[asterisk-users] Which 1.6 subversion is Stable one?

Hi all, I would like to install asterisk as my home pbx, Anyone can suggest which sub version of 1.6 is stable? Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

[asterisk-users] Configure Asterisk with openssl

on TLS. Thanks Nikhil -- _ -- 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 asterisk-users

Re: [asterisk-users] Configure Asterisk with openssl

On 09/16/2010 04:11 PM, A J Stiles wrote: On Thursday 16 Sep 2010, Nikhil wrote: Hi I got the bellow error when I try to configure asterisk code. $./configure --with-ssl=/usr/local/ssl ... ... ... checking for mandatory modules: OPENSSL... fail configure: *** configure

Re: [asterisk-users] Configure Asterisk with openssl

Its working now when I installed openssl using yum. yum install -y openssl-devel. Thanks Nikhil On 09/16/2010 05:26 PM, Nikhil wrote: On 09/16/2010 04:11 PM, A J Stiles wrote: On Thursday 16 Sep 2010, Nikhil wrote: Hi I got the bellow error when I try to configure asterisk

[asterisk-users] Cross compile Asterisk for mipsel-linux

Hi Anyone knows how to do cross compile asterisk 1.6.2.13 using mipsel linux.? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Implementing more than one asterisk instance in the same hardware machine?

/sbin/asterisk -g for first asterisk $ /home/asterisk2/usr/sbin/asterisk -g for second asterisk Thanks Nikhil On 10/05/2010 03:42 PM, bilal ghayyad wrote: Hi All; Did anyone try to implement (installation and configuration and running) for more than one asterisk instance (two or three instances

[asterisk-users] Radius client support

Hi Will radius client in asterisk can use with third party radius servers instead of freeradius ?,if supports how do I configure asterisk to make it work. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] CDR Billing issues

parking , Anyone have a solution on this. And anyone knows how to use accoutcode and amaflags to solve this problem. Thanks Nikhil. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

[asterisk-users] usage of account code in CDR

Hi everyone Anyone please explain me How Account code is use for billing., Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

[asterisk-users] Is existing CDR in Asterisk is enough for complete billing

on this. Thanks Nikhil -- _ -- 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 asterisk-users

Re: [asterisk-users] How to register SIP phone on Asterisk 1.6.2.14 on Centos 5.5 64bit

Check X-lite sending register request or not to asterisk buy checking the asterisk console,if not there would some problem in X-lite configuration settings,if sending check the console and see what error logs you are getting.. Thanks Nikhil On 11/18/2010 04:06 PM, Phuong Hoang wrote: Hi

Re: [asterisk-users] usage of account code in CDR

please reply on this if u know On 11/18/2010 09:24 AM, Nikhil wrote: Hi everyone Anyone please explain me How Account code is use for billing., Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Transfered Number is local extension of PSTN

Hi everyone Does anyone know how to check the TRANSFERRED Target Number is a local extension or a PSTN number. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] Is existing CDR in Asterisk is enough for complete billing

anyone have a idea on this.. On 11/22/2010 10:50 AM, Nikhil wrote: Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough

Re: [asterisk-users] Is existing CDR in Asterisk is enough for complete billing

Thanks,Now I understand the problem,Now I am trying to change CDR to fix these issues. Thanks Nikhil On 12/01/2010 08:31 PM, Steve Murphy wrote: On Wed, Dec 1, 2010 at 5:56 AM, Nikhil d.nik...@cem-solutions.net mailto:d.nik...@cem-solutions.net wrote: anyone have a idea

[asterisk-users] Caller id is not proper when I do call forward

of call forward it will show properly.Please correct me I am wrong Thanks NIkhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] Linkedid member in Channel structure on 1.8

(means copy paste the logic).Does this works or I need to do anything else.waiting for reply to start change the code. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

[asterisk-users] Callee side blind transfer is failing in 1.8

HI callee side blind transfer is failed in 1.8 but caller side blind transfer is succes,Transfer doing by refer method,please help me on this Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] How to find , internal, external inbound or outbound

