Re: [asterisk-users] Asterisk Installation

2016-11-23 Thread Arun Kumar
Hey Chris, Starts from here, https://wiki.asterisk.org/wiki/display/AST/Getting+Started or try Asterisk Complete guide in pdf format. If you are looking for something graphical, go for elastix or freepbx. Thanks ~Arun On Thu, Nov 24, 2016 at 12:28 AM, christopher kamutumwa < chriskamutu...@gma

[asterisk-users] Voice clarity issue

2014-07-03 Thread arun kumar
Hello all, Im using a GSM gateway device for making outbound calls. GSM device is connected to one of my SIP peer. Now am facing a lot of voice signal problems. I checked with my vendor and there is no issues with signal and device. Any settings in asterisk? Thanks Arun -- _

Re: [asterisk-users] Attack on Sip server.

2014-06-27 Thread arun kumar
Hi, Change the protocol from tcp to udp in iptables. ~Arun On 27 Jun 2014 20:07, "Anurag Rana" wrote: > > Hi All. > > Someone is attacking on my SIP server. > There are lot of requests coming in and I am not able to stop it because I > am unable to detect the IP address. > I used wireshark

Re: [asterisk-users] T1 Card RED ALARM

2014-06-24 Thread arun kumar
Its an old box with Asterisk 1.2 On 25 Jun 2014 03:46, "Mc GRATH Ricardo" wrote: > > Why you configure zaptel.conf? should configure on dahdi files > > Mc GRATH Ricardo > E-Mail mcgra...@mail2web.com > > -- > _ > -- Bandwidth and

Re: [asterisk-users] T1 Card RED ALARM

2014-06-24 Thread arun kumar
Cables are workig fine in my other box. On 25 Jun 2014 00:46, "Steve Totaro" wrote: > Remember to always check your cables first. > > Thanks, > Steve T > > > On Tue, Jun 24, 2014 at 1:47 PM, arun kumar > wrote: > >> >> Thank you Josh for your

Re: [asterisk-users] T1 Card RED ALARM

2014-06-24 Thread arun kumar
Thank you Josh for your valuable reply. I will do try changing the server and let you know what happening. ~Arun On Tue, Jun 24, 2014 at 8:39 PM, Josh Metzger wrote: > > > On Tue, Jun 24, 2014 at 5:25 AM, arun kumar > wrote: > >> Hello All, >> >> I h

[asterisk-users] T1 Card RED ALARM

2014-06-24 Thread arun kumar
Hello All, I have a Digium Wildcard TE410P Quad-Span T1 Card, when I do connect T1 lines it goes in RED. When I do connect the same line on a different Server (Same Model T1 Card) it works fine. How do I examine/diagnose my T1 Card for any hardware failures. I heard about loopback test , how h

[asterisk-users] My new blog http://cciev.ciscovoicetech.com/

2011-02-19 Thread Arun Kumar
Hi Guys, Soon, I'll be starting a new section related to Asterisk (around 4 years of full time experience with Asterisk, Trixbox, SER, OpenSer, MediaProxy, AGI*) so let me know if you like to see some topic coming. Cheers Arun -- ___

Re: [asterisk-users] Asterisk CCM, CME Integration

2009-05-23 Thread Arun Kumar
HI All, I got solved this issue. Thanks all for your help Arun On Sun, May 24, 2009 at 1:58 AM, David Backeberg wrote: > On Wed, May 20, 2009 at 12:44 AM, Arun Kumar wrote: > > here is my problem: when I call from 6004 to my cme extension 4615, on > 4615 > > I've conf

[asterisk-users] Fwd: Asterisk CCM, CME Integration

2009-05-21 Thread Arun Kumar
Hi All, please provide some help. I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones > Asterisk SIP Trunk > Call Manager -> CME

[asterisk-users] Asterisk CCM, CME Integration

2009-05-19 Thread Arun Kumar
Hi All, I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones > Asterisk SIP Trunk > Call Manager -> CME -> 461X Phones 461X Phones

