Re: [asterisk-users] Pager Duty Service on Asterisk
On Sat, Nov 17, 2012 at 7:22 AM, Jared Baxley wrote: > You can accomplish this with time conditions. > Thanks Jared. Any docs or tutorials to refer to set up? Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pager Duty Service on Asterisk
Hi, Does Asterisk has pager duty feature and write ups or How To's to setup? Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
On Tue, Jan 3, 2012 at 7:23 PM, Zohair Raza wrote: > all of them have a wiki page > > http://lmgtfy.com/?q=Asterisk > http://lmgtfy.com/?q=freeswitch > http://lmgtfy.com/?q=openser > http://lmgtfy.com/?q=TrixBox > > Regards, > Zohair Raza > > Hi Zohair I was interested in some sort of comparison sheet and its advantages over each other. Regards Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wait_for_answer: Unable to write frametype: 2 on Asterisk *CLI>
On Fri, Dec 30, 2011 at 8:10 PM, Kaushal Shriyan wrote: > Hi, > > I get this warning "[Dec 30 19:59:20] WARNING[17666]: app_dial.c:1353 > wait_for_answer: Unable to write frametype: 2" on Asterisk *CLI> > > Any clue ? > > Please let me know if anyone needs any additional information or details > of configuration files. > > Regards, > > Kaushal > > Hi, Checking in again for my earlier post to this Mailing List. Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wait_for_answer: Unable to write frametype: 2 on Asterisk *CLI>
Hi, I get this warning "[Dec 30 19:59:20] WARNING[17666]: app_dial.c:1353 wait_for_answer: Unable to write frametype: 2" on Asterisk *CLI> Any clue ? Please let me know if anyone needs any additional information or details of configuration files. Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Caller Number in E1 PRI ISDN Lines
On Sat, Dec 17, 2011 at 6:16 AM, Carlos Rojas wrote: > Hello > Did you use callerid(num) in your dial plan? > Thanks Carlos it worked after looking at the link -> http://www.voip-info.org/wiki/view/Asterisk+func+callerid Thanks and Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set Caller Number in E1 PRI ISDN Lines
Hi, I am having an E1 PRI ISDN Lines with 30 bearer channels and 1 D Channel with hundred DIDs (Direct Inward Dialing) numbers attached to a Sangoma PRI Card on the server, I am using asterisk 1.8.5 on CentOS 5.6. How can i configure DIDs so that if i make an outgoing call the DID number should go to the caller not the pilot number For example PRI Numbers Range -> 31303000 - 31303099 Pilot Number -> 31303000 So if i need to set caller number as 31303008 for example and not as 31303000, is there a way to set this in dial plan (extensions.conf) Please guide and let me know if anyone needs more information and have questions. Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
On Thu, Sep 15, 2011 at 11:45 AM, Steve Edwards wrote: > On Wed, 14 Sep 2011, Kaushal Shriyan wrote: > >> Please let me know the correct procedure to get .alaw file format since I >> belong to India region. > > Well, let's see... > > You used '-t ul' and got a 'ulaw.' > > I wonder what '-t al' will give you? > > Failing that, I suspect 'sox --help' or Google would be a more responsive > resource. > Thanks Steve. it worked this time. Also is there a way to verify .alaw file without playing the file. Although i did play obd-demo.alaw it worked fine. I did ran the below command [root@host0040 test]# file obd-demo.alaw obd-demo.alaw: data [root@host0040 test]# [root@host0040 test]# sox obd-demo.alaw -e stat sox: Failed reading obd-demo.alaw: Do not understand format type: alaw [root@host0040 test]# Basically trying to understand the properties of the .alaw file about encoding and details. Please guide. [root@host0040 test]# sox -V obd-demo.wav -r 8000 -b -t al -c 1 obd-demo.alaw sox: Detected file format type: wav sox: WAV Chunk fmt sox: WAV Chunk data sox: Reading Wave file: Microsoft PCM format, 1 channel, 44100 samp/sec sox: 88200 byte/sec, 2 block align, 16 bits/samp, 2041438 data bytes sox: 1020719 Samps/chans sox: Input file obd-demo.wav: using sample rate 44100 size shorts, encoding signed (2's complement), 1 channel sox: Output file obd-demo.alaw: using sample rate 8000 size bytes, encoding a-law, 1 channel sox: Output file: comment "Processed by SoX" sox: resample opts: Kaiser window, cutoff 0.80, beta 16.00 [root@host0040 test]# ls -ltrh total 2.9M -rwxr-xr-x 1 root root 725K Sep 14 06:32 obd-demo.mp3 -rw-r--r-- 1 root root 2.0M Sep 14 06:32 obd-demo.wav -rw-r--r-- 1 root root 181K Sep 15 16:57 obd-demo.alaw -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
On Wed, Sep 14, 2011 at 7:04 AM, Steve Edwards wrote: > On Wed, 14 Sep 2011, Kaushal Shriyan wrote: > >> I have carried out the below steps >> >> [root@host0040 test]# sox -V obd-demo.wav -r 8000 -b -t ul -c 1 >> obd-demo.alaw > >> sox: Output file obd-demo.alaw: using sample rate 8000 >> size bytes, encoding u-law, 1 channel > > Sox v14.x complains about the '-b.' > > You are encoding as u-law, but naming the file alaw. Since [a|u]law are > 'headerless' file formats, this will probably confuse Asterisk. > >> -rwxr-xr-x 1 root root 741459 Sep 14 06:32 obd-demo.mp3 >> -rw-r--r-- 1 root root 2041482 Sep 14 06:32 obd-demo.wav >> -rw-r--r-- 1 root root 185165 Sep 14 06:32 obd-demo.alaw >> >> Am i doing it correctly ? Please comment > > I've never used alaw (I'm in the US). > > I don't think an MP3 needs 'execute' permissions :) > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 Hi Steve, Please let me know the correct procedure to get .alaw file format since I belong to India region. Regards Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