Hi Does anyone knows how to find out a call in a asterisk is external incoming ,external out going or internal Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] How to find , internal, external inbound or outbound

reply please On 12/17/2010 10:03 AM, Nikhil wrote: Hi Does anyone knows how to find out a call in a asterisk is external incoming ,external out going or internal Thanks Nikhil -- _ -- Bandwidth and Colocation

[asterisk-users] Ported Asterisk in Android

Hi Does anyone ported Asterisk to Android OS .please give details Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Ported Asterisk in Android

reply please.. On 12/17/2010 03:51 PM, Nikhil wrote: Hi Does anyone ported Asterisk to Android OS .please give details Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] DIALSTATUS on CANCEL

Hi Enable debug level to more than 1 ,you may get something. Thanks Nikhil On 12/22/2010 11:26 AM, Michael wrote: Spawn extension (incoming-private, , 3) exited non-zero on 'SIP/Proxy-0031' -- _ -- Bandwidth

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

SIPp is a good option. Thanks Nikhil On 12/27/2010 11:38 PM, Bruce B wrote: 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

[asterisk-users] OPTIONS Packet is retransmitting continuously

HI In My asterisk OPTIONS packet is retransmitting continuously ,any one know the reason for this.I am using asterisk 1.6.1.1. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] Why Local Channels are creating

this is using..? Thanks Nikhil -- _ -- 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 asterisk

[asterisk-users] how to get Current Calls details

,and scenarios.Please help me on this. Thanks Nikhil -- _ -- 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] how to get Current Calls details

Any suggestions .. On 02/02/2011 09:04 AM, Nikhil wrote: Hi everyone How can I get the current calls details in asterisk.if I use cli commad core show channels,there is two channels of each call.But the requirement is, need to get caller ,calee,starttime ,duration of the current

Re: [asterisk-users] how to get Current Calls details

Hi Thorsten Thanks of reply. The command core show verbose is working. but the problem is, for one call we can see 2 results,there is no common field on these two.How I parse these result to get the proper values.(Caller,callee,starttime,duration). Thanks nikhil On 02/02/2011 03:11 PM

Re: [asterisk-users] how to get Current Calls details

. Thanks Nikhil On 02/02/2011 03:58 PM, Ishfaq Malik wrote: On Wed, 2011-02-02 at 09:04 +0530, Nikhil wrote: Hi everyone How can I get the current calls details in asterisk.if I use cli commad core show channels,there is two channels of each call.But the requirement is, need to get caller

[asterisk-users] Radius Based Accounting for Asterisk

Hi everyone Any one used Radius based accounting for asterisk.Please give me details. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Ported Asterisk in Android

Thanks for reply. Any other suggestions . On 12/20/2010 05:52 PM, Service clients - VDI CENTER wrote: i believe there is a way to do it using asterisk and flashphoner ++ 2010/12/20 Gilles codecompl...@free.fr mailto:codecompl...@free.fr On Fri, 17 Dec 2010 15:51:33 +0530, Nikhil

Re: [asterisk-users] Ported Asterisk in Android

waiting for replys.. On 02/11/2011 02:20 PM, Nikhil wrote: Thanks for reply. Any other suggestions . On 12/20/2010 05:52 PM, Service clients - VDI CENTER wrote: i believe there is a way to do it using asterisk and flashphoner ++ 2010/12/20 Gilles codecompl...@free.fr

[asterisk-users] Asterisk Using as a SIP Client

Hi I wanted to use asterisk as SIP client in my centOS box.I should able to make calls and receive calls.and should able to talk and listen from the headset that I connected to my CentOS box. I need a direction to start on this. Thanks Nikhil

Re: [asterisk-users] Asterisk Using as a SIP Client

Do I need to modify chan_phone application to make it works or it is available in net. Thanks Nikhil On 02/17/2011 12:52 PM, Khaled W. Chehab wrote: Install asterisknow and begin from there. http://www.asterisk.org/asterisknow/ and don’t miss to read the documentation https

[asterisk-users] wav files are not playing asterisk

Hi I try to play a wav file in asterisk ,but its accepting only gsm files.Do u know where I need to change to make it works. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] wav files are not playing asterisk

