Re: [asterisk-users] Cisco IP 8841 asterisk integration

with Asterisk. Regards. On Tue, 6 Dec 2016, 1:40 p.m. Toshaan Bharvani | VanTosh, < tosh...@vantosh.com> wrote: On 05/12/16 17:57, Gopalakrishnan N wrote: > True agree, problem is somehow the people purchased am supporting to > overcome that. Trying level best... around 2

Re: [asterisk-users] Cisco IP 8841 asterisk integration

TrueAgree. :) On Mon, Dec 5, 2016 at 11:37 PM wrote: > > True agree, problem is somehow the people purchased am > > supporting to overcome that. Trying level best... around 20 > > phones has been purchased > > Ah, yes, the "we purchased these without consulting you, but it

Re: [asterisk-users] Cisco IP 8841 asterisk integration

True agree, problem is somehow the people purchased am supporting to overcome that. Trying level best... around 20 phones has been purchased 😔😔 On Mon, 5 Dec 2016, 8:55 p.m. Victor Villarreal, wrote: > With all the money you plan to invest in firmware, licenses, etc., you > have bou

Re: [asterisk-users] Cisco IP 8841 asterisk integration

e of upgrading the Cisco build to the SIP build > on Cisco 7641 handsets, which have 2 similar builds, but none of the > techniques seemed to apply this time around. > > Cheers, > Steve > > > On Sun, 4 Dec 2016 at 16:03 Gopalakrishnan N > wrote: > > Can't

Re: [asterisk-users] Cisco IP 8841 asterisk integration

ch like the Cisco SPA handsets. > > I also ended up with a non-3PCC handset and it is useless, and as far as I > can tell they cannot be re-flashed. > > Cheers, > Steve > > > > On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan N > wrote: > > Anyone tried integrat

[asterisk-users] Cisco IP 8841 asterisk integration

Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to upload woth TFTP due to some reason it's getting failed. Do I need to load 3pcc firmware or anyway to Configure from the phone itself or from the GUI?

[asterisk-users] Cisco IP phone serup

Hi, I have cisco 8841 IP phone. could someone light up how to configure with Asterisk. Thanks in advance. Regards, Gopal . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk c

Re: [asterisk-users] NAT on IPsec Tunnel

Finally got it worked, the issue was E164 callerid format, where i set it up, after removing the E164 format its was thru. Regards On Fri, Feb 12, 2016 at 9:31 PM Gopalakrishnan N < gopalakrishnan...@gmail.com> wrote: > Now incoming works fine, this is because of my SonicWALL firmw

Re: [asterisk-users] NAT on IPsec Tunnel

Now incoming works fine, this is because of my SonicWALL firmware issue, tried with different SonicWALL inbound works. But for outbound am getting 408 request time out error in the NAT on VPN tunnel. On Fri, Feb 12, 2016 at 3:50 AM Gopalakrishnan N < gopalakrishnan...@gmail.com> wrote: &g

[asterisk-users] Asterisk SIP UUI Protocol

Hi, I came thru ISDN UUI (User-User Information) protocol which is defined in this RFC - http://www.ietf.org/id/draft-ietf-cuss-sip-uui-17.txt But I don't understand how to use this with Asterisk. Any idea would be much appreciated. Thanks. Gopal. --

Re: [asterisk-users] log caller hangup events

> *From:* asterisk-users-boun...@lists.digium.com < > asterisk-users-boun...@lists.digium.com> on behalf of Gopalakrishnan N < > gopalakrishnan...@gmail.com> > *Sent:* Monday, August 18, 2014 4:13 AM > *To:* Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] log caller hangup events

Hi, You can use Hangup handler. May be this post can you help you, http://gblades.blogspot.in/2013/07/how-to-get-sip-response-code-in.html Regards On Mon, Aug 18, 2014 at 9:45 AM, Paul Greenberg wrote: > All, > > > I would like to log a message whenever a party hangs up a call or > session,

Re: [asterisk-users] Question about SIP Dial

It supposed to be like this Dial(SIP/${EXTEN}#ip.add.re.ss) Regards On Fri, Aug 15, 2014 at 6:20 AM, CDR wrote: > In channel PJSIP I use this format > Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss) > what would be the equivalent of this format in old SIP? > I tried > Dial(SIP/peer/${EXTEN}@ip.