[asterisk-users] about the Dial application

2008-06-25 Thread arun kumar
Hi guys I am working in Kanpur, India. When someone calls to my server i forward the call to someone else by Dial command. After dialing it says Native bridging. And after that I am unable to detect whether the call was answered, the called number was busy or the call was not completed. One more

[asterisk-users] Fwd: Detection of Answer, hangup, busy etc while using Dial command

2008-06-21 Thread Arun Kumar Chaudhary
-- Forwarded message -- From: Arun Kumar Chaudhary <[EMAIL PROTECTED]> Date: Sat, Jun 21, 2008 at 4:51 PM Subject: Detection of Answer, hangup,busy etc while using Dial command To: [EMAIL PROTECTED] Hi Guys, I am in kanpur, India. I am using Dial() command in my phpagi scr

[asterisk-users] Asterisk Nokia

2008-01-08 Thread Arun Kumar
Hi, I've two wifi-phones 1. Nokia e65 2. HP Ipaq I've configure two sip exten in my asterisk and using these exten in my phones. But my Nokia phone is keep on loosing the connectivity very soon life 1-2 min the qualify packet will be double of my HP. So, when I try to call my Nokia SIP exten it

[asterisk-users] Need help in selecting DTMF Mode

2007-11-21 Thread Arun Kumar
Hi here is my setup : 1. USER -> PSTN -> Asterisk A -> IAX2 Trunk -> Asterisk B -> SER -> Asterisk C (Accepting DTMF) All Asterisk box has dtmfmode = inband, when user pressed DTMF able to receive and working fine. 2. Asterisk C ---> Dial Customer Customer input DTMF and its not taking any dt

Re: [asterisk-users] Softphone to be installed on the Mobile

2007-11-21 Thread Arun Kumar
try to use http://www.fring.com/download/ On Nov 21, 2007 3:28 PM, Ricardo Carvalho <[EMAIL PROTECTED]> wrote: > Here's one sip softphone for mobiles you can give a try: > http://www.minisip.org/ > > Regards, > Ricardo Carvalho. > > > ___ > --Bandwidth a

[asterisk-users] DTMF Problem

2007-11-15 Thread Arun Kumar
Hi Here is my setup: USER --> PSTN -> Asterisk A > IAX2 Trunk > Asterisk B -> SER > Asterisk C I'm not able to receive DTMF passed by USER on Asterisk C. All my asterisk boxs are configured with same DTMF type (auto) but no luck. Please help on this issue.

Re: [asterisk-users] SER with Asterisk intergration

2007-11-01 Thread Arun Kumar
just configure SER on another port and use. On 11/1/07, satish patel <[EMAIL PROTECTED]> wrote: > > Dear all > > anybody have implement SER with Asterisk in single machine ?? i > have asterisk with 200 SIP device but i voice qulity and load of asterisk is > bit high so i need to implemen

Re: [asterisk-users] flooded by "Maximum trunk data space exceeded" messages

2007-10-31 Thread Arun Kumar
try to reduce number of calls on trunk or create multiple trunks. On 10/31/07, Louis-David Mitterrand <[EMAIL PROTECTED]> wrote: > > Hi, > > Using 1.4.13 and trunking a single iax channel to a similar box my > asterisk console is flooded with: > > [Oct 31 10:49:34] WARNING[5195] chan_iax2.

[asterisk-users] Asterisk H323 Config

2007-10-21 Thread Arun Kumar
Hi Need help on this setup: Incoming DID in H323 > Asterisk Server --> SIP Phone please tell me to achieve this above setup what needs to be done in Asterisk. thanks Arun ___ --Bandwidth and Colocation Provided by http://www.api-di

[asterisk-users] Asterisk Voicemail

2007-10-01 Thread Arun Kumar
Hi I've configured my asterisk and voicemail all works fine but I want to restrict call time to voicemail that is when user calls voicemail he can use voicemail system only for a max of 5 min that is after five minutes asterisk should disconnect the call. thanks Arun

[asterisk-users] RTP Call Disconnect

2007-09-17 Thread Arun Kumar
Hi All, UA <> Asterisk Server <-> UB if there is no rtp for a specified number of minutes / seconds then I want to disconnect the call. I've tried using rtptimout and rtpholdtimeout but no luck pls guide. thanks arun ___ Sign up n

