Re: [asterisk-users] Server-to-server BLF

Hi Ronald I took a bit of interest in your problem as I'm going to have to be doing the same thing in a few weeks. oenais is in the yum repositories so you can install from there if using redhat/centos based OS It is also in apt repositories if you're using a debian based OS Let me know how you

Re: [asterisk-users] Problem connecting to 4569/UDP

iptables -L -n | grep icmp gives you the same on both machines? Is it possible that the other public IP is behind a "main" firewall, provided by your ISP? I know our hosting provider has this. They filter all traffic through their main router, and after that locally with iptables. On Tue, Jan 3,

Re: [asterisk-users] Set Call type in dial plan

I already tried what u posted didnt work but thanx for the reply :) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind [govoi...@gmail.com] Sent: Wednesday, January 04, 2012 11:32 PM

[asterisk-users] Connecting to an Old Phone System

Hi, This is not strictly an asterisk questions, but... ive got a client with an old digital pbx phone systems connected to an isdn30e line. I've been shown a sip gateway that can connect to asterisk on one side, and also has an ISDN30e socket that the old phone system can connect to. But it's a

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

All "screwing up with Asterisk" is supposed to be documented in the relevant UPGRADE*.txt files. Have you checked them? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Thursday, January 05, 2012

Re: [asterisk-users] Connecting to an Old Phone System

On Friday 06 January 2012, Dan Journo wrote: > Hi, > > This is not strictly an asterisk questions, but... ive got a client with an > old digital pbx phone systems connected to an isdn30e line. > > I've been shown a sip gateway that can connect to asterisk on one side, and > also has an ISDN30e so

Re: [asterisk-users] Anyone have a reliable T.38 Solution

David Backeberg writes: > Thanks for clearing that up. I was getting all excited that I could > flash the PAP2T; I've always used regular voice tones over SIP with > the PAP2Ts. SPA-2102 supports T.38. If you ignore the WAN-port, it is practically a PAP2T. The only time you cannot ignore the WAN

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

On Fri, Jan 6, 2012 at 8:16 AM, Eric Wieling wrote: > All "screwing up with Asterisk" is supposed to be documented in the > relevant UPGRADE*.txt files. Have you checked them? > > " is supposed to be" but does NOT happen. There are many examples of regressions introduced after many complains on

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

Correct, but the changes to the insecure= item is clearly documented in the UPGRADE-1.6.txt file * SIP: The old "insecure" options, deprecated in 1.4, have been removed. "insecure=very" should be changed to "insecure=port,invite" "insecure=yes" should be changed to "insecure=port" Be aware

Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)

Yes, I already declared 'use lib "/home/asterisk/lib/lib64/perl5/5.8.8/x86_64-linux-thread-multi/";' in my AGI. When I execute the script as a user Asterisk, i.e. perl -wc test.pl in return I'm getting OK and no error messages and script is running fine when I try to run in shell. Even though I al

Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)

Here's one more thing to try - do a "su - nobody" then run the agi - once it runs under "su - nobody" there's no reason it should run in Asterisk. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir Sent: Friday, January 06, 20

Re: [asterisk-users] Problem connecting to 4569/UDP

El día 6 de enero de 2012 06:00, Roland escribió: > iptables -L -n | grep icmp gives you the same on both machines? Yes. ACCEPT icmp -- 0.0.0.0/00.0.0.0/0 > > Is it possible that the other public IP is behind a "main" firewall, the ISP say to dont have any firewall. And the p

Re: [asterisk-users] Problem connecting to 4569/UDP

I "found" this on another post and "cleaned it up" - might help #!/usr/local/bin/perl use strict; use IO::Socket; my $target = shift; #"192.168.0.255"; my $target_port = 4569; socket(PING, PF_INET, SOCK_DGRAM, getprotobyname("udp")); # Build Packet ... # Names from ethereal filter of registr

