Re: [asterisk-users] Can we use MySQL native connector for ARA?

H, yeah I understand. Maybe with some direction I can go through each realtime modules (peers/friends, extenstions,moh etc...), and use the native database connecters, and some kind of factory pattern (if even possible in C), to support the different databases, maintaining what was possible usi

Re: [asterisk-users] Can we use MySQL native connector for ARA?

> our own agi application. Can we use the native database connectors for > ARA. I currently have > everything working with unixodbc + myodbc however, looking to use the > native DB connector > if possible. Traditionally, the mysql realtime backend was buggy and crash-prone. The odbc backend is ha

Re: [asterisk-users] Can we use MySQL native connector for ARA?

Hello Danny, Thank you for your response. Actually, what I was referring to is Asterisk Realtime, and not our own agi application. Can we use the native database connectors for ARA. I currently have everything working with unixodbc + myodbc however, looking to use the native DB connector if possib

Re: [asterisk-users] Can we use MySQL native connector for ARA?

1. odbc has been successful for some posters 2. I would personally use System or AGI to handle my MYSQL stuff so you have clean "bash-like" handling. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis S

[asterisk-users] Can we use MySQL native connector for ARA?

Hello Everyone, The documentation suggests using unixodbc for asterisk realtime. Is there any way we can just use native database clients such as libmysqlclient from MySQL? The native clients tend to be more up-to-date. Thanks in Advance, Nick. -- ___

[asterisk-users] Scheduled Maintenance for Asterisk Project community services

On Thursday, October 20th, 2011, the Asterisk community services listed below will be undergoing maintenance (software upgrades and updates). The services will be shut down at approximately 9:00 PM CDT (2:00 AM October 21st UTC), and will return no later than 10:00 PM CDT. We apologize in advance

Re: [asterisk-users] G729 and Dahdi: Inbound forcing ulaw!

> Upgrade to 1.8.7.1 There was a bug fixed recently (I think in 1.8.6, > but might have been 1.8.7) which caused Asterisk to sometimes not > transcode when it should. A regression introduced in v1.8.7 broke the ability of the ./configure script to generate the HAVE_PRI_xxx defines for ISDN. Fix co

Re: [asterisk-users] G729 and Dahdi: Inbound forcing ulaw!

Upgrade to 1.8.7.1 There was a bug fixed recently (I think in 1.8.6, but might have been 1.8.7) which caused Asterisk to sometimes not transcode when it should. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf

[asterisk-users] Asterisk replying 491

Hallo, any idea what's wrong with that invite?? help would be greatly appreciated! thanks Markus U XX.199.123.185:5060 -> XX.189.169.66:5060 INVITE sip:07111234567@XX.189.169.66 SIP/2.0..Via: SIP/2.0/UDP 192.168.178.26:5060;rport;branch=z9hG4bK98099..Max-Forwards: 70..To: @XX.189.169.66

Re: [asterisk-users] DTMF fun

Tom Browning writes: > My question is this: Is Asterisk simply relaying the client's DTMF > signalling untouched or do I need to look at Asterisk more > closely and turn some knobs. I would recommend that you grab some wireshark traces before and after the DTMF traverses Asterisk. It should be

Re: [asterisk-users] Problem E1 PRI

On Wed, Oct 19, 2011 at 05:25:44PM -0200, Sebastian wrote: > > I'm having problems with a new ISDN PRI in a new server. > The cable is connected and the E1 modem seems to have issues with > syncing (blinking light on the modem). [snip] > dahdi show status > > T4XXP (PCI) Card 0 Span 1

[asterisk-users] G729 and Dahdi: Inbound forcing ulaw!

