [mysqld] > datadir=/var/lib/mysql > socket=/var/lib/mysql/mysql.sock > user=mysql > old_passwords=1 > log-error=/var/log/mysqld.log > > [mysqld_safe] > log-error=/var/log/mysqld.log > pid-file=/var/run/mysqld/mysqld.pid > l

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

2010-07-08 Thread bruce bruce
I have this in /etc/my.cnf: [mysqld] datadir=/var/lib/mysql socket=/var/lib/mysql/mysql.sock user=mysql # Default to using old password format for compatibility with mysql 3.x # clients (those using the mysqlclient10 compatibility package). old_passwords=1 log-error=/var/log/mysqld.log [mysqld_

Re: [asterisk-users] Not detecting hangup

2010-07-08 Thread Steve Edwards
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian > Lyndon-Smith > > We have had 20 calls over the last month where the SIP channel has not > identified that the person on the receiving end has hung up. > > Is there a way of fixing this ? On Thu, 8 Jul 2010, Danny Nicholas w

Re: [asterisk-users] DTMF issues/redial tones with rfc2833

2010-07-08 Thread das sandesh
Thanks Zeeshan.that server is located at the headquaters and phones are at different locations, even with default rfc2833 mode, other party IVR prompts was not able to detect the tones, also 'Info' works good but not with internal options like voicemail, etc. And inband is not being used as we

Re: [asterisk-users] Not detecting hangup

2010-07-08 Thread Danny Nicholas
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Thursday, July 08, 2010 4:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Not detecting hang

[asterisk-users] Not detecting hangup

2010-07-08 Thread Julian Lyndon-Smith
We have had 20 calls over the last month where the SIP channel has not identified that the person on the receiving end has hung up. Is there a way of fixing this ? TIA Julian -- _ -- Bandwidth and Colocation Provided by http:/

Re: [asterisk-users] DTMF issues/redial tones with rfc2833

2010-07-08 Thread Zeeshan Zakaria
>From what you explained, it seems obvious that there exists some non-SIP device somewhere in your communication path, and since voice is picked up as DTMF, some device is also set to listen for inband DTMF. What is the origination source of incoming calls to your system? Zeeshan A Zakaria -- ww

Re: [asterisk-users] DTMF issues/redial tones with rfc2833

2010-07-08 Thread Alex Bell
Sandesh, Review the bug logs in particular ID: 0017571..there's a recent patch that may apply to your issue. http://issues.asterisk.org Thanks, Al On Thu, Jul 8, 2010 at 4:18 PM, das sandesh wrote: > Hi, > > We have few systems with asterisk 1.4.22.1 and we use sip trunking

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

2010-07-08 Thread bruce bruce
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 --

[asterisk-users] DTMF issues/redial tones with rfc2833

2010-07-08 Thread das sandesh
Hi, We have few systems with asterisk 1.4.22.1 and we use sip trunking for them not PRI's, one of our system is giving a problem of dtmf (rfc2833), like when we dial the number that have IVR and enter the extension or access code, it some time takes it and some times does'nt recognize the digits d

Re: [asterisk-users] Junghanns QuadBRi not really recognized in Dahdi

2010-07-08 Thread Tzafrir Cohen
On Thu, Jul 08, 2010 at 07:11:47PM +0200, Olivier wrote: > 2010/7/8 Tzafrir Cohen > > > On Thu, Jul 08, 2010 at 05:48:53PM +0200, Olivier wrote: > > > # lsmod | grep hfc > > > hfc4s8s_l1 11716 0 > > > hisax 397700 1 hfc4s8s_l1 > > > > I suspect you need to blacklist

Re: [asterisk-users] Level3 reseller needed

2010-07-08 Thread Matt Desbiens
VoIPInnovations from what I understand is pretty good, haven't dealt much with them though. Worth a call and an interop. --Matt Desbiens //EOF On Thu, Jul 8, 2010 at 3:33 PM, Adam Moffett wrote: > I'm in the Northeast US and looking for any recommendations on Level3 > resellers. I don't do eno