On Wed, Sep 14, 2011 at 6:42 AM, Steve Edwards wrote: > On Wed, 14 Sep 2011, Kaushal Shriyan wrote: > >> Also please let me know the difference between .ulaw and .alaw format and >> is there a way i can play this file formats. > > alaw = Europe, ulaw = US & Japan > > Wikipedia has articles on both algorithms if you are interested in the > specifics. > > If you format your file correctly, Asterisk can play it. > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- 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 > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Hi Steve, I have carried out the below steps [root@host0040 test]# lame --decode obd-demo.mp3 obd-demo.wav input: obd-demo.mp3 (44.1 kHz, 1 channel, MPEG-1 Layer III) output: obd-demo.wav (16 bit, Microsoft WAVE) skipping initial 1105 samples (encoder+decoder delay) Frame# 887/886256 kbps [root@host0040 test]# sox -V obd-demo.wav -r 8000 -b -t ul -c 1 obd-demo.alaw sox: Detected file format type: wav sox: WAV Chunk fmt sox: WAV Chunk data sox: Reading Wave file: Microsoft PCM format, 1 channel, 44100 samp/sec sox: 88200 byte/sec, 2 block align, 16 bits/samp, 2041438 data bytes sox: 1020719 Samps/chans sox: Input file obd-demo.wav: using sample rate 44100 size shorts, encoding signed (2's complement), 1 channel sox: Output file obd-demo.alaw: using sample rate 8000 size bytes, encoding u-law, 1 channel sox: Output file: comment "Processed by SoX" sox: resample opts: Kaiser window, cutoff 0.80, beta 16.00 [root@host0040 test]# ls -ltr total 2932 -rwxr-xr-x 1 root root 741459 Sep 14 06:32 obd-demo.mp3 -rw-r--r-- 1 root root 2041482 Sep 14 06:32 obd-demo.wav -rw-r--r-- 1 root root 185165 Sep 14 06:32 obd-demo.alaw Am i doing it correctly ? Please comment Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
On Tue, Sep 13, 2011 at 6:47 PM, Matthew J. Roth wrote: > Kaushal, > > Your version of SoX does not have MP3 support. Since you have LAME > installed, use it as a first step to produce an intermediate file > that SoX supports. Then use SoX to convert the intermediate file > to the desired format. > > Step 1 > -- > > # lame --decode obd-demo.mp3 obd-demo.wav > input: obd-demo.mp3 (8 kHz, 1 channel, MPEG-2.5 Layer III) > output: obd-demo.wav (16 bit, Microsoft WAVE) > skipping initial 1105 samples (encoder+decoder delay) > Frame# 16818/16818 16 kbps > # file obd-demo.wav > obd-demo.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, > mono 8000 Hz > > > Step 2 > -- > > # sox -V obd-demo.wav -r 8000 -b -t ul -c 1 obd-demo.ulaw > sox: Detected file format type: wav > > sox: WAV Chunk fmt > sox: WAV Chunk data > sox: Reading Wave file: Microsoft PCM format, 1 channel, 8000 samp/sec > sox: 16000 byte/sec, 2 block align, 16 bits/samp, 19372126 data bytes > sox: 9686063 Samps/chans > sox: Input file obd-demo.wav: using sample rate 8000 > size shorts, encoding signed (2's complement), 1 channel > sox: Output file obd-demo.ulaw: using sample rate 8000 > size bytes, encoding u-law, 1 channel > sox: Output file: comment "Processed by SoX" > > > Regards, > > Matthew Roth Thanks Matthew Roth. It worked. Also please let me know the difference between .ulaw and .alaw format and is there a way i can play this file formats. Regards Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager Interface (AMI)
Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). Can someone please help me understand using Asterisk Manager Interface (AMI) available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide. Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
Hi, Can someone please comment about the below issue [root@host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root@host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root@host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root@host0040 kaushal]# sox -v 0.125 -V -t au -r 8000 -U -b -c 1 resample -ql -bash: obd-demo.ulaw: No such file or directory [root@host0040 kaushal]# sox -V obd-demo.mp3 -t au -r 8000 -U -b -c 1 obd-demo.ulaw resample -ql sox: Failed reading obd-demo.mp3: Do not understand foReply rmat type: mp3 [root@host0040 kaushal]# When i invoke the same obd-demo.mp3 it works perfectly fine host0040*CLI> channel originate DAHDI/g0/xx Application MP3Player /home/kaushal/obd-demo.mp3 [Sep 9 16:44:56] DEBUG[12691]: sig_pri.c:985 sig_pri_request: sig_pri_request 1 [Sep 9 16:44:56] DEBUG[12691]: sig_pri.c:6427 sig_pri_call: CALLER NAME: NUM: -- Requested transfer capability: 0x00 - SPEECH -- Launching MP3Player(/home/kaushal/obd-demo.mp3) on DAHDI/i1/9833754756-1 [root@host0040 ~]# rpm -qa | grep sox sox-12.18.1-1.el5_5.1 [root@host0040 ~]# rpm -qa | grep lame lame-3.98.4-1.el5.rf lame-devel-3.98.4-1.el5.rf [root@host0040 ~]# MP3 support in SoX is optional and requires access to either or both the external libmad and libmp3lame libraries. To see if there is support for Mp3 run sox -h and look for it under the list of supported file formats as "mp3". [root@host0040 ~]# sox -h sox: Version 12.18.1 Usage: [ gopts ] [ fopts ] ifile [ fopts ] ofile [ effect [ effopts ] ] gopts: -e -h -p -q -S -V fopts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -v volume -x effect: avg band bandpass bandreject chorus compand copy dcshift deemph earwax echo echos fade filter flanger highp highpass lowp lowpass mask mcompand noiseprof noisered pan phaser pick pitch polyphase rate repeat resample reverb reverse silence speed stat stretch swap synth trim vibro vol effopts: depends on effect Supported file formats: aiff al alsa au auto avr cdr cvs dat vms gsm hcom la lu maud nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw ub ul uw voc vorbis vox wav wve Which package contains libmad and libmp3lame libraries available on CentOS 5.6 Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound Dial