Happy day to ya list! I've recently deployed an update on our Asterisk server, taking it from 1.2 up to 1.8.5 - going from zaptel to dahdi. Excitement levels are high and performance so far is wonderful. The issue I have is I've purchased several G729 licenses, registered them, installed the mod

[asterisk-users] Problem E1 PRI

Hi, I'm having problems with a new ISDN PRI in a new server. The cable is connected and the E1 modem seems to have issues with syncing (blinking light on the modem). versions: CentOS 6, asterisk 1.6.2.20, dahdi 2.5.0.1, libpri 1.4.12 --

Re: [asterisk-users] Asterisk call transfers not working

Hi Danny, Warren: This is what I found in extensions_additional.conf: [from-internal-additional] include => from-internal-additional-custom include => app-dialvm include => app-vmmain include => app-recordings include => app-callwaiting-cwoff include => app-callwaiting-cwon include => ext-group i

Re: [asterisk-users] How to use menuselect.makeopts?

On 11-10-19 12:50 PM, Patrick Lists wrote: On 10/19/2011 03:08 PM, Jason Parker wrote: On 10/18/2011 09:52 PM, Luke Hamburg wrote: I think this might actually be a bug. https://issues.asterisk.org/jira/browse/ASTERISK-18137 It is indeed a bug, but it's not the bug you referenced. This issue on

Re: [asterisk-users] Asterisk sponteanous reboot : core dump file

> What does this mean ? What can I do further ? https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introdu

Re: [asterisk-users] strange delay behaviour in SIP call with same codec

> Hello, > I have a strange audio delay behaviour when placing a call between two > SIP devices using the same codec. > In my example, I have two devices forced to use GSM codec. > When placing a call, the first ~9sec have no audio, then the audio > starts trasmitting. > If I force one phone to use

Re: [asterisk-users] Asterisk call transfers not working

On Wed, Oct 19, 2011 at 11:28 AM, Danny Nicholas wrote: > Or you could just add these lines to [from-internal-xfer] > > Exten => _X,1,Dial(SIP/${EXTEN},30,iKkTtt) > > Exten => _XX,1,Dial(SIP/${EXTEN},30,iKkTt) > > ** ** > > If you have 3 or 4 digit extensions you would need these lines***

Re: [asterisk-users] How to use menuselect.makeopts?

On 10/19/2011 03:08 PM, Jason Parker wrote: On 10/18/2011 09:52 PM, Luke Hamburg wrote: I think this might actually be a bug. https://issues.asterisk.org/jira/browse/ASTERISK-18137 It is indeed a bug, but it's not the bug you referenced. This issue only exists in 1.8.8.0-rc1. It has been fixed

Re: [asterisk-users] Asterisk call transfers not working

Now I need to see what is in [from-internal-custom] and [from-internal-additional] Or you could just add these lines to [from-internal-xfer] Exten => _X,1,Dial(SIP/${EXTEN},30,iKkTtt) Exten => _XX,1,Dial(SIP/${EXTEN},30,iKkTt) If you have 3 or 4 digit extensions you would need these lines

Re: [asterisk-users] Asterisk call transfers not working

Hi Danny: Thanks for your response. [from-internal-xfer] include => from-internal-custom include => from-internal-additional ; auto-generated exten => s,1,Macro(hangupcall) exten => h,1,Macro(hangupcall) I have to tell you that we use Freepbx 2.9. I hope you can help me to solve this issue. Let'

Re: [asterisk-users] Problem with video phone call, error in sdp media handling?

Hi, thanks for Your quick response. But as You can see in the commented SIP-Messages, asterisk gets a voice call and sends out a INVITE with two media attributes for video and voice towards the destination. Karsten Am Mittwoch, den 19.10.2011, 10:40 -0500 schrieb Danny Nicholas: > Just a WAG -

Re: [asterisk-users] Problem with video phone call, error in sdp media handling?

Just a WAG - if you start the call in voice-mode, the video codecs aren't loaded. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten Wemheuer Sent: Wednesday, October 19, 2011 10:37 AM To: asterisk-users@l

Re: [asterisk-users] Asterisk call transfers not working

What does your context [from-internal-xfer] look like? (it should either resemble or have an include for [default] context). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ramiro Paz Sent: Wednesday, October 19, 2011 10:33 AM To: aste

[asterisk-users] Problem with video phone call, error in sdp media handling?