[asterisk-users] Concurrent Calls via Manager Originate

Can we have concurrent calls via asterisk manager interface, lets say around 1000 or 1000+ concurrent calls. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live i

Re: [asterisk-users] PAGI

Thanks Johan. Are you using this application for any credit card processing? On Fri, Apr 4, 2014 at 5:29 PM, Johan Wilfer wrote: > 2014-04-03 18:58, Gopalakrishnan N skrev: > > Hi, >> >> Anybody using PAGI scripts, >> http://marcelog.github.io/art

[asterisk-users] PAGI

Hi, Anybody using PAGI scripts, http://marcelog.github.io/articles/pagi_tutorial_create_voip_telephony_application_for_asterisk_with_agi_and_php.html Would like to know the feasibility to build a IVR solutions. Regards -- _ --

[asterisk-users] Enterprise VoIP Trunk

Am looking for a service provider who can provide enterprise SIP trunk with 100 channels concurrent sessions. I see some like Inphonex, Broadvoice... and etc Is there any suggestions for the service providers. Regards -- _

Re: [asterisk-users] Asterisk Crash in asterisk 10.0.0.

Enable debugging module and backtrace and re-compile so that you will bactrace of the crash logs. Regards On 14 Feb 2014 10:29, "Arun Ram" wrote: > Hi guys, > I need a desperate help from you regarding this asterisk crash issue. > > > > On Thu, Feb 13, 2014 at 5:48 PM, Arun Ram wrote: > >> Hi,

Re: [asterisk-users] SIP OPTIONS "storm"?

SIP options message is due to check the peer registration is keepalive. As per my understanding it might be because of network flap may be wireshark trace can give you any clue. Regards On 13 Feb 2014 23:41, "Tim Nelson" wrote: > Greetings- > > I recently experienced an odd situation. I have an

[asterisk-users] Voice XML Asterisk Integration

Which is the best way around to integrate Asterisk with VoiceXML like VoiceGlue...! Am using Asterisk 11.2.1. Regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Integration of OpenVXI

Anyone using Voiceglue with latest Asterisk 11.6 certified version? On Mon, Jun 20, 2011 at 10:00 PM, Jean-Denis Girard wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Hi, > > Le 20/06/2011 04:40, Gopal krishnan a écrit : > > Have anybody integrated > > OpenVXI http://www.speech.cs.

Re: [asterisk-users] Change the preferred audio playback format

Hope basically depends on the codec Asterisk will playback the file automatically On 23 Jan 2014 19:25, "Gareth Blades" wrote: > On 23/01/14 13:38, Ishfaq Malik wrote: > >> Hi >> >> Is there any way to change the preferred audio playback format in >> asterisk (I'm using 1.8.25.0) >> i.e. first ch

Re: [asterisk-users] AGI Script not working

Thanks... I got it working actually I found with this command /usr/bin/perl -d from this I got to know that my library is missing and installed Asterisk-perl module and now its fine. Once again thank you. On Mon, Dec 2, 2013 at 3:05 PM, Gopalakrishnan N < gopalakrishnan...@gmail.com>

Re: [asterisk-users] AGI Script not working

ich Perl AGI library > are you using? > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N > Sent: Saturday, November 30, 2013 1:27 AM > To: Asterisk Users Ma

Re: [asterisk-users] Answering agent

Alao enable cel table that will have all the information On 29 Nov 2013 23:25, "Todd R." wrote: > I do this by writing custom CDR. I write the agents extension write into > the CDR records. This makes is easy to just parse through the CDR and get > all the info you need about the call. > > Google

[asterisk-users] AGI Script not working

I have a Perl AGI script updating some values to database like recorded file path, unique ID and callerid. When I run the script with test dialplan, its not updating to database. Whereas database connection is fine, when I run agi debug I see only Tx packets not Rx packets, firewall is also OFF.

Re: [asterisk-users] Sangoma transcoding card bug - drops audio samples

If you are getting like this dropped packets then nothing to worry.. thisis just an cli message in my case I face this but there is no voice delay in actual call. On 22 Nov 2013 21:11, "Eric Wieling" wrote: > Are you getting errors like this? > > > > [Nov 22 10:39:36] WARNING[6307][C-09a1

[asterisk-users] Channel not releasing immediately for Attended Transfer

I have a situation where Asterisk is not releasing the channel for Attended transfer immediately once I transferred and hangup from my side. The call is still ongoing and disconnecting after the third party disconnected. I see that its bug in the Asterisk, but not sure its fixed in version 11.2.1.