On Thu, Aug 25, 2011 at 5:18 AM, Kaushal Shriyan wrote: > On Wed, Aug 24, 2011 at 5:23 AM, Kaushal Shriyan > wrote: >> On Tue, Aug 23, 2011 at 12:09 PM, Faisal Hanif wrote: >>> U can also use VICIDIAL for it >> >> >> Hi Faisal >> >> Please help me understand the difference between VICIDIAL and >> astguiclient http://astguiclient.sourceforge.net. >> Are they both the same or interdependent on each other and also can i >> exclusively use it for my set up of 8 E1 PRI Lines meaning 240 bearer >> channels to make Outbound calls only and run a campaign --> meaning i >> have 200 phone numbers and a sound file of playtime of 30 secs. I need >> to dial out to all the 200 phone numbers and play the sound file >> concurrently at the same time. I have asterisk server version 1.8.5.0 >> on my server. >> >> Regards >> >> Kaushal >> > > Hi, > > Can someone please comment on my earlier email thread > > Regards > > Kaushal > Hi Again, Can someone please comment on my earlier email thread Regards Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound Dial
On Wed, Aug 24, 2011 at 5:23 AM, Kaushal Shriyan wrote: > On Tue, Aug 23, 2011 at 12:09 PM, Faisal Hanif wrote: >> U can also use VICIDIAL for it > > > Hi Faisal > > Please help me understand the difference between VICIDIAL and > astguiclient http://astguiclient.sourceforge.net. > Are they both the same or interdependent on each other and also can i > exclusively use it for my set up of 8 E1 PRI Lines meaning 240 bearer > channels to make Outbound calls only and run a campaign --> meaning i > have 200 phone numbers and a sound file of playtime of 30 secs. I need > to dial out to all the 200 phone numbers and play the sound file > concurrently at the same time. I have asterisk server version 1.8.5.0 > on my server. > > Regards > > Kaushal > Hi, Can someone please comment on my earlier email thread Regards Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound Dial
On Tue, Aug 23, 2011 at 12:09 PM, Faisal Hanif wrote: > U can also use VICIDIAL for it Hi Faisal Please help me understand the difference between VICIDIAL and astguiclient http://astguiclient.sourceforge.net. Are they both the same or interdependent on each other and also can i exclusively use it for my set up of 8 E1 PRI Lines meaning 240 bearer channels to make Outbound calls only and run a campaign --> meaning i have 200 phone numbers and a sound file of playtime of 30 secs. I need to dial out to all the 200 phone numbers and play the sound file concurrently at the same time. I have asterisk server version 1.8.5.0 on my server. Regards Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outbound Dial
Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). is there a utility available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide Regards Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libpri rpm version 1.4.12 for CentOS 5.6
Hi, Is libpri rpm version 1.4.12 for CentOS 5.6 made available ? [root@ ~]# rpm -qa | grep libpri libpri-1.4.11.5-1_centos5 [root@ ~]# cat /etc/redhat-release CentOS release 5.6 (Final) [root@ ~]# [root@ ~]# yum list updates | grep libpri [root@ ~]# Please suggest/guide further. Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error Code 101
On Mon, Jul 25, 2011 at 5:08 AM, Kaushal Shriyan wrote: > On Mon, Jul 25, 2011 at 4:39 AM, Kaushal Shriyan > wrote: >> Hi, >> >> I have 8 port PRI Sangoma Card connected to the Server running >> Asterisk 1.8.5 on CentOS 5.6. I see these errors in sangoma file under >> /var/log/asterisk/ >> http://pastebin.ubuntu.com/651451/ >> Error code 101 is "Message not compatible with call state". The >> explanation for this is "The remote equipment received an unexpected >> message that does not correspond to the current state of the >> connection. This is usually due to a D-channel error. >> >> Please suggest/guide and let me know if anyone needs any information >> about configs. >> >> Regards, >> >> Kaushal >> > > Hi, > > The versions are as below :- > > asterisk18.x86_64 1.8.5.0-1_centos5 > libpri-1.4.11.5-1_centos5.x86_64 > WANPIPE Release: 3.5.20 > > Regards, > > Kaushal > Hi, I have Package libpri-1.4.11.5-1_centos5.x86_64 already installed and latest version on CentOS 5.6, is there a rpm version of 1.4.12 for CentOS 5.6 as per http://downloads.asterisk.org/pub/telephony/libpri/ ? Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error Code 101