Hi, I try to setup a video call and I sometimes get no video. I set up a Yealink VP 2009 and a Ninja Softphone. Both a in the same LAN. Asterisk release is 1.8.7.0. Call from Yealink to the Ninja is working fine, if I start the call in video mode. In this case I can switch between voice-only and

[asterisk-users] Asterisk call transfers not working

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears

Re: [asterisk-users] DID and how the caller id will appear

On Tue, Oct 18, 2011 at 8:46 PM, bilal ghayyad wrote: > Dear; > > By the way, the asterisk version that I have is 1.8.4.2 and DAHDI version is > 2.4.1.2 > > Here I would like to mention the following: > > 1) As per the telecom provider, they said they openned for us all the digits > to send (two

Re: [asterisk-users] Any help with these error messages???

> [trunkgroups] > > [channels] > > [my-phones](!) > usecallerid = yes > hidecallerid = no > callwaiting = yes > usecallingpres = yes > callwaitingcallerid = yes > threewaycalling = yes > transfer = yes > canpark = yes > cancallforward = yes > callreturn = yes > echocancel = yes > echocancelwhenbr

Re: [asterisk-users] Running as non-root

On 11-10-19 05:50 AM, Torbjörn Abrahamsson wrote: Hello. I would like to run asterisk as an user other than root. I have seen some tutorials on the web, but I would like to know if there is some “official” how-to for this. Is there? I looked at a thread on reviewboard regarding this (https://re

Re: [asterisk-users] How to use menuselect.makeopts?

On 10/19/2011 03:08 PM, Jason Parker wrote: On 10/18/2011 09:52 PM, Luke Hamburg wrote: I think this might actually be a bug. https://issues.asterisk.org/jira/browse/ASTERISK-18137 It is indeed a bug, but it's not the bug you referenced. This issue only exists in 1.8.8.0-rc1. It has been fixed

Re: [asterisk-users] Running as non-root

On Wed, Oct 19, 2011 at 7:19 AM, Torbjörn Abrahamsson wrote: > Thank you, I actually found the asterisk.conf settings after sending the > mail. So next question is which folders/files do I need to change ownership > of to make it work? > > > > /etc/asterisk > > /var/lib/asterisk > > /usr/lib/aster

[asterisk-users] DTMF fun

I'm chasing down some DTMF interop issues would like to hopefully rule out Asterisk in the following configuration: RTP path is: Linux/PC/Mac SIP clients -> [MediaProxy as needed] -> Asterisk 1.8.7 -> SIP termination provider(s) DTMF is strictly RFC2833 with no in-band. Asterisk stays in the med

Re: [asterisk-users] DID and how the caller id will appear

The callerid= option in chan_dahdi.conf is normally used on FXO or FXS ports, not for PRI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta Sent: Wednesday, October 19, 2011 9:21 AM To: Asterisk U

Re: [asterisk-users] DID and how the caller id will appear

Dear, Callerid you need to add parameter in chan_dahdi.conf file. so what is you chan_dahdi.conf file ? Best Regards, Mahesh On Wed, Oct 19, 2011 at 6:46 PM, Eric Wieling wrote: > CallerID is your country code + city/area code + telephone number. Do not > set the leading 0, that is not part

Re: [asterisk-users] DID and how the caller id will appear

CallerID is your country code + city/area code + telephone number. Do not set the leading 0, that is not part of the Caller*ID. Example London UK number, country code 44, area code 20, number 1234-5678: Set(CALLERID(num)=442012345678 -Original Message- From: asterisk-users-boun...@lis

Re: [asterisk-users] How to use menuselect.makeopts?

On 10/18/2011 09:52 PM, Luke Hamburg wrote: I think this might actually be a bug. https://issues.asterisk.org/jira/browse/ASTERISK-18137 It is indeed a bug, but it's not the bug you referenced. This issue only exists in 1.8.8.0-rc1. It has been fixed for 1.8.8.0-rc2 which will be released thi

[asterisk-users] strange delay behaviour in SIP call with same codec

Hello, I have a strange audio delay behaviour when placing a call between two SIP devices using the same codec. In my example, I have two devices forced to use GSM codec. When placing a call, the first ~9sec have no audio, then the audio starts trasmitting. If I force one phone to use GSM and the