Re: [asterisk-users] Multi-Voicemail Message?

You can have something like this, exten => _,1, Answer exten => _, 2, voicemail ($EXTEN) On 25 Sep 2013 05:04, "Tim Nelson" wrote: > Greetings- > > I have an odd scenario where I need to dial an extension (lets call it > 555), the system prompts for a list of voicemail boxes, then once c

[asterisk-users] Bad Magic Internal Error

What does this mean of bad magic internal error, SIP to SIP calling is fine, when I use SIP via GSM I have this, and asterisk restarts automatically. Asterisk version which am using is 11.1.2. Regards -- _ -- Bandwidth and Coloc

[asterisk-users] G729 CPU Utilization

Hi, How much CPU utilization will it take when I use G729 transcoding via hardware based transcoder. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introduct

Re: [asterisk-users] Kepress while on Queue

also if am not wrong queue timeout will also applicable for this.. ! On Wed, Aug 28, 2013 at 11:37 PM, Gopalakrishnan N < gopalakrishnan...@gmail.com> wrote: > oh great thanks... > > > On Wed, Aug 28, 2013 at 1:12 PM, Satish Barot > wrote: > >> Yes you can. Ch

Re: [asterisk-users] Kepress while on Queue

send caller to different > queue from this context. > > --Satish Barot > Ahmedabad, India. > +919978599700 > > > On Wed, Aug 28, 2013 at 12:59 AM, Gopalakrishnan N < > gopalakrishnan...@gmail.com> wrote: > >> Hi, >> >> Will Keypress option will wo

[asterisk-users] Kepress while on Queue

Hi, Will Keypress option will work when am in the queue and hearing MoH? Lets say a caller is waiting in queue and while he is hearing MoH, can he key in some DTMF and go to some other queue? is that possible? Regards -- _ -- Ba

Re: [asterisk-users] How to get the original SIP result code

You can use AMI Commands and run sip set debug from that you have to capture the response code. http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command Regards, On Thu, Aug 22, 2013 at 10:43 PM, Mordechay Kaganer wrote: > B.H. > > Hello, i'm using AMI Originate action (with asyn

Re: [asterisk-users] Ingress and Egress

Basically I have some background noise like keyboard stoke or clicking sound in random basis, I need to measure that, when I check my IPLC its fine, and with my Telco service provider its fine... So am trying to conclude with some solution... trying to identify the root cause. Any advice would be

[asterisk-users] Ingress and Egress

Hi, Can Ingress and Egress can be used in Asterisk, so that Jitter can be calculated...! Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

ster = on server 1 >> >> On 8/18/2013 6:29 PM, Gopalakrishnan N wrote: >> >> Even I tried the type as friend.. but no use... >> >> >> On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N < >> gopalakrishnan...@gmail.com> wrote: >&

Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

Thanks for the comments. Without changing anything, adding fromuser=usman02 in both side worked for me.. Thanks. On Mon, Aug 19, 2013 at 1:01 AM, Andrew Colin wrote: > change server two to host = dynamic > > then add register = on server 1 > > On 8/18/2013 6:29 PM, G

Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

Even I tried the type as friend.. but no use... On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N < gopalakrishnan...@gmail.com> wrote: > Hi, > > Am making a simple SIP trunk between two Asterisk server, > > Server 1 > sip.conf > [usman02] > type=peer > usernam

[asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes inse

[asterisk-users] Random dead calls

Hi, Am getting dead or silence calls at sometimes for my agents, when I checked my CDR the caller-id shows my vendor's name and some shows as real customer name. When I call back again the real customer's number its reaching, the answering machine owned by customer. I have a confusion, or how to

Re: [asterisk-users] auto answer

If am not wrong even without doing any setting in asterisk side, if the phone has Auto Answer it works.. ! Correct me if am wrong. On Wed, Jul 17, 2013 at 9:14 PM, Steve Edwards wrote: > Please don't top post. > > > On Wed, 17 Jul 2013, bilal ghayyad wrote: > > So it is not at asterisk configu

Re: [asterisk-users] auto answer

yes its not asterisk configuration, its phone feature and phone configuration. On Wed, Jul 17, 2013 at 3:27 PM, bilal ghayyad wrote: > So it is not at asterisk configuration? > > Regards > Bilal > > -- > *From:* A J Stiles > > *To:* bilal ghayyad ; Asterisk Users