On Mon, Jul 25, 2011 at 4:39 AM, Kaushal Shriyan wrote: > Hi, > > I have 8 port PRI Sangoma Card connected to the Server running > Asterisk 1.8.5 on CentOS 5.6. I see these errors in sangoma file under > /var/log/asterisk/ > http://pastebin.ubuntu.com/651451/ > Error code 101 is "Message not compatible with call state". The > explanation for this is "The remote equipment received an unexpected > message that does not correspond to the current state of the > connection. This is usually due to a D-channel error. > > Please suggest/guide and let me know if anyone needs any information > about configs. > > Regards, > > Kaushal > Hi, The versions are as below :- asterisk18.x86_64 1.8.5.0-1_centos5 libpri-1.4.11.5-1_centos5.x86_64 WANPIPE Release: 3.5.20 Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error Code 101
Hi, I have 8 port PRI Sangoma Card connected to the Server running Asterisk 1.8.5 on CentOS 5.6. I see these errors in sangoma file under /var/log/asterisk/ http://pastebin.ubuntu.com/651451/ Error code 101 is "Message not compatible with call state". The explanation for this is "The remote equipment received an unexpected message that does not correspond to the current state of the connection. This is usually due to a D-channel error. Please suggest/guide and let me know if anyone needs any information about configs. Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk binaries on CentOS version 6
Hi, Any time line of availability of Asterisk binaries on CentOS version 6. Regards Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted
On Fri, Jul 1, 2011 at 11:13 AM, Kaushal Shriyan wrote: > Hi > > Please help me understand about the below issue ? > > [root@asterisk1 ~]# /etc/init.d/asterisk restart > Stopping safe_asterisk: [ OK ] > Shutting down asterisk: [ OK ] > Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open > files: cannot modify limit: Operation not permitted > [ OK ] > (reverse-i-search)`d': /etc/init.d/asterisk restart > [root@asterisk1 ~]# rpm -qa | grep asterisk > asterisk-sounds-core-en-gsm-1.4.21-1_centos5 > asterisk18-1.8.4.4-1_centos5 > asterisk18-core-1.8.4.4-1_centos5 > asterisk18-doc-1.8.4.4-1_centos5 > asterisk18-dahdi-1.8.4.4-1_centos5 > asterisk18-configs-1.8.4.4-1_centos5 > asterisk18-voicemail-1.8.4.4-1_centos5 > [root@asterisk1 ~]# uname -a > Linux asterisk1 2.6.18-238.el5 #1 SMP Thu Jan 13 15:51:15 EST 2011 > x86_64 x86_64 x86_64 GNU/Linux > [root@asterisk1 ~]# cat /proc/version > Linux version 2.6.18-238.el5 (mockbu...@builder10.centos.org) (gcc > version 4.1.2 20080704 (Red Hat 4.1.2-48)) #1 SMP Thu Jan 13 15:51:15 > EST 2011 > [root@asterisk1 ~]# cat /etc/redhat-release > CentOS release 5.6 (Final) > [root@asterisk1 ~]# > > Regards > > Kaushal > Hi Again, Can someone please reply on my earlier post to this emailing list. Regards Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted
Hi Please help me understand about the below issue ? [root@asterisk1 ~]# /etc/init.d/asterisk restart Stopping safe_asterisk:[ OK ] Shutting down asterisk:[ OK ] Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted [ OK ] (reverse-i-search)`d': /etc/init.d/asterisk restart [root@asterisk1 ~]# rpm -qa | grep asterisk asterisk-sounds-core-en-gsm-1.4.21-1_centos5 asterisk18-1.8.4.4-1_centos5 asterisk18-core-1.8.4.4-1_centos5 asterisk18-doc-1.8.4.4-1_centos5 asterisk18-dahdi-1.8.4.4-1_centos5 asterisk18-configs-1.8.4.4-1_centos5 asterisk18-voicemail-1.8.4.4-1_centos5 [root@asterisk1 ~]# uname -a Linux asterisk1 2.6.18-238.el5 #1 SMP Thu Jan 13 15:51:15 EST 2011 x86_64 x86_64 x86_64 GNU/Linux [root@asterisk1 ~]# cat /proc/version Linux version 2.6.18-238.el5 (mockbu...@builder10.centos.org) (gcc version 4.1.2 20080704 (Red Hat 4.1.2-48)) #1 SMP Thu Jan 13 15:51:15 EST 2011 [root@asterisk1 ~]# cat /etc/redhat-release CentOS release 5.6 (Final) [root@asterisk1 ~]# Regards Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Missing Config Files under /etc/asterisk
Thanks Jose it worked like a charm :) On Mon, May 16, 2011 at 9:52 PM, Jose P. Espinal wrote: >> >> Any thing i am missing ? Please suggest/guide. >> > > Hello Kaushal, try this: > yum install asterisk18-configs* > > > (You could do a 'yum list asterisk18*' to see what packages you might > want/need) > > Regards, > > > -- > Jose P. Espinal > http://www.eSlackware.com > IRC: Khratos @ #asterisk / -doc / -bugs > > > -- > _ > -- 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 > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Step by step guide
I have Digium Card -> Two (2) span digital T1/E1/J1/PRI PCI-Express x1 card On Mon, May 16, 2011 at 9:35 PM, Kaushal Shriyan wrote: > Hi, > > Are there step by step guide to configure Digium Card in Asterisk ? I > have done it using Sangoma Card. > Please suggest/guide. > > Regards, > > Kaushal > -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Step by step guide
Hi, Are there step by step guide to configure Digium Card in Asterisk ? I have done it using Sangoma Card. Please suggest/guide. Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Missing Config Files under /etc/asterisk
Hi I have followed https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS%2FRedHat%29, to my surprise there is only one config file by the name zapata.conf under /etc/asterisk/ There are no other config files. Any thing i am missing ? Please suggest/guide. Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