[asterisk-users] Level3 reseller needed

2010-07-08 Thread Adam Moffett
I'm in the Northeast US and looking for any recommendations on Level3 resellers. I don't do enough volume to go to Level3 directly. If there's anybody you'd definitely avoid I'd love to hear about that too. Thanks, Adam -- _

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

2010-07-08 Thread Dan Austin
Manmohan wrote: > I was looking for audio conferencing solution where i got Web-meetme. > I had installed Asterisk 1.6.2.9 on Centos  5.4. Its perfecting working > fine. I tried using Meetme even meetme app is working perfectly fine. > I installed Webmeetme 4.0 and integrated with my asterisk. When

Re: [asterisk-users] Externnotify on pollmailboxes=yes

2010-07-08 Thread Tilghman Lesher
Please respond after the text which you are quoting and trim footers. On Thursday 08 July 2010 13:23:18 Eric Hiller wrote: > Tilghman Lesher wrote: > > On Tuesday 06 July 2010 22:51:03 Eric Hiller wrote: > > > Tilghman Lesher wrote: > > > > On Monday 05 July 2010 19:17:00 Eric Hiller wrote: > > >

Re: [asterisk-users] Externnotify on pollmailboxes=yes

2010-07-08 Thread Eric Hiller
Yes, I understand this part, but then what is the purpose of pollmailboxes if not to trigger mwi? > From: tles...@digium.com > To: asterisk-users@lists.digium.com > Date: Wed, 7 Jul 2010 01:23:34 -0500 > Subject: Re: [asterisk-users] Externnotify on pollmailboxes=yes > > On Tuesday 06 July 2010

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

2010-07-08 Thread Adil Zaaraoui
Yes i agree; ok here the output of verbosity at level 3: -- Executing [00212664800...@pstn2:1] GotoIf("SIP/100-081e3648", "0?internal:external") in new stack -- Goto (pstn2,00212664800450,2) -- Executing [00212664800...@pstn2:2] Dial("SIP/100-081e3648", "SIP/lo...@pstn2/011212664800450

Re: [asterisk-users] Junghanns QuadBRi not really recognized in Dahdi

2010-07-08 Thread Olivier
2010/7/8 Tzafrir Cohen > On Thu, Jul 08, 2010 at 05:48:53PM +0200, Olivier wrote: > > Hi, > > > > I'm using Asterisk 1.6.1.18 with latest Dahdi Linux and Tools (rev 8854) > and > > libpri 1.4.10.2. > > > > During installation, I'm facing this : > > # dahdi_genconf -v system > > Default parameters

Re: [asterisk-users] Asterisk + Hylafax + Iiaxmodem - Outbound number.

2010-07-08 Thread Doug Lytle
Marta Silva wrote: > Hi All, > I am trying to costumise the out number (not just the CallerID), for > faxes sent from a specific sip client a fax machine connected to a This was recently discussed on the iaxmodem mailing list. You can read up on it here: http://sourceforge.net/mailarchive/for

[asterisk-users] Incoming call doesn't finish when internal phone hangs up

2010-07-08 Thread Daniel - Asterisk
Hello guys, I have this problem when a call is received in my PBX: (Caller) --> (Redirecting Service) --> (E1 PRI) --> (Asterisk PBX) --> (Internal Phone) Reception works fine, but when conversation finishes and the agent at internal phone hangs up, the call at caller's side is still alive for m

Re: [asterisk-users] Junghanns QuadBRi not really recognized in Dahdi

2010-07-08 Thread Tzafrir Cohen
On Thu, Jul 08, 2010 at 05:48:53PM +0200, Olivier wrote: > Hi, > > I'm using Asterisk 1.6.1.18 with latest Dahdi Linux and Tools (rev 8854) and > libpri 1.4.10.2. > > During installation, I'm facing this : > # dahdi_genconf -v system > Default parameters from /etc/dahdi/genconf_parameters > Empty