[asterisk-users] FLAC script to convert from wav to FLAC and also with other 3 to 4 formats

Hi, Below link is the script which i found while surfing, this script basically converts your voice file to flac format, where the file is reduced to 50%. http://legroom.net/files/software/convtoflac.sh The quality is really good, I tested. this... In large production environment this script ca

Re: [asterisk-users] External Recording Server for Asterisk Voicemail

If you want to store in external, why can't you have a NAS device and mount to Asterisk server, let the mounted be a part in asterisk.conf, so that voicemail will get recorded in external server... Will it makes sense... ! Thanks. On Mon, Jul 15, 2013 at 4:19 PM, Amit Salunkhe wrote: > Hello A

Re: [asterisk-users] Executing Script after MixMonitor is called

; -- fixed jitterbuffer destroyed on channel SIP/4092-003b == MixMonitor close filestream == *Executing [/root/flac.sh 4090-2013-07-05_14:43:11-OUT]* == End MixMonitor Recording SIP/4092-003b But the file is not converted, I suspect it could be a path issue. Regards On Fri, Jul

Re: [asterisk-users] Asterisk crash

Ok thanks posting now On 5 Jul 2013 03:09, "Matthew Jordan" wrote: > > On Thu, Jul 4, 2013 at 3:30 PM, Gopalakrishnan N < > gopalakrishnan...@gmail.com> wrote: > >> Suddenly my asterisk restarted automatically and came up in seven seconds, >> >&g

[asterisk-users] Asterisk crash

Suddenly my asterisk restarted automatically and came up in seven seconds, While checking core dump I see some message related to snmp. No symbol table info available. #5 0x7fc7e6249faa in agent_thread (arg=) at snmp/agent.c:206 __PRETTY_FUNCTION__ = "agent_thread" #6 0x0056dd0b in du

Re: [asterisk-users] Executing Script after MixMonitor is called

TRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}) exten => _4X.,n,Dial(SIP/${EXTEN},30) exten => _4X.,n,Hangup Regards On 4 Jul 2013 11:18, "Satish Barot" wrote: > On Thu, Jul 4, 2013 at 1:30 AM, Gopalakrishnan N < > gopalakrishnan...@gmail.com>

Re: [asterisk-users] Executing Script after MixMonitor is called

to mp3. I pass ${FILENAME} as an argument to my script. >> * >> >> *You should have something like >> *MixMonitor(filename.wav,m,/PathToYourScript/YourScriptName^filename.wav) in >> your dialplan. >> >> Hope this helps. >> >> --Satish Barot >&

Re: [asterisk-users] Asterisk trunking between two location

By having different server, i made it work. I suspect some network issue... On Wed, Jul 3, 2013 at 3:27 AM, Asghar Mohammad wrote: > make a call and post cli log > > > On Tue, Jul 2, 2013 at 11:54 PM, Gopalakrishnan N < > gopalakrishnan...@gmail.com> wrote: >

Re: [asterisk-users] Asterisk trunking between two location

fy=yes > nat=force_rport,comedia > disallow=all > allow=g729 > allow=ulaw > allow=alaw > > 2st location dialplan > exten => _2XXX,1,Dial(SIP/india/${EXTEN} <http://10.30.2.5/$%7BEXTEN%7D>) > exten => _2XXX,n,Hangup > > then you should handle the call when it arr

Re: [asterisk-users] Asterisk trunking between two location

al(SIP/trunka/${EXTEN}) on side b and exten => > _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a. > make a call from a to b and one from b to and post cli log here or upload > anyware else. > > > On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N < > gopalakrishnan...@gmail.com>

Re: [asterisk-users] Asterisk trunking between two location

ress used for staticly > defined > ; hosts. This helps avoid the > configuration > ; error of allowing your users to register > at > ; the same address as a SIP provider. > > > > On Tue, Jul 2, 2013 at 10:

Re: [asterisk-users] Asterisk trunking between two location

t=dynamic? > both servers are on 10.10.10.0 ? if no then check your deny permit seting. > > > On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N < > gopalakrishnan...@gmail.com> wrote: > >> Also tried one more scenario, particularly from one IP to other IP not >> regi

