[asterisk-users] From external IP am not able hear the audio on the SIP extensions

on this -- Regards Upendra -- _ -- 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] Message Waiting indicator setup in ELASTIX ?

Hi, i wanted to know that if i have a message indicator SIP phone , then MWI will work in ELASTIX ?? Let me know the Details of MWI and how test it. *--* *Thanks Regards* *upendra* -- _ -- Bandwidth and Colocation Provided

[asterisk-users] Elastix Architecture

Hi ALL, Am new to Elastix and wanted to try build new modules in the Elastix , so i want to know how the PHP is running ?? as i see no Apache server inside ?? so wanted to know how its running ? which server and architecture? *--* *Thanks Regards* *Upendra

Re: [asterisk-users] Not able to build the chan_sip.c module

hi, anyone can help me to debug this ?? -- upendar On Mon, May 27, 2013 at 4:09 PM, upendra uppi...@gmail.com wrote: hi, chan_local and res_crypto are building but the chan_sip is not building . installed openssl also but still the chan_sip not building. -- Upendra On Mon, May 27

Re: [asterisk-users] Not able to build the chan_sip.c module

hi, there is no build errors , but the thing is that on Elastix Machine i want to install asterisk1.8.11.0 , while make the chan_sip module is not building, and when i see in the memuselect the chan_sip module driver showing as XXX to enable for building. -- Upendra. On Tue, May 28, 2013 at 1

Re: [asterisk-users] Not able to build the chan_sip.c module

hi all, After installing packages openssl and openssl-devel packages the chan_sip is building . :) :) thanks to all for ur help. -- Upendra. On Tue, May 28, 2013 at 1:33 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, May 27, 2013 at 04:09:06PM +0530, upendra wrote: hi

[asterisk-users] Not able to build the chan_sip.c module

Hi, i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing XXX -- extended , please let me know how to enable it and make build chan_sip module. -- Upendra

Re: [asterisk-users] Not able to build the chan_sip.c module

hi, chan_local is enabled but chan_sip is showing XXX -- Upendra On Mon, May 27, 2013 at 2:13 PM, qasimak...@gmail.com qasimak...@gmail.comwrote: It depends on chan_local see if that is enabled or not. Regards, Qasim On Mon, May 27, 2013 at 11:56 AM, upendra uppi...@gmail.com wrote

Re: [asterisk-users] Not able to build the chan_sip.c module

hi, chan_local and res_crypto are building but the chan_sip is not building . installed openssl also but still the chan_sip not building. -- Upendra On Mon, May 27, 2013 at 2:59 PM, Alec Davis siva...@paradise.net.nz wrote: i am trying to install asterisk newer version on the Elastix

[asterisk-users] Delay in call asterisk

delay after dialing. regards Upendra. -- _ -- 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

Re: [asterisk-users] Delay in call asterisk

Hi, yes if i press # then immediately ring , i configured all these by GUI only so how should i fix this issue?? -- Upendra On Fri, Jan 18, 2013 at 11:06 AM, Don Kelly d...@donkelly.biz wrote: If you dial 2001# does it complete the call immediately? ** ** Your dial plan may

[asterisk-users] Gateway setup

Hi, can anyone help me how to setup a simple gateway for voip phones on elastix system. I dnt no really how it should be connected in reality...? and how to test it . Regards Upendra. -- _ -- Bandwidth and Colocation Provided

[asterisk-users] info on : CONFIG_VOICEBUS_ECREFERENCE

Hi, just going through the code i found that this CONFIG_VOICEBUS_ECREFERENCE undef in the voicebus then how it will defined to run the echo cancell on the respective drievers wctdm24xxp ?? explain how this CONFIG_VOICEBUS_ECREFERENCE enabled and where it is enabled while run time.

Re: [asterisk-users] info on : CONFIG_VOICEBUS_ECREFERENCE

Hi , it implies that under this CONFIG_VOICEBUS_ECREFERENCE code inside this condition will never execute. if we need it should be defined . Regards Upendra On Tue, Nov 20, 2012 at 8:55 PM, Shaun Ruffell sruff...@digium.com wrote: On Tue, Nov 20, 2012 at 05:18:12PM +0530, upendra wrote

[asterisk-users] On which kernel version the DAHDI-2.0 release will work ??