On Mon, May 2, 2011 at 7:00 PM, Jim Dickenson wrote: > On May 1, 2011, at 9:07 PM, Kaushal Shriyan wrote: > > Hi Jim, > > Thanks for the explanation, I have couple of questions here. > > 1) Does the xorcom box support *8 Port PRI E1 Interface*. ? > 2) Also the Primary and Secondary Asterisk Server can be any server which > will run Asterisk or AsteriskNow (http://www.asterisk.org/asterisknow) > Application and customizable or do i also need to buy this from Xorcom ? Not > sure i understand that. > 3) How does the xorcom box communicate with the Asterisk Server which do > not contain any PRI Card inside the system. > > Much Appreciated. > > Thanks and Regards, > > Kaushal > > > Yes Xorcom supports E1. > > You can run any version of Asterisk as far as I know. I have used 1.4.x and > ABE. The drivers are actually built in to Dahdi as supplied by Digium. > > The Xorcom box communicates to both system via USB cables, one connected to > each system. > > -- > Jim Dickenson > mailto:dicken...@cfmc.com > > Hi Jim Thanks for your reply. I found out from xorcom folks that we cannot re-program astribank, it is proprietary solution working on any given asterisk distribution out there. You can build linux HA cluster for asterisk then connect astribanks to the cluster. They recommend twinstar (http://www.xorcom.com/optional-extras/twinstar.html) if we need HA solution from Xorcom. Is there a way i can have my own customized Asterisk server behind Xorcom Astribank PRI box (http://www.xorcom.com/catalog/xr0111.html) and implement linux-ha ? Do you have any docs for setting up linux-ha using Astribank as the PRI box and Customized Asterisk Server (Primary and Secondary) Setup. Please let me know if you need more information about my specific requirement. Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
On Mon, May 2, 2011 at 1:46 AM, Jim Dickenson wrote: > > On May 1, 2011, at 10:09 AM, Kaushal Shriyan wrote: > > > > On Sun, May 1, 2011 at 10:14 PM, Jim Dickenson wrote: > >> Xorcom makes a box that connects via USB that can do failover. You connect >> the box to the two system via a USB cable to each system. When the box >> detects the primary system fails it switches over the the second one. No >> need for any extra hardware, except a USB cable. >> >> http://www.xorcom.com/catalog/xr0015.html >> >> http://www.xorcom.com/optional-extras/twinstar.html >> > > > Hi Jim, > > Thanks for sharing the technical details. Still not able to understand the > setup. Let me explain what i understand is the 8 PRI line would be connected > to the xorcom box and from there USB out would be connected to Primary > Asterisk Server and Secondary Asterisk Server. > > So we do not need any 8 port PRI Card on the Primary Asterisk Server and > Secondary Asterisk Server ? > > Please correct me if i am wrong. > > Thanks > > Kaushal > > > > Correct, there are no cards inside any system. You have an external box > that can have a combination of PRI, FXO and FXS ports; depending on need. > The external box is connected via USB to the two systems. The twinstar > option allows you to connect the external box to two systems via USB and > provides fall over from primary to secondary on failure of the primary. > > Hi Jim, Thanks for the explanation, I have couple of questions here. 1) Does the xorcom box support *8 Port PRI E1 Interface*. ? 2) Also the Primary and Secondary Asterisk Server can be any server which will run Asterisk or AsteriskNow (http://www.asterisk.org/asterisknow) Application and customizable or do i also need to buy this from Xorcom ? Not sure i understand that. 3) How does the xorcom box communicate with the Asterisk Server which do not contain any PRI Card inside the system. Much Appreciated. Thanks and Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
On Sun, May 1, 2011 at 10:14 PM, Jim Dickenson wrote: > Xorcom makes a box that connects via USB that can do failover. You connect > the box to the two system via a USB cable to each system. When the box > detects the primary system fails it switches over the the second one. No > need for any extra hardware, except a USB cable. > > http://www.xorcom.com/catalog/xr0015.html > > http://www.xorcom.com/optional-extras/twinstar.html > Hi Jim, Thanks for sharing the technical details. Still not able to understand the setup. Let me explain what i understand is the 8 PRI line would be connected to the xorcom box and from there USB out would be connected to Primary Asterisk Server and Secondary Asterisk Server. So we do not need any 8 port PRI Card on the Primary Asterisk Server and Secondary Asterisk Server ? Please correct me if i am wrong. Thanks Kaushal -- > Jim Dickenson > mailto:dicken...@cfmc.com > > CfMC > http://www.cfmc.com/ > > > > On Apr 30, 2011, at 8:31 PM, Kaushal Shriyan wrote: > > > > On Sun, May 1, 2011 at 8:48 AM, Michelle Dupuis wrote: > >> Yes that's it - one PRI line in, 2 out (one to the PRI card in each >> server). If you have lots of PRI lines, you may want to consider a >> dedicated PRI-to-SIP appliance.. >> > Hi, > > Thanks a Lot Michelle, Also please let me know the model/make for > dedicated PRI-to-SIP appliance. Would appreciate if you can share the > details along with the Network Diagram in case of 8 PRI Lines. > > Much appreciated. > > Regards, > > Kaushal > > > >> ____________ >> From: asterisk-users-boun...@lists.digium.com [ >> asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan [ >> kaushalshri...@gmail.com] >> Sent: Saturday, April 30, 2011 11:03 PM >> To: Asterisk Users List >> Subject: Re: [asterisk-users] HA Asterisk >> >> On Sun, May 1, 2011 at 2:13 AM, Michelle Dupuis > mdup...