[asterisk-users] Change port from 5060 on Snom phone

Hi I'm experimenting with using a port other than 5060 on one of our asterisk servers. Does anyone know how to change the target port on a Snom phone. I have tried adding : to the end of the registrar but this doesn't work. Advanced -> SIP/RTP -> Network identity(port) is something else before a

Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)

Add a BEGIN {...} block prior to the use statements and in there redirect STDERR to a file. This will aloow you to capture compilation errors You should also add some debugging statements at key points in the script. Then run the script and review the file to see what errors it generated. -- R

Re: [asterisk-users] Change port from 5060 on Snom phone

Interestingly enough, they just list port 5060 in the Asterisk interoperability guide: http://wiki.snom.com/Interoperability/PBX/Asterisk Could that mean it is a fixed setting? (crappy) *José Pablo Méndez * On Fri, Jan 6, 2012 at 10:17 AM, Ishfaq Malik wrote: > Hi > > I'm experiment

[asterisk-users] Streaming Music to 75 callers ..

Hi I want to play streaming music from an internet lp like http://114.23.245.234:9000 . there will be maximum of 75 callers at a time. Is combination of SetMusicon Hold and WaitMusiconHold command the best option ? All callers would be calling on PRI lines. The streaming source would be from a

[asterisk-users] Why write your dialplan using Lua?

Hello, Reading through the Wiki: "Asterisk supports the ability to write dialplan instructions in the Lua programming language. This method can be used as an alternative to or in combination with extensions.conf and/or AEL. PBX lua allows users to use the full power of lua to develop telephony ap

Re: [asterisk-users] Where are the fax instructions?

Kevin, I rolled back clean my virtual asterisk testbed and instead of libspandsp2 and libspandsp-dev, I installed libtiff4 - libtiff4-dev and downloaded/compiled spandsp 0.0.6, then I fixed the libraries with echo "/usr/local/lib" > /etc/ld.so.conf.d/spandsp.conf cuz it wasn't able to load the

Re: [asterisk-users] Where are the fax instructions?

On 01/06/2012 12:54 PM, José Pablo Méndez Soto wrote: Kevin, I rolled back clean my virtual asterisk testbed and instead of libspandsp2 and libspandsp-dev, I installed libtiff4 - libtiff4-dev and downloaded/compiled spandsp 0.0.6, then I fixed the libraries with echo "/usr/local/lib" > /etc/ld.

Re: [asterisk-users] Where are the fax instructions?

Ah ok, I got the incredible idea to go look into the make menuselect for a res_fax_spandsp option after reading this: http://lists.digium.com/pipermail/asterisk-dev/2010-September/046344.html I found it in 1.8, now you say it doesn't come with gateway support. Thanks for the clarification Kevin

Re: [asterisk-users] Where are the fax instructions?

On 01/06/2012 01:02 PM, José Pablo Méndez Soto wrote: Ah ok, I got the incredible idea to go look into the make menuselect for a res_fax_spandsp option after reading this: http://lists.digium.com/pipermail/asterisk-dev/2010-September/046344.html I found it in 1.8, now you say it doesn't come w

Re: [asterisk-users] Why write your dialplan using Lua?

On Fri, 6 Jan 2012, José Pablo Méndez Soto wrote: My question is, what is the benefit of using Lua? I've never used Lua, but I also have a curiosity about it. A couple of years ago, I wrote my first dialplan in AEL. Some bits were clumsy, minor syntax errors caused major parts of my dialplan

Re: [asterisk-users] Why write your dialplan using Lua?