Re: [asterisk-users] Can't dial out through AMI

2010-07-08 Thread Mike Ely
On 7/8/10 5:07 AM, "Paul Belanger" wrote: > On Wed, Jul 7, 2010 at 2:46 PM, Mike Ely wrote: >> Maybe I missed something here?  SIP users configured within Asterisk can >> dial out just fine through the trunk.  It's just when I try to use AMI that >> it fails. >> > The far end is rejecting your

Re: [asterisk-users] not sure what to change to point the timing to the at&t circuits?

2010-07-08 Thread Jonathan Thurman
On Thu, Jul 8, 2010 at 8:30 AM, Jared Terrell wrote: > # Span 1 > span=1,1,0,esf,b8zs > bchan=1-23 > dchan=24 > echocanceller=mg2,1-23 > # Span 2 > span=2,2,0,esf,b8zs > bchan=25-47 > dchan=48 > echocanceller=mg2,25-47 > # Span 3 > span=3,3,0,esf,b8zs > bchan=49-71 > dchan=72 > echocanceller=mg2,4

[asterisk-users] connecting to EON millenium getting 405 message

2010-07-08 Thread Jerry Geis
I am trying to connect asterisk to EON millenium with SIP. I am getting a 405 message back from EON Looks like asterisk sends a register command , EON says OK then asterisk sends sip options and EON says 405. Is there a way not to send sip options? THanks, Jerry -- __

[asterisk-users] Asterisk + Hylafax + Iiaxmodem - Outbound number.

2010-07-08 Thread Marta Silva
Hi All, I am trying to costumise the out number (not just the CallerID), for faxes sent from a specific sip client a fax machine connected to a Grandstream GXW-4004 V1.3B (I believe it uses the default number for my PRI). Can it be done? Thank you very much in advance. Regards, - Marta -- _

Re: [asterisk-users] Update the LCD with the callee's name after dialing

2010-07-08 Thread Jonas Kellens
Hello, using asterisk 1.4.30. No patch or anything else. I just do the following in dialplan : exten => 20,n,SIPAddHeader(Remote-Party-ID: "Testing" ) When my Cisco calls my Grandstream, the name "Testing" appears on the screen of my Cisco. When my Grandstream calls my Cisco, only "20" app

[asterisk-users] Junghanns QuadBRi not really recognized in Dahdi

2010-07-08 Thread Olivier
Hi, I'm using Asterisk 1.6.1.18 with latest Dahdi Linux and Tools (rev 8854) and libpri 1.4.10.2. During installation, I'm facing this : # dahdi_genconf -v system Default parameters from /etc/dahdi/genconf_parameters Empty configuration -- no spans Of course, I've got span descriptions in /etc/d

[asterisk-users] not sure what to change to point the timing to the at&t circuits?

2010-07-08 Thread Jared Terrell
# Span 1 span=1,1,0,esf,b8zs bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2 span=2,2,0,esf,b8zs bchan=25-47 dchan=48 echocanceller=mg2,25-47 # Span 3 span=3,3,0,esf,b8zs bchan=49-71 dchan=72 echocanceller=mg2,49-71 # Span 4 span=4,4,0,esf,b8zs bchan=73-95 dchan=96 echocanceller=mg2,73-95 # Glo

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

2010-07-08 Thread Dave Fullerton
On 07/08/2010 10:19 AM, Zeeshan Zakaria wrote: > That's why I specifically mentioned Cat5 networks, because giga bit networks > which use four pairs are called Cat6 networks. > > This is true that Cat5 networks are also used with gigabit hardware, but > technically it is wrong. Cat6 hardware uses d

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

2010-07-08 Thread Zeeshan Zakaria
That's why I specifically mentioned Cat5 networks, because giga bit networks which use four pairs are called Cat6 networks. This is true that Cat5 networks are also used with gigabit hardware, but technically it is wrong. Cat6 hardware uses different frequencies over copper than Cat5, and mixing a

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

2010-07-08 Thread Benny Amorsen
Zeeshan Zakaria writes: > making use of the fact that both Cat5 networks and BRI ports > don't use all the 8 pins, so why not use extra wires in the cable for > something useful instead of wasting them. For Ethernet, this is only true for 10Mbps and 100Mbps. Gigabit and up uses all four pairs.