@ocg.ca>> wrote: >> There are lots out there, but here's the result of a quick search... >> >> http://www.zycon.com/News-Press-Releases/Read/RJ45-CAT5e-Line-Off-Line-Switch-Series-Offers-3-Control-Methods-R1357.html >> >> and the software to trigger the switch: >> www.generationd.com<http://www.generationd.com> >> >> >> >> Hi Michelle >> >> So what i understand is that the Single PRI Line from telco is connected >> to RJ45 (8 wire) A-B switched controllable by serial port and then there >> will be two patch cord from the A-B switch which will be connected to the 2 >> Asterisk Box containing PRI Card on each box. >> >> Please let me know if i am understanding you correctly or if you can help >> me with Network Diagram that would be really helpful. >> Also I have 8 PRI in my setup. How it would fit in this setup. The reason >> being we need to have atleast 320 Outbound Calls per min if i have 8 PRI >> Lines for our Voice Application. >> >> Regards, >> >> Kaushal >> >> -- >> _ >> -- 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 >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > _ > -- 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 > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- 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 > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
On Sun, May 1, 2011 at 8:48 AM, Michelle Dupuis wrote: > Yes that's it - one PRI line in, 2 out (one to the PRI card in each > server). If you have lots of PRI lines, you may want to consider a > dedicated PRI-to-SIP appliance.. > Hi, Thanks a Lot Michelle, Also please let me know the model/make for dedicated PRI-to-SIP appliance. Would appreciate if you can share the details along with the Network Diagram in case of 8 PRI Lines. Much appreciated. Regards, Kaushal > > From: asterisk-users-boun...@lists.digium.com [ > asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan [ > kaushalshri...@gmail.com] > Sent: Saturday, April 30, 2011 11:03 PM > To: Asterisk Users List > Subject: Re: [asterisk-users] HA Asterisk > > On Sun, May 1, 2011 at 2:13 AM, Michelle Dupuis mdup...@ocg.ca>> wrote: > There are lots out there, but here's the result of a quick search... > > http://www.zycon.com/News-Press-Releases/Read/RJ45-CAT5e-Line-Off-Line-Switch-Series-Offers-3-Control-Methods-R1357.html > > and the software to trigger the switch: > www.generationd.com<http://www.generationd.com> > > > > Hi Michelle > > So what i understand is that the Single PRI Line from telco is connected to > RJ45 (8 wire) A-B switched controllable by serial port and then there will > be two patch cord from the A-B switch which will be connected to the 2 > Asterisk Box containing PRI Card on each box. > > Please let me know if i am understanding you correctly or if you can help > me with Network Diagram that would be really helpful. > Also I have 8 PRI in my setup. How it would fit in this setup. The reason > being we need to have atleast 320 Outbound Calls per min if i have 8 PRI > Lines for our Voice Application. > > Regards, > > Kaushal > > -- > _ > -- 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 > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
On Sun, May 1, 2011 at 2:13 AM, Michelle Dupuis wrote: > There are lots out there, but here's the result of a quick search... > > http://www.zycon.com/News-Press-Releases/Read/RJ45-CAT5e-Line-Off-Line-Switch-Series-Offers-3-Control-Methods-R1357.html > > and the software to trigger the switch: > www.generationd.com > > Hi Michelle So what i understand is that the Single PRI Line from telco is connected to RJ45 (8 wire) A-B switched controllable by serial port and then there will be two patch cord from the A-B switch which will be connected to the 2 Asterisk Box containing PRI Card on each box. Please let me know if i am understanding you correctly or if you can help me with Network Diagram that would be really helpful. Also I have 8 PRI in my setup. How it would fit in this setup. The reason being we need to have atleast 320 Outbound Calls per min if i have 8 PRI Lines for our Voice Application. Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
On Sat, Apr 30, 2011 at 11:47 AM, Alex Balashov wrote: > On 04/30/2011 02:13 AM, RAJNIKANT VANZA wrote: > > Hi Kaushal, >> >> I have done HA for Asterisk servers as well as SIP Server (kamailio). >> >> Please write your detail requirement. >> >> -> how many Asterisk Sever require for HA? >> -> How much down time acceptable during Asterisk Sever failover? >> -> Which type Asterisk Sever Failover u required? >> >> Send me your detail requirement and answer of above question ASAP. >> > > Requests for additional details are a lot more persuasive when conveyed in > a semi-literate manner. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > > -- > _ > -- 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 > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Hi, Can someone please answer the rest of the questions in my earlier post to this Mailing List.? Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HA Asterisk
Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf, but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA -> High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems it would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive application. Please suggest/guide. Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware Server Configuration/8 or 4 port PRI Card