Any reply.. On 03/01/2011 02:50 PM, Nikhil wrote: Hi I try to play a wav file in asterisk ,but its accepting only gsm files.Do u know where I need to change to make it works. Thanks Nikhil -- _ -- Bandwidth and Colocation

Re: [asterisk-users] wav files are not playing asterisk

sory satish...my thunderbrid was not load. Thanks for reply... On 03/02/2011 09:59 AM, Nikhil wrote: Any reply.. On 03/01/2011 02:50 PM, Nikhil wrote: Hi I try to play a wav file in asterisk ,but its accepting only gsm files.Do u know where I need to change to make it works. Thanks Nikhil

Re: [asterisk-users] wav files are not playing asterisk

) -- Executing [#@default:1] Playback(ALSA/hw:0,0, demo-thanks) in new stack -- ALSA/hw:0,0 Playing 'demo-thanks' (language 'en') -- Executing [#@default:2] Hangup(ALSA/hw:0,0, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'ALSA/hw:0,0' Thanks Nikhil On 03/01/2011 07:55

[asterisk-users] How to send Hold invite from asterisk to other

Hi all how to send SIP HOLD Invite from asterisk to other sip client/server.? Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] How to send Hold invite from asterisk to other

ok..that means I have to modify chan_sip . I wondering why this is not available in asterisk. Thanks Nikhil. On 03/16/2011 04:39 AM, Kevin P. Fleming wrote: On 03/15/2011 04:18 AM, Nikhil wrote: how to send SIP HOLD Invite from asterisk to other sip client/server.? Asterisk's chan_sip

[asterisk-users] Usage of lock in CDR

Hi all In asterisk source code we can see lots of places AST_CDR_FLAG_LOCKED flags is used.This is for CDR purpose. Does anyone what is exact usage of this lock in CDR.If I remove this flags where it will impact,any data overwrite will happen..? Thanks Nikhil

Re: [asterisk-users] Usage of lock in CDR

document which explain how CDR works in asterisk and what are its limitations. Thanks Nikhil On 03/22/2011 11:35 AM, Tilghman Lesher wrote: On Tuesday 22 March 2011 00:56:05 Nikhil wrote: Hi all In asterisk source code we can see lots of places AST_CDR_FLAG_LOCKED flags is used.This

Re: [asterisk-users] Usage of lock in CDR

Thanks .. which version of asterisk have the CEL completely. Now I am using 1.6.2 . Thanks Nikhil On 03/22/2011 03:31 PM, Tilghman Lesher wrote: On Tuesday 22 March 2011 02:20:24 Nikhil wrote: Thanks for reply. I am trying to understand how CDR in asterisk is working(Code wise),because some

[asterisk-users] Asterisk using as a SIP client

Hi all I am planning to use asterisk as a IP phone(Porting asterisk into a hardware).Is there any limitations if I use asterisk as a SIP client?,and asterisk has any advantages if use like this? Thanks Nikhil

[asterisk-users] Compiling asterisk using NDK build

Hi all, Does anyone compiled asterisk using NKD build in android. Please give some suggestions. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] Make Multiple Calls using Chan_alsa module

Hi I would like to use asterisk as a SIP client(IP phohone ) with multiple user,multiple line support . Using existing chan_alsa driver I am not able to achieve my requirement . Please give some hint to do this . Thanks Nikhil

[asterisk-users] Three-way conference in Asterisk

Hi How to set a threeway conference in asterisk only for VOIP (I am using only SIP channel). Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] Does anyone know about asterisk 1.10

I read about asterisk 1.10 in website https://wiki.asterisk.org. but didnt find this release from asterisk community. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for

[asterisk-users] MOH uploading is not working with 1.4

Hi am I am using 1.4 asterisk and asterisk GUI. If I do moh upload its is not working .help me on this Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] MOH uploading is not working with 1.4

in google that there is some known issue in asterisk 1.4 regarding this ,I suspect this als o same .Can anyone explain what is expecting to happen in back end when upload MOH file from asterisk GUI ,then I am will able to debug more on this . Thanks Nikhil On 06/03/2011 07:25 PM, Steve