Re: [asterisk-users] How to verify IAX trunking

2007-08-10 Thread Arun Kumar
run iax2 show peers and see next to port (T) if it comes then you are using IAX2 Trunking feature. On 8/10/07, George Pajari <[EMAIL PROTECTED]> wrote: > > How can one verify that IAX trunking is in effect and that Asterisk is > trunking multiple call paths between two Asterisk servers? > > With 1

[asterisk-users] Asterisk-1.2.22 DeadAGI Hangup

2007-07-22 Thread Arun Kumar
Hi I've upgraded my server to asterisk-1.2.22 from 1.2.10 after that my DeadAGI scripts are not working properly. Like after hangup I used to do some more work now its not working. Please help. thanks arun ___ --Bandwidth and Colocation Provided by

[asterisk-users] Asterisk Freeze

2007-07-20 Thread Arun Kumar
HI Here is my info: Asterisk - 1.2.10 with zaptel 1.2.7, 10 queues with 7 sip agents this asterisk box is connected to another asterisk box using 5 IAX trunk to load balance no of calls on each IAX trunk (g729 over trunk). Suddenly my cli start flooding with message: Maximum trunk data space ex

Re: [asterisk-users] Asterisk PRI Busy Problem

2007-07-18 Thread Arun Kumar
issue is got solved by moving to another pri card and now congestion works fine with my ISP. thanks all. On 7/18/07, Andrew Joakimsen <[EMAIL PROTECTED]> wrote: On 7/17/07, Jared Smith <[EMAIL PROTECTED]> wrote: > > On Tue, 2007-07-17 at 12:52 -0400, Andrew Joakimsen wrote: > > I did a quick

[asterisk-users] Asterisk PRI Busy Problem

2007-07-17 Thread Arun Kumar
Hi, I've an PRI coming to my asterisk ,calls are coming fine and my agents are able to answer no prob. but I've an agreement with my telco with some incoming no if the no of calls on these no are more then 3 then send to another no. they use busy signal to divert call on another number so I'm sen

[asterisk-users] Asterisk Help

2007-07-08 Thread Arun Kumar
Hi I need help in configuring a auto dialer system using Asterisk. I'm holding my customers number in MySQL want to fetch 10 numbers one time and dial if gets connected and answered by customer wants to play a sequence of message . Please help . I've tried here is my code to place calls but in

[asterisk-users] Early Media Handling

2007-07-08 Thread Arun Kumar
Hi using php script and Asterisk manager I'm dialing numbers and once gets connected send to an exten in my dial plan that plays an automated message but some time without answering even it goes to my exten. How can I handle early media in Asterisk that is I want only when user answer the call i

[asterisk-users] Asterisk Manager

2007-07-06 Thread Arun Kumar
Hi this is my code for * manager: $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die("Connection to host failed"); fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Username: $strUser\r\

Re: [asterisk-users] Help with IAX Trunk

2007-07-03 Thread Arun Kumar
thanks for reply. I've same setup with siml. incoming calls 10-12 it works fine but some time my machies get hang and gives same IAX max data space error. thanks On 6/27/07, Jared Smith <[EMAIL PROTECTED]> wrote: On 6/27/07, Arun Kumar <[EMAIL PROTECTED]> wrote: > so ,

[asterisk-users] Help with IAX Trunk

2007-06-27 Thread Arun Kumar
Hi I've two servers : 1. UK 2. Pakistan Pakistan * server has ISDN30. Pakistan(ISDN30) <> UK ===> User Im planning to setup an IAX2 trunk between these two server ? so , how much bandwidth I need for 30 simul. calls ? Im planning to use G729 on both my server ? to support 30 calls ove

Re: [asterisk-users] iax trunking on OpenBSD

2007-06-07 Thread Arun Kumar
you can use FreeBSD 6.1 its working fine for me with ztdummy and I'm able to use IAX2 trunk. On 6/7/07, Sebastian Reitenbach <[EMAIL PROTECTED]> wrote: Hi, do I have a chance to use iax trunking on OpenBSD where there is no zaptel driver or ztdummy available? Do I can use sth. else as timing s