Re: [asterisk-users] Problems during calls

Aksel, i faced a similar issue with remote sip extensions. and seems to be happening due to internet problems. one way audio that is .. one of the parties (on site) stops hearing the other party. and it happens with one extension at a random timing and random extension.. and if all extensions

Re: [asterisk-users] nvfaxdetect in 10.0

I used to use NV FAX but since version 1.8 I found this not require. Did u used NV for version 10? On Tue, Oct 18, 2011 at 11:37 PM, Danny Nicholas wrote: > +1 for you Andrew - easiest fix I've had for Asterisk in a while. > > -Original Message- > From: asterisk-users-boun...@lists.dig

Re: [asterisk-users] GoogleTalk Calls

Thank you Vladimir but this patch not applicable in Asterisk 1.6.2.20. if (!strcasecmp(name, "error") && -(redirect = iks_find_cdata(traversenodes, "redirect")) && (redirect = strstr(redirect, "xmpp:"))) { redirect += 5; ast_debug(1, "redirect %s\n", redirect); This block not ex

[asterisk-users] Asterisk sponteanous reboot : core dump file

Hello, when I try to get something out of the core dump file, I get this : [root@jonas Desktop]# gdb asterisk core.sip1.server.be GNU gdb (GDB) Fedora (7.2-51.fc14) Copyright (C) 2010 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later

Re: [asterisk-users] Running as non-root

Thank you, I actually found the asterisk.conf settings after sending the mail. So next question is which folders/files do I need to change ownership of to make it work? /etc/asterisk /var/lib/asterisk /usr/lib/asterisk /var/spool/asterisk /var/log/asterisk And the files in them of cours

[asterisk-users] Detecting Special Information Tone in Asterisk

Hi, Has anybody any idea about detecting Special Information Tone(SIT) when making utbound calls? http://en.wikipedia.org/wiki/Special_information_tone I googled for detecting SIT in Asterisk but couldn't find useful results. Thanks, --Sam -- __

Re: [asterisk-users] How to use menuselect.makeopts?

On 10/19/2011 04:52 AM, Luke Hamburg wrote: I think this might actually be a bug. https://issues.asterisk.org/jira/browse/ASTERISK-18137 Thank you very much for pointing that out Luke. Seems I bumped into the same bug. Regards, Patrick --

Re: [asterisk-users] voicemail

thanks a lot for your help i will try to do that best regards 2011/10/19 Sammy Govind > 1- Are you sure your Asterisk Box is configured with an MTA / email utility > to send emails ? > 2- Like Ishfaq suggested you should be getting into the voicemail > application after 10 seconds of Dial timeo

Re: [asterisk-users] Running as non-root

What do you use _now_ to run asterisk: safe_asterisk, init-script, from command-line? What distribution do you use? To run asterisk from command line as user "asterisk", just run asterisk -U asterisk (asterisk user should be created), or edit /etc/asterisk.conf to run as user "asterisk". In product

[asterisk-users] Running as non-root

Hello. I would like to run asterisk as an user other than root. I have seen some tutorials on the web, but I would like to know if there is some “official” how-to for this. Is there? I looked at a thread on reviewboard regarding this (https://reviewboard.asterisk.org/r/654/). It was Paul B

Re: [asterisk-users] Outgoing call failure

Hi all, My issue was resolved. It was an issue with service provider. I got help from some smart guys, they have helped me a lot to setup my PRI up and running. Thank you very much. Sample link may helpful to identify the simmilar issues : http://networking.ringofsaturn.com/Routers/is

Re: [asterisk-users] Problems during calls

Thank you for replying also, I will as you and Zeeshan suggest, look at the firewall issue first, i have been suspecting network issue, because i cannot see anything in the log, so again thanks! Best regards Aksel Fra: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] voicemail

1- Are you sure your Asterisk Box is configured with an MTA / email utility to send emails ? 2- Like Ishfaq suggested you should be getting into the voicemail application after 10 seconds of Dial timeout. Are you even recording and saving a voicemail? 3- To recieve an SMS to notify you of voicemail