[asterisk-users] Hi

Hello everyone --- Regards, Raj. -- _ -- 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 a

[asterisk-users] How to run Music while looking for the caller in Database

Hi, We are using Asterisk and PERL. We have all the call logic in PERL. We are trying to identify the caller using the CID in the Database. As the Database lookup is taking more time (>15 seconds), we want to play some tune while the caller is waiting. How can we do that? Any ideas will be

[asterisk-users] MixMonitor and Call Latency during conversation

Hi, We are using MixMonitor to record the call. When the call is bridged, the latency is significant. We tried to increase the internet speed and the server RAM and processor speed and still we are having that issue. We use VoiceTrading and Gafachi's Termination minutes to make calls. As we

[asterisk-users] How to stop IVR once system receives DTMF?

Hi, We are trying to implement a complex business logic in Asterisk. Executing "Wait_For_Digit" command after playing IVR. We want to stop the IVR once we receive the digit. It is not recognizing the Digit until it completes the IVR. How can we stop the IVR once we receive the digit? Thanks

[asterisk-users] How to detect if the call is being answered by Voice Mail?

->agi->get_variable("ANSWEREDTIME"); to_log($self, "Physician Call Status: $CallStatus; ANSWEREDTIME: $ANSWEREDTIME", 2); return 0; } Is this not the correct way to do this? Or Are there any other methods? Thanks Bharath B. Reddy Bynagari

[asterisk-users] Dial plan sample for detecting Voice Mail

Hi, I am trying to implement a macro-screen mentioned at http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial I put the following code in my extensions_additional.conf screen-from: You have a call; screen-accept: Press 1 to accept this call or any other key to reject.; [macro-screen]

[asterisk-users] Call back DIALSTATUS is empty

TUS is empty. But the Channel status is coming as 6. I have the user decline the call and still not getting any DIALSTATUS Thanks a lot in advance. Thanks MavenSphere-2-logo-web-orange Bharath B. Reddy Bynagari President & CEO <mailto:bynag...@mavensphere.com

[asterisk-users] Rajkiran Reddy sent you a Friend Request on Yaari

Rajkiran Reddy wants you to join Yaari! Is Rajkiran your friend? http://yaari.com/?controller=user&action=mailregister&friend=1&sign=YaariLBF849RQF972ZTA396ZMZ718";>Yes, Rajkiran is my friend! http://yaari.com/?controller=user&action=mailregister&friend=0&sign=Ya

[asterisk-users] FreePBX

Hi All, I am trying to install Asterisk with FreePBX while running install_amp following error is coming can any one help in this regards Thanks in advance.. Linga Reddy Connecting to database..OK Connecting to Asterisk manager interface..OK DB Error: no such tableGenerating AMP configs..OK

Re: [asterisk-users] Working softphone for poket PC

Oops here is the link http://qtechinc.com/speaq_download.htm --Giridhar Bandi On 5/23/07, ram <[EMAIL PROTECTED]> wrote: On 5/23/07, Philipp von Klitzing <[EMAIL PROTECTED]> wrote: > > Hi! > > > Googling arround I found a number of pocket pc softphones. Of those I > > was only able to insta

Re: [asterisk-users] Working softphone for poket PC

Hi try put Speaq speaQ is a VoIP softphone which runs on either Windows Mobile 5.0 or Sharp Zaurus Linux. It can be used to make and record Internet phone calls using any SIP compliant Internet Phone Server. The free Beta Trial Version which can be downloaded from this page, lets you record phone

[asterisk-users] Help needed to server code on Vxworks

Folks, How much efforts are needed to make Asterisk code to run on Vxworks? Is there any document in the distribution which describes the steps to follow to run on Vxworks. Is there any limitation in Vxworks which should be disabled or remove in Asterisk server code. Thanks

[asterisk-users] Caller Initiated Conference

Hi ALL, configuring the caller initiated conference, any one has idea about this please replay me back... -Linga Reddy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] IAX2 Bandwidth setting

Dear Friends, I have configured mobile users through IAX2, when they are calling from out side, my Internet bandwidth is taking 80 Kbits ps ( per channel ) for reducing the bandwidth how will configure the IAX2 in asterisk for good quality of voice, can any help in this regard. Regards Lin

[asterisk-users] Asterisk Configuration

ble to establish the call. I am able to here all automated playback IVR. ex.500, 600 can any one help to configure the inbound / outbound calls and how to add sip users. -Linga Reddy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users m

[asterisk-users] Asterisk Configuration

ble to establish the call. I am able to here all automated playback IVR. ex.500, 600 can any one help to configure the inbound / outbound calls and how to add sip users. -Linga Reddy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users m

[asterisk-users] CDRTools please help

Hi users I have been trying to install CDRTool but due to lack of documents i cannot do it properly so any body please respond me :-(  here what i did is  installed CDRTool in /var/www/  and create database cdrtool  and there are two more files in that setup directory 1) create_tables.mysql 2)

[Asterisk-Users] using mediaproxy for both ASTERISK and SER

Hello Masters,   I am using SER as proxy and registrar and redirect server and iam using MEDIAPROXY for SER to handle Nated calls , So, now i want to connect Asterisk to SER to handle only pstn calls so how the process here goes on ... I mean is, when a sip user who registerd at S