Re: [asterisk-users] IAX2 Trunk No Sound

2007-06-05 Thread Arun Kumar
On 6/5/07, Noah Miller <[EMAIL PROTECTED]> wrote: Hi Arun - > I've two boxes connected over IAX2 trunk before IAX I was using SIP trunk > and they were working fine b'coz of bandwidth issue I changed from SIP to > IAX now I'm facing a strange problem after some time on the cli of my > asterisk

[asterisk-users] IAX2 Trunk No Sound

2007-06-05 Thread Arun Kumar
Hi I've two boxes connected over IAX2 trunk before IAX I was using SIP trunk and they were working fine b'coz of bandwidth issue I changed from SIP to IAX now I'm facing a strange problem after some time on the cli of my asterisk box I see lots of messages of IAX2 trunk and b'coz of that my agent

[asterisk-users] G729 License

2007-06-04 Thread Arun Kumar
HI I bought 20 license from Digium and install in my server and b'coz of some problem I've to change my server is it possible that I can use those lice and register again in my new server ? Is it possible that I'll be able to use those lice in my old box also ? thanks arun

[asterisk-users] IAX2 Trunk Problem

2007-06-04 Thread Arun Kumar
Hi I've two boxes connected over IAX2 trunk but suddenly my cli is getting flood with these messages: iax2_trunk_queue: Maximum data space exceeded and b'coz of that my agents are not able to hear any thing. when this happened that time there were 9 calls. my * version is 1.2.18 and 1.2.14 t

[asterisk-users] Digium Card

2007-06-04 Thread Arun Kumar
HI I'm looking for a card that support both PRI and TDM. Please suggest me ? thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailm

Re: [asterisk-users] Asterisk Queue

2007-06-03 Thread Arun Kumar
Hi sorry for not details. when ever I see this message on * console my agents are not able to listen to announcement. thanks arun On 6/3/07, Mattt <[EMAIL PROTECTED]> wrote: And you don't find that sufficiently self-explanatory? On Sun, 2007-06-03 at 13:02 +0400, Arun Kumar wro

[asterisk-users] Asterisk Queue

2007-06-03 Thread Arun Kumar
Hi my * box is giving me these warning and b'coz of second warning line my agents are not able to hear the announcement in the queue some time it happen many time 2007-06-03 13:40:30 WARNING[28016]: chan_sip.c:2612 sip_write: Can't send 4113568 type frames with SIP write 2007-06-03 13:40:30 WAR

[asterisk-users] Asterisk Queue

2007-06-03 Thread Arun Kumar
HI Im getting strange message on asterisk console WARNING[26853]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun ___ --Bandwidth and Colocation provided by Easynews.

[asterisk-users] Asterisk Crash

2007-06-03 Thread Arun Kumar
Hi I've two boxes connected via IAX2 Trunk were working fine from few days suddenly today one box is got crashed with this message 2007-06-03 12:25:37 WARNING[26511]: chan_sip.c:2612 sip_write: Can't send 4113608 type frames with SIP write my version of * is 1.2.14 on FC4 thanks arun _

[asterisk-users] Fwd: TC400B load problem

2007-05-24 Thread Arun Kumar
-- Forwarded message -- From: Arun Kumar <[EMAIL PROTECTED]> Date: May 13, 2007 5:40 PM Subject: TC400B load problem To: Asterisk Users Mailing List - Non-Commercial Discussion < asterisk-users@lists.digium.com> Hi Im trying to install my TC400B trans coder card

Re: [asterisk-users] Re: TC400B load problem

2007-05-14 Thread Arun Kumar
thanks Matthew, I'll try to call Digium. On 5/14/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote: On May 14, 2007, at 4:53 AM, Arun Kumar wrote: > Im trying to install my TC400B trans coder card when I do: > > modprobe wctc4xxp > > tail -f /var/log/messages says

[asterisk-users] Re: TC400B load problem

2007-05-14 Thread Arun Kumar
Hi Im trying to install my TC400B trans coder card when I do: modprobe wctc4xxp tail -f /var/log/messages says: May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) May 13 14:56:36 pbx2 kernel: Registered codec translator