I wouldn't jump to a whole different language just to have an elegant script plan. There must be another reason why Lua is being so widely implemented than elegance and execution performance. Anyone? *José Pablo Méndez * On Fri, Jan 6, 2012 at 1:12 PM, Steve Edwards wrote: > On Fri, 6

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

On 01/05/12 21:24, Joseph wrote: On 01/05/12 22:12, Bruce B wrote: but not it is not working again. I wish they stop screwing up with that Asterisk, they keep introducing new version and more bugs :-/ Wish not granted !!! :-) You will be the guinea pig to new features !!! Same i

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

AFAIK insecure=very has been replaced by insecure=port,invite. Also, you might want to put wait(2) or progress after answer in the dialplan to allow CID to process. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf

Re: [asterisk-users] Why write your dialplan using Lua?

On Fri, 6 Jan 2012, José Pablo Méndez Soto wrote: I wouldn't jump to a whole different language just to have an elegant script plan. I would and probably will. Maybe it's just me, but I look at a dialplan filled with database access, conditional gotos and priority labels and my eyes just glaz

Re: [asterisk-users] Why write your dialplan using Lua?

Not to "purposely" open a flame war here, but is lua preferable to ael? Aren't we better off just properly documenting our original conf files since new changes often introduce "bad opportunities"? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun

Re: [asterisk-users] Why write your dialplan using Lua?

On Fri, 6 Jan 2012, Danny Nicholas wrote: Not to "purposely" open a flame war here, but is lua preferable to ael? Aren't we better off just properly documenting our original conf files since new changes often introduce "bad opportunities"? One of the advantages of Lua is that it is embedded i

Re: [asterisk-users] Anyone have a reliable T.38 Solution

On Fri, Jan 6, 2012 at 10:01 AM, Benny Amorsen wrote: > David Backeberg writes: > > > Thanks for clearing that up. I was getting all excited that I could > > flash the PAP2T; I've always used regular voice tones over SIP with > > the PAP2Ts. > > SPA-2102 supports T.38. If you ignore the WAN-port,

[asterisk-users] best softphone for 2012?

Just installed asterisknow 1.6. I can access freepbx. I need to test system on my LAN. Which softphone is best to use? I'm running ubuntu on Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX for incoming/outgoing calls. No video. Tom -- ___

Re: [asterisk-users] best softphone for 2012?

On Sat, Jan 7, 2012 at 12:00 AM, Tom Poe wrote: > Just installed asterisknow 1.6. I can access freepbx. I need to test > system on my LAN. Which softphone is best to use? I'm running ubuntu on > Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX for > incoming/outgoing call

Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)

On 01/06/2012 10:30 AM, Ron Bergin wrote: Add a BEGIN {...} block prior to the use statements and in there redirect STDERR to a file. This will aloow you to capture compilation errors You should also add some debugging statements at key points in the script. Then run the script and review the

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

On 01/06/12 14:44, Danny Nicholas wrote: AFAIK insecure=very has been replaced by insecure=port,invite. Also, you might want to put wait(2) or progress after answer in the dialplan to allow CID to process. I've tried putting even "wait(7) it didn't help. The problem is I'm getting this error:

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

Check your sip.conf and users.conf - my guess is that the pstn-1270 is an assigned value that you need to remove or comment out. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Friday, January 06, 2

Re: [asterisk-users] best softphone for 2012?

On 01/06/2012 04:03 PM, Ross Cameron wrote: On Sat, Jan 7, 2012 at 12:00 AM, Tom Poe > wrote: Just installed asterisknow 1.6. I can access freepbx. I need to test system on my LAN. Which softphone is best to use? I'm running ubuntu on Dell optiplex G260

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

On Fri, 6 Jan 2012, Danny Nicholas wrote: AFAIK insecure=very has been replaced by insecure=port,invite. Also, you might want to put wait(2) or progress after answer in the dialplan to allow CID to process. Is this an issue with SIP? I thought either it was in the INVITE or it wasn't. --

Re: [asterisk-users] Why write your dialplan using Lua?