[asterisk-users] How to integrate thirdparty RTP with Asterisk

2010-07-08 Thread garge rama
Hi, I'm working on Asterisk and would like to use only Asterisk SIP signalling for my Voip application. I have written my own channel driver and want to integrate my own RTP with Asterisk. SIP signalling is working fine. But i could not find API's to get RTP Port and IP address to start witho

[asterisk-users] call deflection support in chan_dahdi, libpri

2010-07-08 Thread Wolfgang Pichler
Hi all, i do have the following setup ISDN BRI Line -> openVOX Card/Asterisk 1.6.2.6/libpri 1.4.11.2 -> Dialplan Dial DAHDI -> ISDN PBX -> ISDN Equipment The user on the ISDN Equipment das enable call forwarding - Teilrufumleitung / Call deflection - so that call will get forwarded by the telco

[asterisk-users] Recordings in the bank.

2010-07-08 Thread Rodrigo Lang
Hello list. I've been researching if there is a way of putting the recordings of Mixmonitor in database (PostgreSQL or MySQL) in an automated way. I've read that the native has voicemail in Asterisk via ODBC. And for the MixMonitor has some way? Someone on the list have it implemented? Thanks, R

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

2010-07-08 Thread Paul Belanger
On Thu, Jul 8, 2010 at 2:51 AM, Manmohan Singh Jandu wrote: > crashes giving segmentation fault. > Read doc/backtrace.txt on how to capture and generate a backtrace. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.

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

2010-07-08 Thread Paul Belanger
On Thu, Jul 8, 2010 at 6:29 AM, Adil Zaaraoui wrote: > But it does not work. > Any suggestion > Without posting a debug log it makes it hard to troubleshoot. http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt -- Paul Belanger | dCAP Polybeacon | Consultant Jabber:

Re: [asterisk-users] Can't dial out through AMI

2010-07-08 Thread Paul Belanger
On Wed, Jul 7, 2010 at 2:46 PM, Mike Ely wrote: > Maybe I missed something here?  SIP users configured within Asterisk can > dial out just fine through the trunk.  It's just when I try to use AMI that > it fails. > The far end is rejecting your call; SIP/2.0 401 Unauthorized. If you can dialout w

[asterisk-users] AGI get full variable

2010-07-08 Thread velusamy Krishnan
Dear All, I have "get full variable" AGI call to get the ANSWEREDTIME channel variable. I have originated the call to one extension, once answered I have called DeadAGI to control the call. I have problem that after hangup the call AGI "GET FULL VARIABLE" returns -1 for ANSWEREDTIME channel va

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

2010-07-08 Thread Adil Zaaraoui
Hello, Thank you for your reply. Ok it works; now i have an other issue: My asterisk(A) communicate with an other asterisk(B) via SIP protocole, my user (U) communicate with asterisk(A) via IAX2 protocole; i tried as you said but unfortunatly it does not work; here is my configuration : sip.

[asterisk-users] Problem with call-limit

2010-07-08 Thread Jonas Kellens
Hello list, asterisk 1.4.30 2 situations in which call-limit should work, but it does not : [Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The device state of this queue member, test12, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct

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

2010-07-08 Thread Chandrakant Solanki
On Thu, Jul 8, 2010 at 12:21 PM, Manmohan Singh Jandu wrote: > Hello Team, > > I was looking for audio conferencing solution where i got Web-meetme. > I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working > fine. I tried using Meetme even meetme app is working perfectly fine. > I