Hi, Can someone please recommend me the Hardware Server Configuration/8 or 4 port PRI Card to make Outbound Call at the rate of around 320 outbound Calls/min ? Thanks and Regards, Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configure IVR(Inbound and Outbound)
On Wed, Apr 20, 2011 at 2:56 PM, Gopalakrishnan A.N wrote: > Outbound call only you can make like dial out rules, IVR is only for > Inbound calls. In outbound I dont how you are asking this IVR facility. You > mean like voice broadacasting? like dial the number and playing a voice > file? > > Hi Gopal, Yes in those lines. Regards, Kaushal > > On Wed, Apr 20, 2011 at 2:14 PM, Kaushal Shriyan > wrote: > >> Hi Gopal, >> >> Is there a way to Configure OutBound IVR. Correct me if i might have >> missed reading the URL you pasted. >> >> Thanks >> >> Kaushal >> >> >> On Wed, Apr 20, 2011 at 1:04 PM, Gopalakrishnan A.N wrote: >> >>> Try this >>> http://www.freepbx.org/support/documentation/administration-guide/creating-an-ivr >>> >>> On Wed, Apr 20, 2011 at 8:11 AM, Kaushal Shriyan < >>> kaushalshri...@gmail.com> wrote: >>> >>>> Hi, >>>> >>>> Is there a step by step guide to Configure IVR(Inbound and Outbound) in >>>> AsteriskNow using FreePBX ? >>>> >>>> Thanks >>>> >>>> Kaushal >>>> >>>> -- >>>> _ >>>> -- 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 >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> >>> >>> >>> -- >>> Thank you with regards, >>> Gopalakrishnan A.N. >>> VoIP call - sip:sai...@gtalk2voip.com >>> >>> >>> >> > > > -- > Thank you with regards, > Gopalakrishnan A.N. > VoIP call - sip:sai...@gtalk2voip.com > > > -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configure IVR(Inbound and Outbound)
Hi, Is there a step by step guide to Configure IVR(Inbound and Outbound) in AsteriskNow using FreePBX ? Thanks Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A101DE Sangoma Card in AsteriskNow 1.7.1
On Mon, Apr 18, 2011 at 11:42 PM, Kaushal Shriyan wrote: > Hi, > > I have A101DE Sangoma Card( > http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a101.html > ) > > lspci shows as 03:04.0 Network controller: Sangoma Technologies Corp. > A200/Remora FXO/FXS Analog AFT card > [root@asterisk ~]# lspci -vvv -s 03:04.0 > 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS > Analog AFT card > Subsystem: Unknown device a111:3713 > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- > Stepping- SERR- FastB2B- > Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- > SERR- Latency: 255 (1250ns min, 3750ns max) > Interrupt: pin A routed to IRQ 185 > Region 0: Memory at fd8f8000 (32-bit, non-prefetchable) [size=32K] > [root@asterisk ~]# > > Please suggest/guide > > Thanks > > Kaushal > > Hi, I have A101DE Sangoma Card( http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a101.html ) [root@asterisk ~]# wanrouter list Devices currently active: wanpipe1 [root@asterisk ~]# wanrouter status Devices currently active: wanpipe1 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1| N/A | A101/1D/A102/2D/4/4D/8| 185 | 4 | 1| N/A | 0 | Wanrouter Status: Device name | Protocol | Station | Status| wanpipe1| AFT TE1 | N/A | Disconnected | [root@asterisk ~]# wanrouter hwprobe --- | Wanpipe Hardware Probe Info | --- 1 . AFT-A101-SH : SLOT=4 : BUS=3 : IRQ=185 : CPU=A : PORT=1 : HWEC=32 : V=37 Card Cnt: A101-2=1 [root@asterisk ~]# [root@asterisk ~]# lspci 00:00.0 Host bridge: ATI Technologies Inc RS480 Host Bridge (rev 01) 00:01.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:04.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:05.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:12.0 IDE interface: ATI Technologies Inc IXP SB400 Serial ATA Controller 00:13.0 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller 00:13.1 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller 00:13.2 USB Controller: ATI Technologies Inc IXP SB400 USB2 Host Controller 00:14.0 SMBus: ATI Technologies Inc IXP SB400 SMBus Controller (rev 10) 00:14.1 IDE interface: ATI Technologies Inc IXP SB400 IDE Controller 00:14.3 ISA bridge: ATI Technologies Inc IXP SB400 PCI-ISA Bridge 00:14.4 PCI bridge: ATI Technologies Inc IXP SB400 PCI-PCI Bridge 00:14.5 Multimedia audio controller: ATI Technologies Inc IXP SB400 AC'97 Audio Controller (rev 01) 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:05.0 VGA compatible controller: ATI Technologies Inc RS480 [Radeon Xpress 200G Series] 01:05.1 Display controller: ATI Technologies Inc Radeon Xpress Series (RS480) 02:00.0 PCI bridge: PLX Technology, Inc. PEX8112 x1 Lane PCI Express-to-PCI Bridge (rev aa) 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card 04:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit Ethernet PCI Express (rev 20) 05:0a.0 Ethernet controller: Accton Technology Corporation EN-1216 Ethernet Adapter (rev 11) [root@asterisk ~]# lspci -vvv -s 03:04.0 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card Subsystem: Unknown device a111:3713 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- SERR- -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A101DE Sangoma Card in AsteriskNow 1.7.1
Hi, I have A101DE Sangoma Card( http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a101.html ) lspci shows as 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card [root@asterisk ~]# lspci -vvv -s 03:04.0 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card Subsystem: Unknown device a111:3713 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- SERR- -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card