Re: [asterisk-users] MOH uploading is not working with 1.4

any reply on below issue.. Nikhil On 06/06/2011 07:23 PM, Nikhil wrote: Hi when I Upload MOH file from Asterisk GUI ,it is getting success and even not getting any error,But if check the destination path the file is not showing , even the source file and destination path and formate

[asterisk-users] Dial out conference

into the conference.I am using ConfBridge application for asterisk version 1.6.2 Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Dial out conference

Thanks for the helps I use channel originate command to achieve this. Command: asteriskCLI channel originate SIP/201 application ConfBrigde 1234 This will make a call to the 201 user and when connected,it will be routed to conference room . Thanks NIkhil On 06/15/2011 02:17 PM, virendra

[asterisk-users] different format in asterisk

Nikhil -- _ -- 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 asterisk-users mailing list

[asterisk-users] Blind Transfer Connected

). Does anyone knows about this? Thanks Nikhil -- _ -- 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

Re: [asterisk-users] Blind Transfer Connected

in the SIP CC Transfer draft. (Transfer without a GRUU) In asterisk comment is written correct but it is not working. Thanks Nikhil On 07/05/2011 09:44 PM, Kevin P. Fleming wrote: On 07/05/2011 01:54 AM, Olivier wrote: 2011/7/5 Nikhil d.nik...@cem-solutions.net mailto:d.nik...@cem

[asterisk-users] How to Get all users list in asterisk

Hi how to get all the users list that available in asterisk, analog,sip,iax etc.Any cli command or AMI actions available to get this. thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Asterisk as a Operator Phone

. Is any other application available in asterisk to do this . Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Asterisk as a Operator Phone

,and using CLI command I can make calls outside and once call connected I can hear and talk from my Headphone. I planing to enhance chan_alsa module to get the features same as in SIP client. Thanks Nikhil On 07/26/2011 12:57 AM, Duncan Turnbull wrote: Asterisk can run operator phones

[asterisk-users] Radius billing for asterisk

Hi Any company proving radius based billing for asterisk only for accounting ,not authenication and atherization.Please provide some links Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

Can you share the dialplan ,where SIP call is dialing... Thanks Nikhil On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called

Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

The rest of the logic happens in adhearsion. -- Thanks, Ishwar. On Thu, Jul 28, 2011 at 6:33 PM, Nikhil d.nik...@cem-solutions.net mailto:d.nik...@cem-solutions.net wrote: Can you share the dialplan ,where SIP call is dialing... Thanks Nikhil On 07/28/2011 06:15 PM, Ishwar

Re: [asterisk-users] different format in asterisk

Does anyone know about this... On 06/20/2011 04:34 PM, Nikhil wrote: Hi In asterisk channel ,I so number of variable regarding the Codec ,Can anyone explain what are those variable variable means.Below are the variables 1. chan-readformat 2. chan-writeformat 3. chan

[asterisk-users] Send Refer with replaces from asterisk

Hi How to send REFER with replaces from asterisk (Sending out) for doing attended transfer. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] How to get presence using AMI

Hi Using AMI how can I get the presence feature.Below are the requirement. -- List of all users in the PBX including analog and SIP including registration status. -- Status(BUSY or available ) of all users both analog and SIP Please help on this.. Thanks Nikhil

Re: [asterisk-users] How to get presence using AMI

any answer on below.. On 08/18/2011 03:50 PM, Nikhil wrote: Hi Using AMI how can I get the presence feature.Below are the requirement. -- List of all users in the PBX including analog and SIP including registration status. -- Status(BUSY or available ) of all users both analog

[asterisk-users] Invite with replaces handling issue

handle_invite_replaces function,the sip_scheddestroy fun is calling properly but still that dialog is not hangup up. Asterisk version : 1.6.2.13 Note: Asterisk running in VOIP environment. Please help on this. Thanks Nikhil

Re: [asterisk-users] Invite with replaces handling issue

Does anyone face this issue. Thanks Nikhil On 08/24/2011 10:10 AM, Nikhil wrote: Hi I am getting an issue when doing attended transfer from remote server to asterisk.Asterisk is not sending BYE to replaced call once it got invite with replaces from remote server. scenario