Ok so its not a cosmetic thing only. I eases your administration. Do a point for performance. Now, what about my questions regarding extending the systems caps by building things asterisk could not build by itself. does it hold true? On Jan 6, 2012 3:28 PM, "Steve Edwards" wrote: > On Fri, 6 Ja

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

On 01/06/12 16:49, Danny Nicholas wrote: Check your sip.conf and users.conf - my guess is that the pstn-1270 is an assigned value that you need to remove or comment out. What do you mean "assigned value" My user.conf: [general] fullname = New User userbase = 6000 hasvoicemail = yes vmsecret =

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

Putting in a Wait(n) is only (sometimes) needed to wait for the CallerID NAME on PRI or BRI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, January 06, 2012 6:06 PM To: Asterisk U

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

On 01/06/12 18:15, Eric Wieling wrote: Putting in a Wait(n) is only (sometimes) needed to wait for the CallerID NAME on PRI or BRI. Putting wait(5) in my dial plan doesn't work as I'm getting this error message: WARNING[2344]: chan_sip.c:13930 check_auth: username mismatch, have <11>, digest

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

As I said, the Wait is only useful for PRI and BRI. It is TOTALLY USELESS for SIP and IAX. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Friday, January 06, 2012 6:35 PM To: Asterisk Users Mail

Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

On 01/06/12 16:35, Joseph wrote: On 01/06/12 18:15, Eric Wieling wrote: Putting in a Wait(n) is only (sometimes) needed to wait for the CallerID NAME on PRI or BRI. Putting wait(5) in my dial plan doesn't work as I'm getting this error message: WARNING[2344]: chan_sip.c:13930 check_auth: use

Re: [asterisk-users] Speech recognition in asterisk using google voice API

Does sox have more features on a Debian system than RHEL? Is that why it won't work on RHEL? Cheers, On Wed, Jan 4, 2012 at 6:42 PM, Lefteris Zafiris wrote: > Fresh code is out! The use of sox can be now optionally enabled by the > user if the system has a recent version of the program (won't w

Re: [asterisk-users] Speech recognition in asterisk using google voice API

On Fri, 6 Jan 2012 20:46:14 -0500 Bruce B wrote: > Does sox have more features on a Debian system than RHEL? Is that why > it won't work on RHEL? > RHEL's 5 version of sox is really old and outdated. The command syntax and the switches are totally different compared to recent versions of sox. An

Re: [asterisk-users] best softphone for 2012?

On Saturday 07 Jan 2012, Tom Poe wrote: > Just installed asterisknow 1.6. I can access freepbx. I need to > test system on my LAN. Which softphone is best to use? I'm running > ubuntu on Dell optiplex G260 desktop at home. I'm hoping to setup > basic IP PBX for incoming/outgoing calls. No vid

Re: [asterisk-users] Speech recognition in asterisk using google voice API

Thanks. I have been testing Aastra phones with SIP and had great results. I am testing my cell phone now and sometimes get "-1" for id, status, utterance, and confidence. What does that mean? Cheers On Fri, Jan 6, 2012 at 9:40 PM, Lefteris Zafiris wrote: > On Fri, 6 Jan 2012 20:46:14 -0500 > B

Re: [asterisk-users] Speech recognition in asterisk using google voice API

NVM. I explored the code and see the logic. I had sox = 1 so it was failing on RHEL. To report, my cell phone from a PRI gets same confidence level just like SIP. Building my control app now. Should make my life much easier while driving. Thanks again :-) -Bruce On Fri, Jan 6, 2012 at 10:50 PM,

[asterisk-users] Asterisk 10.0 & 1.4 - iax codec are not compatible

I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk 10.0 is no better. I'm still getting: WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have <11>, digest has NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device "KMIEC Z

Re: [asterisk-users] Speech recognition in asterisk using google voice API

Added two new features to the script: Timeout value and speechdata type. *exten => s,n,agi(speech-recog.agi,en-US,3000,phoneNumb)* - Will listen for 3 seconds and sanitize return as a single number without any spaces in between. This helps when one reads phone number in format 415-554-2323 and goo