Hi, Can any one tell me on which linux kernel version i can compile and run the DAHDI-2.0 release and test it . *Regards Upendra.* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] Asterisk Test Suite error

still what i am missing in the testsuite. Regards Upendra On Thu, Sep 6, 2012 at 6:41 PM, Matthew Jordan mjor...@digium.com wrote: - Original Message - From: upendra uppi...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com

Re: [asterisk-users] how to load our own .wav sound files in the dial plans for playback

Hi Steve , Thanks for the info . I will gothrough the resources and do the required testing. Regrads Upendra. On Sat, Sep 8, 2012 at 10:58 PM, Steve Edwards asterisk@sedwards.comwrote: On Sat, 8 Sep 2012, upendra wrote: i am trying to add my own sound file in the asterisk dial plan

[asterisk-users] how to load our own .wav sound files in the dial plans for playback

Hi, i am trying to add my own sound file in the asterisk dial plan extension for playback option , i dont no where to put the file and how to give the path in extension file and all so is need that the sound file should be convert in asterisk as .wav file??? regards Upendra

[asterisk-users] Asterisk Test Suite error

-- tests/cause_answered_elsewhere --- SKIPPED -- Dependency: twisted -- Met: True -- Dependency: starpy -- Met: True * give suggestions . Thanks and regards Upendra -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] Load test for FXS and FXO cards

Hi, Can anyone tell me how to do the load test for the FXS and FXO cards and find how much the asterisk machine can load for different processors configuration . Regards Upendra. -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] How to Auto Answer a sip phone

i am using a x lite phone. regards upendra On Fri, Jul 13, 2012 at 10:29 AM, James Sharp ja...@fivecats.org wrote: Different phones use different methods. What kind of sip phones do you have? On Jul 13, 2012, at 12:17 AM, upendra uppi...@gmail.com wrote: Hi, i wanted to make

Re: [asterisk-users] How to Auto Answer a sip phone

hi, thats fine , i am using now xlite-4 , not able to find the option to enable it, let me know . regards Upendra. On Fri, Jul 13, 2012 at 1:03 PM, James Sharp ja...@fivecats.org wrote: From my experience with xlite, the soft phone itself must be configured for auto answer. There is no way

Re: [asterisk-users] How to Auto Answer a sip phone

hi, oh k thanks then i will re-install the xlite 3. Regards Upendra On Fri, Jul 13, 2012 at 2:42 PM, Bharat Lalcheta bharatlalch...@gmail.comwrote: From X-lite 4 version, auto answer feature removed. So use old version if you have or try some other softphone On Fri, Jul 13, 2012 at 1:51

[asterisk-users] How to set SIP to auto answer in the dial plan .

not able Todo this. help me if anyone already done this setup. Regards Upendra. -- _ -- 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 set SIP to auto answer in the dial plan .

Hi, thanks , i need to put this in the sip context... regards Upendra. On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: try with SipAddHeader(uri=answer-after=0) check syntax for Addheader Regards, Zohair Raza On Fri, Jul 13, 2012 at 1:42 PM

Re: [asterisk-users] How to set SIP to auto answer in the dial plan .

Hi, its not working for me ! let me know anyone having sample dialplan so that i can use for test 1 sip call answer. regards Upendra On Fri, Jul 13, 2012 at 9:57 PM, Jared Baxley jared.bax...@gmail.comwrote: You also have to send the alert info you particular phone needs to make

[asterisk-users] How to Auto Answer a sip phone

Hi, i wanted to make dial plan in such a way that the any incoming call to the sip phone should auto answer.(auto pickup) . Help. regards Upendra -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] Tools for Load testing

Hi, Which are the tools for testing the load test for dahdi/Asterisk . - Call load test . - Stress test. Regards Upendra -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Dahdi-2.4.0+2.4.0 means ??

hi, in the documents its says about dahdi-linux + asterisk change, but there is no explanation about it .If any know clearly about dahdi versioning then please let us know . regards Upendra On Wed, Apr 11, 2012 at 6:11 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 04/11/2012

[asterisk-users] Simple Gateway to setup calls bewteen two or 3 asterisk systems .

Hi, i am trying to setup calls between two asterisks , so can anyone tell how establish the calls between two asterisk servers which are on diffrent networks . Regards Upendra. -- _ -- Bandwidth and Colocation Provided

[asterisk-users] Dahdi-2.4.0+2.4.0 means ??

Hi, can anyone tell me what does that 2.4.0+2.4.0 means in dahdi release numbering ??? 2.4.0? regards upendra. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Dahdi-2.4.0+2.4.0 means ??

Hi, thanks for reply , i want to know the 2.4.0 or 2.6.0 means what , how they naming it , by the kernel version or its just a official release number of digum...?? regards upendra. On Wed, Apr 11, 2012 at 6:11 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 04/11/2012 01:39 PM

Re: [asterisk-users] Asterisk auto-dial out a SIP .

thnks for the reply.. i want to know is there any way to call a SIP to SIP by command line regards Upendra -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live