[asterisk-users] TC400B load problem

2007-05-13 Thread Arun Kumar
Hi Im trying to install my TC400B trans coder card when I do: modprobe wctc4xxp tail -f /var/log/messages says: May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) May 13 14:56:36 pbx2 kernel: Registered codec translator

Re: [asterisk-users] Queue Status

2007-05-08 Thread Arun Kumar
Hi I already tried asterisk manager but Im not able to get status for each queue member. thanks On 5/8/07, Edoardo Serra <[EMAIL PROTECTED]> wrote: Hi, you can use an AGI to connect to asterisk manager and retrieve the info you need about the queue. Hope it helps Arun Kumar ha s

Re: [asterisk-users] zaptel compile error

2007-05-08 Thread Arun Kumar
hi vi /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c this file and look for line that says 2.6.19 change it to 2.6.18 and save and compile arun On 5/7/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: On Fri, May 04, 2007 at 01:55:20PM -0400, mail-lists wrote: > I get the following error when

Fwd: [asterisk-users] Change Codec

2007-05-06 Thread Arun Kumar
NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Arun Kumar *Sent:* Tuesday, May 01, 2007 9:24 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Change Codec Hi I've inst

[asterisk-users] Fwd: Queue Status

2007-05-06 Thread Arun Kumar
Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun ___ --

[asterisk-users] Manager API Output

2007-05-05 Thread Arun Kumar
Hi, Is there any way that I can store my manager API output that is: My question is that is there any why using that I can get the QueueStatus and store the result in some text file for further processing. thanks arun ___ --Bandwidth and Colocation

[asterisk-users] Queue Status

2007-05-05 Thread Arun Kumar
Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun ___ --

[asterisk-users] Queue Answer

2007-05-04 Thread Arun Kumar
Hi this is my setup: Customer <-> PRI <-> Server A with G729 <-> IAX2 Trunk(G729) <-> Server B <-> SIP Exten allowed codec=g729 <-> Snom phone Agents setup is working fine. I want when my agents are not available (queue) like not logged in or all are busy so no calls should come to my server b

Re: [asterisk-users] Change Codec

2007-05-02 Thread Arun Kumar
NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Arun Kumar *Sent:* Tuesday, May 01, 2007 9:24 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Change Codec Hi I've inst

[asterisk-users] Change Codec

2007-05-01 Thread Arun Kumar
Hi I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've allowed ulaw and g729. I want to change the codec for outbond calls. Please help not able to find anything using search. thanks arun ___ --Bandwidth and Colocation provided by Ea

[asterisk-users] don't want call to get answered

2007-04-30 Thread Arun Kumar
In my * box I've configured two queues and incoming number and whenever any one calls those number call comes to my *box and it sends call to my agents in queue. but if no agent is available it still answer the call. Is there any why when my agents are not available I don't want call to get answer

Re: [asterisk-users] Call Connection Problem

2007-04-25 Thread Arun Kumar
ne. I'll need more information to help further. On 4/24/07, Arun Kumar <[EMAIL PROTECTED]> wrote: > Hi, > > I'm running a php script to generate calls using Asterisk Manager and > its working fine. this script call a specified land line number if the phone > is answ

[asterisk-users] Call Connection Problem

2007-04-24 Thread Arun Kumar
Hi, I'm running a php script to generate calls using Asterisk Manager and its working fine. this script call a specified land line number if the phone is answered then It will connect to an extension and play an IVR. But I see in Asterisk CLI its placing the call and it shows channel answered but

[asterisk-users] Exten Length

2007-04-22 Thread Arun Kumar
Hi, I've configured my exten.conf for few exten. But I'm curious to know how long can be my exten like (exten => XXX.). Is there any limit for this or not. B'coz I've noticed one strange problem. I'm usnig snom300 as my hard phone to make calls. when my exten length is 14 then calls goes

[asterisk-users] CallerID Auth

2007-04-20 Thread Arun Kumar
Hi, in my dial plan I've configured two trunks to make outbound calls (one for national calls and other international). I want to allow only 2-3 extension to make use of my international trunk to make outbound calls so I want some kind of auth. based on their callerid . Please guide. thanks ar