[asterisk-users] Presence for channels other than SIP.

then also same result. Is this feature available in asterisk ? Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] How to unregister a sip trunk

. Thanks Nikhil On 11/17/2011 08:43 PM, Danny Nicholas wrote: In Asterisk 10.0 (you didn't state your version) you have sip qualify peer and sip unregister peer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil

Re: [asterisk-users] Confrence call is not make

Conference instead of MeetMe. Thanks Nikhil On 12/08/2011 11:12 AM, Durgesh Mishra wrote: Hi, I am making confrence application. In sip.conf [phone1] type=friend host=dynamic Takes an alphanumeric string. context= employees [phone2] type=friend host=dynamic context= employees [phone3] type

[Asterisk-Users] Doing a # transfer on calls needing a #

Evening All, I was wondering how I would go about enabling the usual # + ext transfer on a call requiring user input followed by a #. For example, when I say ring a bank, they ask me for my account number, I key it in and then press the # key. Of course this doesn't work if I have the T option

[Asterisk-Users] Call Waiting

Hi, I have just set up an Asterisk box.it sure is a big job to get everything perfect, especially when you have picky users. Anyway, the box has 2 X100P's and a couple of sipura spa-2000's connected to the LAN. 1 of the lines connected to the X100P's goes straight to extension 1000 after a

[asterisk-users] US-based echo test servers?

to pay a modest amount for this. Thanks in advance for any suggestions! Best wishes, Nikhil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http

Re: [asterisk-users] Thank you! [Was: Re: US-based echo test servers?]

, with these options, I should have ample opportunity to test things properly. BTW, Igor and Atis: sorry, I certainly didn't intend to start any sort of an argument, however short! For what it's worth, I feel you've both been generous, and I very much appreciate that. Best regards, Nikhil

[asterisk-users] Outbound Call Transfer Problem

Hi I am using Asterisk 1.2 and for the life of me, I am unable to transfer outbound calls (eg calls I initiate from sip extensions). When I press #, nothing happens. Inbound calls transfer fine, but only once per call. The problem happens: - With both software and hardware phones. - With

Re: [asterisk-users] Outbound Call Transfer Problem

Noah Miller wrote: I am using Asterisk 1.2 and for the life of me, I am unable to transfer outbound calls (eg calls I initiate from sip extensions). When I press #, nothing happens. Inbound calls transfer fine, but only once per call. Any suggestions? I have questions: 1) what version of

[asterisk-users] _ALERT_INFO replacement in 1.4?

does not ring. Extension 600, 602 and 603 are all hooked up to Sipura ATAs and need the bellcore-r4 ringtone to differentiate from other incoming lines. Any ideas? Thanks Nikhil Jogia ___ --Bandwidth and Colocation provided by Easynews.com

Re: [asterisk-users] _ALERT_INFO replacement in 1.4?

Bruce Reeves wrote: Does SIPAddHeader(Alert-Info:) not do it? No, but from another thread, setting the _SIPADDHEADER variable works. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

[asterisk-users] Monitor problem, Asterisk 1.2.13

, I'm blind, which partly limits my choice of applications (I've been using console based ones, as I haven't worked out how to use X). Best wishes, Nikhil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

[asterisk-users] Echo cancellation and DTMF from the Asterisk console?

much appreciated! Best wishes, Nikhil. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Echo cancellation and DTMF from the Asterisk console?

, apparently). Maybe Linphone running on the same box as Asterisk will be the solution, after all. That is, if Linphonec will consent to run for me without giving all sorts of errors and segfaulting... and I thought this release of Debian was supposed to be stable! ;) Ah well. Cheers, Nikhil

[asterisk-users] Losing local SIP phones when internet goes down?

that's correct. eth0 has address 10.9.8.1, while eth1 has a global internet IP address. Cheers, Nikhil. - Extract from sip.conf: [general] context=incoming srvlookup=yes realm=nikhil-nair.net ; Various register= statements, not relevant to the local phones [101] ; Aastra 9112i at 10.9.8.101

Re: [asterisk-users] Losing local SIP phones when internet goes down?