Re: [asterisk-users] Asterisk trunking between two location

is no packet loss.. with mtr it is fine, tracepath is fine, ping is fine... :( On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N < gopalakrishnan...@gmail.com> wrote: > Am using Asterisk 11.2 in one location and 11.1 in another location. > > when I trunk between two servers

[asterisk-users] Asterisk trunking between two location

Am using Asterisk 11.2 in one location and 11.1 in another location. when I trunk between two servers, the status is unreachable. But with different server with 11.2 and 11.2 it works fine. I tried both IAX and SIP. the trunk in sip.conf what i have is, [serverb] type=friend username=serverb se

Re: [asterisk-users] IAX2 netsock error with name resolution

After changing my dialplan as suggested, there is no socket error, but getting Busy/Congested, and the call is hanging up, let me check that part... Earlier my dialplan was, ;exten => _2XXX,1,Dial(SIP/${EXTEN}@${MANIAX},30) and I changed like this exten => _2XXX,1,Dial(${MANIAX}/${EXTEN},30) whe

[asterisk-users] IAX2 netsock error with name resolution

Am getting netsock error like this when using IAX2, Connected to Asterisk 11.2.1 currently running on indiaprimaryast01 (pid = 4270) == Using SIP RTP CoS mark 5 -- Executing [2001@Test:1] Dial("SIP/4090-0005", "SIP/2001@IAX2/IND-MAN,30") in new stack [Jun 23 06:31:36] NOTICE[4383][C-

Re: [asterisk-users] Asterisk Queue Frame

: > On Thu, Jun 20, 2013 at 6:55 PM, Gopalakrishnan N < > gopalakrishnan...@gmail.com> wrote: > >> What happens when we increase the queue frame size in channels.c >> >> if ((queued_frames + new_frames > 128 || queued_voice_frames + >> new_voice_frames >

[asterisk-users] Asterisk Queue Frame

What happens when we increase the queue frame size in channels.c if ((queued_frames + new_frames > 128 || queued_voice_frames + new_voice_frames > 96)) { Be default it is 128 and 96 if i increase it to 256 and 192 what will happen? will it impact to default behavior? Regards, Gopal. --

Re: [asterisk-users] Executing Script after MixMonitor is called

Hi Satish, I tried with sox, without any parameter, just sox filename.wav to filename.mp3, in linux shell prompt... the file is been converted... Now If i want to run that command using dialplan, MixMonitor(filename.wav,m) Monitor_Exec(sox filename.wav filename.mp3) Or to use System command? R

[asterisk-users] Asterisk HA

I was go through'ing the following links for HA, https://wiki.asterisk.org/wiki/display/TOP/Failover+-+Linux - which doesn't have file syncing. https://www.johncahill.net/wiki/index.php/2_Node_Active/Passive_cluster - this one has file syncing with pacemaker Any other HA applications available o

[asterisk-users] Codec Mismatch

Sometimes in huge call volume am facing this type of error, [Jun 4 08:42:46] WARNING[8459][C-79fa]: channel.c:5075 ast_write: Codec mismatch on channel Local/8038@xss-call-out-4774;1 setting write format to slin from ulaw native formats (ulaw) [Jun 4 08:43:04] WARNING[8285][C-79da]:

Re: [asterisk-users] Most suitable version for Production ENV

Asterisk 1.8 is stable On 1 Jun 2013 16:40, "luke devon" wrote: > Hi > > As I seen on the Asterisk web site , there is packages called , > > AsteriskLatest Version - 11.4.0 > > asterisk-11-current.tar.gz > and > > a

Re: [asterisk-users] Queue Periodic Announce not working...

It works. Thanks On 30 May 2013 19:39, "Doug Lytle" wrote: > >> periodic-announce = /var/lib/asterisk/sounds/en/test/AVG-15.wav > > Try it without the .wav > > Doug > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neith

[asterisk-users] Queue Periodic Announce not working...