[asterisk-users] Ser as IVR

2007-04-19 Thread Arun Kumar
Hi, Is it possible to design an IVR using SER ? If yes please advice. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listi

[asterisk-users] Asterisk Queue Call Transfer

2007-04-19 Thread Arun Kumar
Hi I've configured the queue on my asterisk box and everything is working fine. In my queue I've 3 agents logged in the queue. When call comes they are able to receive the calls without any problem. But some time they are on break and there extension rings and no one is there to answer the call (

Re: [asterisk-users] No of Calls

2007-04-18 Thread Arun Kumar
ing that you are looking for. Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: "Arun Kumar" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Re: [asterisk-users] No of Calls

2007-04-17 Thread Arun Kumar
how do I check that whether trunking is working or not ? No I don't any timing soure (like zaptel card) b'coz these are test server. what else I can use for timing. thanks On 4/17/07, Thomas Kenyon <[EMAIL PROTECTED]> wrote: Arun Kumar wrote: > I've tried this but sti

Re: [asterisk-users] No of Calls

2007-04-17 Thread Arun Kumar
how do I check that whether trunking is working or not ? No I don't any timing soure (like zaptel card). thanks On 4/17/07, Thomas Kenyon <[EMAIL PROTECTED]> wrote: Arun Kumar wrote: > I've tried this but stil some problem Like if I use this link that you > gave m

Re: [asterisk-users] No of Calls

2007-04-17 Thread Arun Kumar
ote: http://site.asteriskguide.com/bandcalc/bandcalc.php On Tue, 17 Apr 2007 11:54:28 +0400, "Arun Kumar" <[EMAIL PROTECTED]> wrote: > Hi > > > sorry for asking the same question again: > > here is my details: > > I've 50 exten in my sip and I'm using s

[asterisk-users] No of Calls

2007-04-17 Thread Arun Kumar
Hi sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how

[asterisk-users] Adding Noise or background noise

2007-04-08 Thread Arun Kumar
Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to tru

[asterisk-users] Number of calls

2007-04-02 Thread Arun Kumar
HI, Here is my setup: USERS -> PSTN -> Service Provider -> Asteriskbox1 -> IAX2 trunk -> Internet -> IAX2 trunk -> Asteriskbox2 ->Sip Clients between asteriskbox1 and asterisk box2, I've VPN configured. from Asteriskbox2 to internet my line speed is 1MB. Is there any why that I can calculate

Re: [asterisk-users] Asterisk Inbound Problem

2007-02-21 Thread Arun Kumar
branch=z9hG4bK74ac10cb8c5d89375bf77d4aaa15fcea..Content-Length: 0 > > > # > U :5060 -> :5060 > BYE sip:800942@ SIP/2.0..Max-Forwards: 5..To: < > sip:[EMAIL PROTECTED]:5060>;tag=as7 8bcde29..From: > >;tag=3380960452-790279..Contact: > :5060>..Call-ID: > [EMAIL PROTECTED]: 2 BYE..Via: > SIP/2.0/UDP :5060; > branch=z9hG4bK610e4f29ad9631a0065d4

Re: [asterisk-users] Asterisk Inbound Problem

2007-02-19 Thread Arun Kumar
et that your IAX phone is using ulaw and your DID provider is using something else like G729. Mark On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote: > HI > > I've configred an Incoming DID in my asterisk and when I call from > outside I see call is coming to my Asterisk server and

[asterisk-users] Asterisk Inbound Problem

2007-02-19 Thread Arun Kumar
HI I've configred an Incoming DID in my asterisk and when I call from outside I see call is coming to my Asterisk server and then from asterisk it rings on a particulat exten but when I pickup the call the call get disconnect immediate and on the other end it keep trying (ringing). here is my ex

[asterisk-users] Re: Load Balancing

2007-01-22 Thread Arun Kumar
use LCR is really good. On 1/22/07, raviprakash sunkara <[EMAIL PROTECTED]> wrote: Hello Users, How can I perform the load Balancing in My SIP server of Both OpenSER and Asterisk , Currently I have One OpenSER server and Asterisk Server, For OpenSER is to need use these modules, and is