On Thu, 4 Feb 2010, Joseph wrote: [...] Does your router runs DHCPD, assigning network addresses on on your LAN? Nope, the DHCPD on the ADSL router is disabled, because I'm running DHCPD on the Debian box. In any case, the ADSL router is not directly accessible from the local net. If

Re: [asterisk-users] Losing local SIP phones when internet goes down?

On Thu, 4 Feb 2010, Joseph wrote: On 02/05/10 02:35, Nikhil Nair wrote: On Thu, 4 Feb 2010, Joseph wrote: [...] Does your router runs DHCPD, assigning network addresses on on your LAN? Nope, the DHCPD on the ADSL router is disabled, because I'm running DHCPD on the Debian box. In any

Re: [asterisk-users] Losing local SIP phones when internet goes down?

On Thu, 4 Feb 2010, Joseph wrote: On 02/05/10 02:05, Nikhil Nair wrote: Extract from sip.conf: [general] context=incoming srvlookup=yes realm=nikhil-nair.net Your resolve authentication to an outside server, isn't it? No, that's just a Realm string which has to match when the Asterisk

Re: [asterisk-users] Losing local SIP phones when internet goes down?

scenarios (router or phone line down) the DNS requests weren't being answered, so Asterisk was waiting, preventing it from doing other things properly. So, as suggested, I'll go ahead and install dnsmasq (or similar), and that should fix things. Cheers, Nikhil

Re: [asterisk-users] Losing local SIP phones when internet goes down?

: no luck. - Rebooted machine and tried again: still no luck. Again, the logs indicate that Asterisk thinks the SIP phones are unreachable. Was there anything special I needed to do with the setup of dnsmasq, or its interface with Asterisk? If not, I'm stuck again. Thoughts? Nikhil

[asterisk-users] PSTN-IVR call

i dial through PSTN it gives beeps sound, but without this line program runs smoothly. Can someone help??? -- Nikhil Kumar rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033

[asterisk-users] asterisk registration

explanation needed, please mail me.I am stuck in this so please help. Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in

[asterisk-users] Fwd: asterisk registration

-- Forwarded message -- From: nikhil singhania niksingha...@gmail.com Date: 10 June 2010 14:08 Subject: asterisk registration To: asterisk-users@lists.digium.com Cc: Ma Hu Ma anshumishra6...@gmail.com Hi all, I think i understand the problem, actually I have two asterisk server

[asterisk-users] calling peer from server

written a simple php script which utilises the exec_dial function inbuilt in phpagi.php file. I have tried diff ways but can't seem to get it work. Can please some one suggest me anything in this regard. -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT

[asterisk-users] calling peer from server

information as the ip of the asterisk server i.e. 172.26.48.208. In the softphone it shows registration successful. Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in

[asterisk-users] can't seem to register, status unmonitored

Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations sip:2...@172.26.48.208 sip%3a2...@172.26.48.208;expires=3013 208 is ip

Re: [asterisk-users] can't seem to register, status unmonitored

-15 4:11 AM, nikhil singhania niksingha...@gmail.com wrote: Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations sip:2

[asterisk-users] Fwd: can't seem to register, status unmonitored

-- Forwarded message -- From: nikhil singhania niksingha...@gmail.com Date: 16 June 2010 12:15 Subject: Re: [asterisk-users] can't seem to register, status unmonitored To: Zeeshan Zakaria zisha...@gmail.com Here is my extensions.conf: [general] static=yes ; default

[asterisk-users] writing echo in inbound file

hello world echoed on output. -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033

Re: [asterisk-users] can't seem to register, status unmonitored

, Zeeshan Zakaria zisha...@gmail.com wrote: you should post this to the list, not to my personal email. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-16 2:45 AM, nikhil singhania niksingha...@gmail.com wrote: Here is my extensions.conf: [general] static=yes ; default

[asterisk-users] calling machine over sip

of the asterisk server i.e. 172.26.48.208. In the softphone it shows registration successful. Thanks in advance -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com

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