On Wed, Apr 13, 2011 at 9:23 PM, Tim Nelson wrote: > > On Wed, Apr 13, 2011 at 8:07 PM, satish patel wrote: > >> Try dmesg command >> >> root@:~# dmesg | grep -i Sangoma >> [ 2303.473601] WANPIPE(tm) Hardware Support Module 3.5.19.0 (c) 1994-2010 >> Sangoma Technologies Inc >> [ 2303.481115] WANPIPE(tm) Interface Support Module 3.5.19.0 (c) 1994-2010 >> Sangoma Technologies Inc >> [ 2303.494824] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.19.0 (c) >> 1994-2010 Sangoma Technologies Inc >> [ 2303.506498] WANPIPE(tm) Socket API Module 3.5.19.0 (c) 1994-2010 >> Sangoma Technologies Inc >> [ 2303.525334] WANPIPE(tm) WANEC Layer 3.5.19.0 (c) 1995-2006 Sangoma >> Technologies Inc. >> >> > Hi Satish > > > dmesg | grep -i Sangoma does not show anything > > > If lspci isn't showing the device, then your board is not recognizing it. > Try another slot, and make sure the card is seated 100% tightly into the > slot. > > And, of course, make sure the system is powered off while doing this. > 99.999% of people already know this, but I've been bitten by the other > 0.001% that don't. :-) > > If you're still not showing it, a call to Sangoma support is probably in > order. They are top notch. > > --Tim > > -- > _ > -- 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 > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Hi I get anaconda.log:23:07:59 DEBUG : ignoring driverless device Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card Linux asterisk 2.6.18-194.11.1.el5 #1 SMP Tue Aug 10 19:05:06 EDT 2010 x86_64 x86_64 x86_64 GNU/Linux lspci | grep sangoma does not return anything Please suggest further. Thanks Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card
On Wed, Apr 13, 2011 at 8:07 PM, satish patel wrote: > Try dmesg command > > root@:~# dmesg | grep -i Sangoma > [ 2303.473601] WANPIPE(tm) Hardware Support Module 3.5.19.0 (c) 1994-2010 > Sangoma Technologies Inc > [ 2303.481115] WANPIPE(tm) Interface Support Module 3.5.19.0 (c) 1994-2010 > Sangoma Technologies Inc > [ 2303.494824] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.19.0 (c) > 1994-2010 Sangoma Technologies Inc > [ 2303.506498] WANPIPE(tm) Socket API Module 3.5.19.0 (c) 1994-2010 Sangoma > Technologies Inc > [ 2303.525334] WANPIPE(tm) WANEC Layer 3.5.19.0 (c) 1995-2006 Sangoma > Technologies Inc. > > Hi Satish dmesg | grep -i Sangoma does not show anything -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card
Hi, I have Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card installed on the box. *Its not detected.* Details are as below :- [root@asterisk ~]# lspci 00:00.0 Host bridge: ATI Technologies Inc RS480 Host Bridge (rev 01) 00:01.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:04.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:05.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:12.0 IDE interface: ATI Technologies Inc IXP SB400 Serial ATA Controller 00:13.0 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller 00:13.1 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller 00:13.2 USB Controller: ATI Technologies Inc IXP SB400 USB2 Host Controller 00:14.0 SMBus: ATI Technologies Inc IXP SB400 SMBus Controller (rev 10) 00:14.1 IDE interface: ATI Technologies Inc IXP SB400 IDE Controller 00:14.3 ISA bridge: ATI Technologies Inc IXP SB400 PCI-ISA Bridge 00:14.4 PCI bridge: ATI Technologies Inc IXP SB400 PCI-PCI Bridge 00:14.5 Multimedia audio controller: ATI Technologies Inc IXP SB400 AC'97 Audio Controller (rev 01) 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:05.0 VGA compatible controller: ATI Technologies Inc RS480 [Radeon Xpress 200G Series] 01:05.1 Display controller: ATI Technologies Inc Radeon Xpress Series (RS480) 02:00.0 PCI bridge: PLX Technology, Inc. PEX8112 x1 Lane PCI Express-to-PCI Bridge (rev aa) 04:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit Ethernet PCI Express (rev 20) 05:0a.0 Ethernet controller: Accton Technology Corporation EN-1216 Ethernet Adapter (rev 11) [root@asterisk ~]# cat /etc/redhat-release CentOS release 5.5 (Final) [root@asterisk ~]# asterisk -v Asterisk 1.6.2.11, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect. [root@asterisk ~]# Please suggest/guide Thanks Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk
On Sun, Mar 20, 2011 at 8:02 PM, John Novack wrote: > > > Kaushal Shriyan wrote: > > Hi, > > I have couple of questions regarding Asterisk. > > a) Does it has Automated Dialing Feature like dialing 1000 and 1000 of > phone numbers? > > NO > And if you try and make it dial that way, your gonads will wither away and > fall off > > b) Does it Support VoiceXML ? > c) What PRI Card is recommended for using Asterisk ? > > Thanks > > Kaushal > > > > > > -- > _ > -- 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 > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > > Dog is my Co-pilot > > Hi Again, I have not got convincing reply for my query to this Mailing List. Can some one please pitch in to clarify my questions? Thanks Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk
Hi, I have couple of questions regarding Asterisk. a) Does it has Automated Dialing Feature like dialing 1000 and 1000 of phone numbers? b) Does it Support VoiceXML ? c) What PRI Card is recommended for using Asterisk ? Thanks Kaushal -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