Re: [asterisk-users] Happy X-mas

2006-12-22 Thread Arun Kumar
Hi All, Wish you a very HAPPY and Merry Christmas to all and your beloved once. Arun On 12/23/06, Josué Conti <[EMAIL PROTECTED]> wrote: Hi ALL, ** I like very to desire you and your family, a Merry Christmas, with much love, peace, professional and personal success. Best Regards Josue 20

[asterisk-users] ASterisk and SER

2006-12-04 Thread Arun Kumar
HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 6 asterisk passes this is ser and then again ser passes this no (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wron

Re: [asterisk-users] Re:Call Transfers in SER + Asterisk

2006-11-24 Thread Arun Kumar
HI, thanks for your reply. Here is my ser.cfg and other config files please guide me. ser.cfg -- debug=5 fork=no log_stderror=yes listen=2xx.xxx.xxx.xxx # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:[EMA

[asterisk-users] Asterisk with SER

2006-11-23 Thread Arun Kumar
HI, I'm not able to find some good doc or manual regarding Integration of Asterisk with SER. Bacially, I want to forward my calls from SER to asterisk. If some one already done this please guide me. thanks in advance arun ___ --Bandwidth and Colocati

Re: [asterisk-users] Audiocodes MP-20x

2006-10-30 Thread Arun Kumar
hican you please post some user or config guide.thanks in advancearunOn 10/24/06, Ed Greenberg < [EMAIL PROTECTED]> wrote:I will sign in with good experiences with MP124 and Mediant 1000. I have an MP202 under test.--On Tuesday, October 24, 2006 10:10 AM +0300 Paul Ianas<[EMAIL PROTECTED]> wrote:>>

Re: [asterisk-users] Iax Netstat Output

2006-09-22 Thread Arun Kumar
b'coz I have same setup at other client is working fine no problem.On 9/22/06, Tzafrir Cohen <[EMAIL PROTECTED] > wrote:On Fri, Sep 22, 2006 at 03:09:47PM +0530, Arun Kumar wrote:> can please some one tell me where is what wrong. >> iax2 show netstats>  

[asterisk-users] Iax Netstat Output

2006-09-22 Thread Arun Kumar
can please some one tell me where is what wrong.iax2 show netstats                                 LOCAL - REMOTE Channel                    RTT  Jit  Del  Lost   %  Drop  OOO  Kpkts Jit  Del  Lost   %  Drop  OOO  KpktsIAX2/callaus-3          

[asterisk-users] Iax2 show netstat

2006-09-22 Thread Arun Kumar
can please some one tell me where is what wrong. iax2 show netstats LOCAL - REMOTE ChannelRTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts IAX2/callaus-3

Re: [asterisk-users] Iax Netstat Output

2006-09-21 Thread Arun Kumar
no zap -> iax2 -> iax2 only iax2 -> iax2 -> iax2thanksOn 9/21/06, Ma Zhiyong <[EMAIL PROTECTED] > wrote:I know what, if I use ZAP->IAX2 --->IAX2, I also got one direction poor. But if I use SIP->IAX2 --->IAX2->, every think is OK. ___--Bandwidth and Coloc

[asterisk-users] Iax Netstat Output

2006-09-21 Thread Arun Kumar
HiI've * running but I'm other side voice is not so clear and delay. this is my iax netstat output can someone help me where is the problem.here is the iax netstat output Channel   RTT   Jit Del Lost    Drop    OOO Kpkts   Jit Del Lost

[asterisk-users] Asterisk Outgoing Spool Failed

2006-09-08 Thread Arun Kumar
himy asterisk -r shows me Most of the times Outgoing Spool Failed. Can some one tell me why is it happening and how to solve this issue. Is it a problem ? thanks in advance.arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users ma

Re: [asterisk-users] Asterisk Outgoing Spool Failed

2006-09-08 Thread Arun Kumar
hithanks for reply.I'm using vicidial to make calls at 2.0 dial level it is able to make calls but when I see the asterisk -r most of the time it shows Outgoing Spool Failed. Which Spool File ?thanks arunOn 9/8/06, Matt Riddell (IT) <[EMAIL PROTECTED]> wrote: -BEGIN PGP SIGNED MESSAGE-Hash:

Re: [asterisk-users] asterisk + centos 4.3

2006-07-14 Thread Arun Kumar
hi,can you describe what you want.../ArunOn 7/14/06, varun <[EMAIL PROTECTED] > wrote:Hello,We were able to get asterisk going withX100p cards on centos 4.2.But could on centos 4.3 due to kernelissues.Anybody has faced this issue ?And how do sort it out so that wecan use centos 4.3 ?ThanksVarun___

Re: [Asterisk-Users] a2billing

2006-06-28 Thread Arun Kumar
Hi,you check your a2billing.conf file:; Please enter here the file you want to play when we prompt the calling party to enter his destination number; file_conf_enter_destination = prepaid-enter-number-u-calling-1-or-011 file_conf_enter_destination = prepaid-enter-destI think this file should help y

[Asterisk-Users] Asterisk auto-dial Help

2006-06-28 Thread Arun Kumar
Hi,When you originate a call asterisk essentially callouts to the Specified channel and the when answers connects the the context,extension,priority. What if I want my dial plan to make the origination call and the destination call. What would I specify for my dialplan/callout file?thanks in advanc

Re: [Asterisk-Users] Voip / AudioCodes MP-108 Help Needed

2006-06-27 Thread Arun Kumar
Hi,Here are the step by step instructions for setting up a brand new AudiocodesFXS gateway for use with an Asterisk server: Connect the gateway to a network switch and connect a computer to the same switch. Then configure the IP address of the computer to 10.1.10.2. Then runyour web browser and poi

Re: [Asterisk-Users] free sun boxes

2006-06-17 Thread Arun Kumar
Hi, What is the Location. I'm studying in India. Is it possible. thanks ArunOn 6/17/06, Bob Knight <[EMAIL PROTECTED]> wrote: I have 4 sparc based sun boxes I am about to pay money so I canget rid of them.  They are running older versions of Solaris.You should be able to load Solaris 10 and play

Re: [Asterisk-Users] Microsoft CRM & Asterisk

2006-06-05 Thread Arun Kumar
Hi Calvis, Its good if I can help you in any why with this project. thanks ../ArunOn 6/2/06, calvis <[EMAIL PROTECTED]> wrote: Has anyone done any integration with Asterisk & Microsoft Dynamics CRM?  Ijust wanted to check with the list before I pursue a project with the aboveintegration.  In addi

RE: [Asterisk-Users] Help Needed

2003-07-18 Thread Arun Kumar Sharma, Noida
you will need PhoneJACK (PCI or ISA) Few days ago they announced that there is a new PhoneJACK PCI available - with new DSP and etc. Best regards Lubo Arun Kumar Sharma, Noida wrote: > Thanks Adam, > > This document provides me a high level architecture of Asterisk. Can you > ple

RE: [Asterisk-Users] Help Needed

2003-07-17 Thread Arun Kumar Sharma, Noida
Message- From: Low, Adam [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 19:11 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] [Asterisk-Users]Help Needed http://www.digium.com/handbook-draft.pdf > -Original Message- > From: Arun Kumar Sharma, Noida [mailto:[EMAIL PR

[Asterisk-Users] [Asterisk-Users]Help Needed

2003-07-17 Thread Arun Kumar Sharma, Noida
Hi Everybody, I am new to Asterisk. Can anybody suggest me some link where I can find architecture level detail of this system. My aim is to find out how easy it is to port it on a new hardware (T1/E1 and POTS)? Any input is highly appreciated. Regards Arun

RE: [Asterisk-Users] Segmentation fault with chan_oh323

2003-07-17 Thread Arun Kumar Sharma, Noida
Hi Everybody, I am new to Asterisk. Can anybody suggest me some link where I can find architecture level detail of this system. My aim is to find out how easy it is to port it on a new hardware (T1/E1 and POTS)? Any input is highly appreciated. Regards Arun -Original Message- From: Mar