I am having a queue where included periodic announce like the below, [test] context = default member = Agent/1001 member = Agent/1002 music = default strategy = rrmemory ringinuse = no timeout = 15 retry = 1 maxlen = 0 joinempty = yes leavewhenempty = no periodic-announce = /var/lib/asterisk/sound

Re: [asterisk-users] DTMF recognized after call establishment

Let me try with dtmfmode as auto... On 28 May 2013 19:32, "Asghar Mohammad" wrote: > work around was block dtmf. > set wrong type of dtmf in incoming trunk. > > > On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N < > gopalakrishnan...@gmail.com> wrote: > &

Re: [asterisk-users] DTMF recognized after call establishment

So any resolution for this? I suspect it could be related to RE INVITE On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad wrote: > i had this in past there was an ATA configured to send 9 at the end of > dialing in my case. > > > On Tue, May 28, 2013 at 8:21 AM, G

[asterisk-users] DTMF recognized after call establishment

Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_di

Re: [asterisk-users] Asterisk 11 dtmf not recognised

lse getmore will called exten = 300,n(done),Playback(letters/c) ; Didn't go to pin-accepted, so play badPIN and hangup exten = pin-accepted,1,Playback(letters/b) ; correct pin, play Thanks On 25 May 2013 15:38, "Gopalakrishnan N" wrote: > Am using Read application to get the

Re: [asterisk-users] Asterisk 11 dtmf not recognised

correct pin, play what happens its keep on asking to enter digit If my DTMF didnt match. Do i need to use any return function... ? Actually my goal is to ask for 3 times and if not matched then return to some other application. Thanks in advance. On Sat, May 25, 2013 at 3:19 PM, Gopalakr

Re: [asterisk-users] Asterisk 11 dtmf not recognised

With Asterisk 1.8 I got it working. Regards On Sat, May 25, 2013 at 2:37 AM, Gopalakrishnan N < gopalakrishnan...@gmail.com> wrote: > Tried info, rfc2833, inband and finally kept as auto. > On 25 May 2013 02:20, "Doug Lytle" wrote: > >> >> dtmfm

Re: [asterisk-users] Asterisk 11 dtmf not recognised

Tried info, rfc2833, inband and finally kept as auto. On 25 May 2013 02:20, "Doug Lytle" wrote: > >> dtmfmode=auto > > dtmfmode=info > > or > > dtmfmode=rfc2833 > > Doug > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve

[asterisk-users] Asterisk 11 dtmf not recognised

Hi I have a dialplan as per the following, extensions.conf [avgtest] exten = 100,n,Playback(avgtest/message1) exten = 100,n,Set(rightPIN=1) exten = 100,n,Read(inPIN,,1,,5,3) ; Attempts for 5 times with 3 seconds of timeout exten = 100,n,GotoIf($["${inPIN}" = "${rightPIN}"]?pin-accepted,1) exten =

Re: [asterisk-users] Error 488 Not Acceptable Here

488 not acceptable is due to codec error. Make sure you have right codec in place between the end points. On Fri, May 24, 2013 at 12:18 AM, Maximilian Grobecker < m.grobec...@portunity.de> wrote: > Hi, > > Maybe you have not allowed T.38 as acceptable codec ;-) > You can try with "allow=all" in

Re: [asterisk-users] GotoIf function

I just want to make some increment... to 3 and yes go to the desired option not to one more option. On Thu, May 23, 2013 at 7:19 PM, Gopalakrishnan N < gopalakrishnan...@gmail.com> wrote: > Hi, > > Actually i would like to get the input from the user and he should not try >

[asterisk-users] GotoIf function

Hi, Actually i would like to get the input from the user and he should not try more than 3 times, he can try more than 3 times, if yes it will get routed to the next priority and if not it goes to the loopback again from the beginning. And following is the one I created, I just want to know wheth

Re: [asterisk-users] debug strategy for one-way audio calls

@Marrie For one way audio as a debug strategy you can enable RTP debug and see whether you have both way packets flow SENT and GOT. Regards On Thu, May 2, 2013 at 6:05 PM, Johan Wilfer wrote: > 2013-05-02 13:19, Marie Fischer skrev: > > Hello everybody, > > > > from time to time, we get so-cal

[asterisk-users] VoIP Incoming Issue

I have made the SIP bind port to 5070, and already I have one VoIP trunk configured in my Asterisk 1.6. Now the problem is after changing the bind port at some point of time, am not able to dial in the DID number of the VoIP trunk! Changing the bind port matters for this? Regards. -- ___

[asterisk-users] Asterisk with R2D configuration

Hi, Has anybody worked on R2D Brazillian setup. I have configured R2 using OpenR2 with Asterisk. While doing some analysis I found R2D is already included in libopenr2. Have anyone tested the same. Regards, Gopal. -- _ -- Bandw