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 2

Hello Guru      Thanks for giving reply.  so, i can use mediaproxy for both SER and as well as ASTERISK but you told me about voip-info but i dint find much docs regarding SER+asterisk cookbooks  and you told me you can use Asterisk as billing for pstn of incoming calls and

[Asterisk-Users] connecting asterisk to pstn help

Hello Masters     Here i going explain what Iam doing and where i need help ..      Iam running Sip Express Router ,Asterisk, on same box (for testing) my Sip express router is working fine and i can accept global register requests with valid account  and in front of S

Re: [Asterisk-Users] IVR sounds not on certain inbound route

please check if you are able to hear the sounds using alaw on IAX i had some problem listening to the sounds using G729 on sip client --Giridhar BandiOn 5/29/06, MC <[EMAIL PROTECTED]> wrote:Coming in on the IAX route is G729. On the SIP lines it is alaw.Giridhar Reddy Bandi wrote:> I d

Re: [Asterisk-Users] IVR sounds not on certain inbound route

I doubt this would be a codec issue . can you tell us what is the codec used on the sip and iax --Giridhar BandiOn 5/29/06, MC < [EMAIL PROTECTED]> wrote:Got 1 issue I can't seem to knock out of this particular box. The IVR works fine on the zap channels and the incoming SIP routes. Butcoming in vi

Re: [Asterisk-Users] features.conf *1 Call Recording

AIL PROTECTED] <mailto:[EMAIL PROTECTED] >   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Record

Re: [Asterisk-Users] features.conf *1 Call Recording

did you include automon => *1 in your features.conf ?? it should be somthing like this [featuremap]automon => *1   --Giridhar Bandi On 5/12/06, Dave Morrow <[EMAIL PROTECTED]> wrote: Thanks for the response.  How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutod

[Asterisk-Users] please help

Hello all,      Iam using this Asterisk server since three weeks and i have to clarify some thing about Asterisk here is my problem Iam trying to use my Asterisk as a gateway to pstn and SER as a proxy and redirection server so,here in SER i had added three or four  users by

[Asterisk-Users] Asterisk IVR / Scalability

Hi i am looking for a good ivr system for my company. these are my question are there any good ivr's that can be easily integrated with asterisk ? and are there any  large scale deployment of asterisk to date ? thanks Giridhar Bandi ___ --Bandw

[Asterisk-Users] polycom unable to start recoding

Hi i have a Polycom Soundstation premier Basic Conference Telephone which is connected to Linksys pap2 boxi am unable to start recording using *1 . but with a normal analog phone connected to linksys  pap2 i am able to start and stop recording .. i tried changing the DTMF setting but no use . can

Re: [Asterisk-Users] Asterisk service crashes

Hi  Billy Try using safe_asterisk and see . safe_asterisk be useful if you fear asterisk may crash.--Giridhar BandiOn 4/19/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: List,   The past few days the asterisk service on my server has crashed several times. I have had it running fo

Re: [Asterisk-Users] Voicemail problem

Hi Daniel can you give us more information so that it would be easy to debug.like voice mail configuration etc Thanks,GIridhar Bandi.On 4/18/06, Daniel Korndorfer <[EMAIL PROTECTED]> wrote: Hi,when I call the voicemail app, it starts and die suddenly. Has anyonealready had this problem?Log:app.c:6

Re: [Asterisk-Users] IVR: playing multiple streams simultaneously?

hi i don't know if we can do that . but i guess we can use audacity .. to mix both the files and get what you want.-Giridhar BandiOn 4/18/06, Herchi Silviu < [EMAIL PROTECTED] > wrote: Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the calle

Re: [Asterisk-Users] Asterisk behind NAT

so that  means that a sip client can access asterisk server which is behind NAT ( assuming that SIP and RTP ports are properly farwarded ) even is nat=no in sip.conf thanks,Giridhar Bandi. ___ --Bandwidth and Colocation provided by Easynews.com -- Asteri

Re: [Asterisk-Users] Asterisk behind NAT

IT works fine behind firewall .enable NAT in sip.conf and it works fine.Giridhar BandiOn 4/6/06, Joao Pereira < [EMAIL PROTECTED]> wrote:Hello to allCan we put Asterisk in a company that has an ADSL connection with just one public IP address? Because with just one public IP, Asterisk musthave a pri

[Asterisk-Users] DID registration status

HII have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ?i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this thanksGiridhar Bandi _

Re: [Asterisk-Users] How is Teliax ?

thank you all for the feed back. --Giridhar Bandi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] How is Teliax ?

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi ___ --Bandwidth and Coloc

[Asterisk-Users] getting problem in Picking up the parked call

Hi all. I am trying following scenerio for call park & pickup. voice is flowing established between B & C, after call-pickup ( instead of A & B ). can anyone please clarify why it is happening like this, ( or ) do i need some more configuration for park&pickup ?

[Asterisk-Users] getting problem in Picking up the parked call

Hi all.   I am trying following scenerio for call park & pickup.   voice is flowing established between  B & C, after call-pickup ( instead of A & B ).   can anyone please clarify why it is happening like this,  ( or )  do i need some more configuration for park&pickup ?       

[Asterisk-Users] (HI,new to asterisk)connecting asterisk to telephonyhardware

the telecom network. SIP-softphone--->---asterisk-->Dialogic card--->telephone line -->called party or vice a versa. Do i have to make any software modifications to achieve this Thanks & Regards Jagadheeswar Reddy Senior Wireless Engineer