Re: [asterisk-users] SIP Retransmitting REGISTER message

ithus register attempts !! > > See this page from callwithus and configure your asterisk accordingly for > both accounts. > > http://www.callwithus.com/configuration > > BR > Sammy > > > > On Thu, Sep 27, 2012 at 12:09 PM, Gopalakrishnan N < > gopalakrishnan...@gma

Re: [asterisk-users] SIP Retransmitting REGISTER message

m the softphone first and the >> provider took the IP address from your PC and “locked out” the IP address >> of your Asterisk server. >> >> ** ** >> >> *From:* asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digi

Re: [asterisk-users] SIP Retransmitting REGISTER message

, Gopalakrishnan N < gopalakrishnan...@gmail.com> wrote: > ahh... ! OK.. I though of this... > > > > On Wed, Sep 26, 2012 at 6:24 PM, Danny Nicholas wrote: > >> Another possibility – you registered from the softphone first and the >> provider took the IP address

Re: [asterisk-users] SIP Retransmitting REGISTER message

** > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N > *Sent:* Wednesday, September 26, 2012 7:45 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] SIP R

[asterisk-users] SIP Retransmitting REGISTER message

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but sinc

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

ote: > > On 08/30/2012 09:45 AM, Gopalakrishnan N wrote: > > >Hi, > > > > > >I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, > > >I am not using any virtualbox, still i struck in loading the modules. > > > > Please do not top

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

.) > 616-855-1030 Ext. 2003 > > > ---------- > *From*: "Gopalakrishnan N" > *Sent*: Tuesday, August 28, 2012 1:13 PM > *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > > *Subject*: Re: [asterisk-users] Aste

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

ething like libpri is > biting you. > > ** ** > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N > *Sent:* Tuesday, August 28, 2012 11:52 AM > *To:* Asterisk Users Mailing List - Non-Com

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

es.conf not load anything to start with so you can eliminate a rogue > module as the problem. Just change autoload=yes to autoload=no. > > ** ** > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

isk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N > *Sent:* Monday, August 27, 2012 8:52 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse > 12.2 > > ** ** > >

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

-12 08:25, Gopalakrishnan N wrote: > >> This is really tuff working with OpenSuse. I am clueless how to sort our >> this. >> > > Maybe switch to a different distribution? I have used CentOS and RHEL for > years without any problems and as far as I know both debian and ub

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

ad_modules: 186 modules will be loaded.* This is really tuff working with OpenSuse. I am clueless how to sort our this. Regards. On Fri, Aug 24, 2012 at 3:55 AM, Hans Witvliet wrote: > On Thu, 2012-08-23 at 15:01 +0530, Gopalakrishnan N wrote: > > Hi, > > > > > > Aga

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

ppreciated. Regards, Gopal. On Tue, Aug 21, 2012 at 11:24 AM, Gopalakrishnan N < gopalakrishnan...@gmail.com> wrote: > Ok Thanks Bryant, let me try with OpenSuse 12.1. > > Regards. > > > On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman wrote: > >> I have the curre

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

ll of our boxes are complied from source. > > Thanks > > Bryant Zimmerman (ZK Tech Inc.) > 616-855-1030 Ext. 2003 > > > ------ > *From*: "Gopalakrishnan N" > *Sent*: Monday, August 20, 2012 10:11 AM > *To*: "Bryant Zimmerman

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

>From the forum I understand OpenSuse 12.2 is pre-relase and better to use OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N < gopalakrishnan...@gmail.com> wrote: > Its really weird working with OpenSuse. I am not sure how othe

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

Its really weird working with OpenSuse. I am not sure how others are using with OpenSuse. Through Yast also I tried to install Asterisk package, it didn't find. Now I am clueless to work with OpenSuse. Regards. On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N < gopalakrishnan...@g

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

Hi Patrick, Thanks for your suggestion, even though I added my hostname in the /etc/hosts, still the problem persists. Also I tried to install in OpenSuse 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like hanging at modules while starting Asterisk. Regards, Gopal. > Pleas

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

Edwards wrote: > On Tue, 14 Aug 2012, Gopalakrishnan N wrote: > > while executing asterisk -c from the root prompt, its stuck as below >> and the CPU usage is fully utilized, >> > > [snip] > > >== Parsing '/etc/asterisk/modules.conf': == Fou

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