similar to: Asterisk Integration with Android device

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 since in softphone it is getting registered from the same network. Any ideas to isolate things would be

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 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and downloaded Asterisk 1.8 current version, after installing Asterisk, while starting Asterisk it hangs at the stage of loading modules.conf, I checked the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still after updating through yast also i am facing the issue. Have anybody faced this type of issue with

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi users, I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file to G729 using file convert, but I am facing error as follows, file convert /tmp/welcome.gsm /tmp/welcome.g729 Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729! Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed. [Dec 20 17:24:18] WARNING[2221]: translate.c:256

Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any

Hi, I am using Asterisk 1.2.24, I have written my dialplan to land with an IVR with the same time if the customer knows the parties extensions they can dial directly, but what happens is sometimes its working and sometime its not working. My extensions.conf as follows, [incoming] exten => 052477302,1,Wait(2) exten => 052477302,2,NoOp(${CALLERIDNUM}) exten =>

On 9 November 2011 10:40, Micol lupen <galaxi_999 at yahoo.it> wrote: > Hi to all, > i am a university student, i have this problem, i want to create application for android this application must to record and listen ogg file, > i read that android use only ogg decoder for this problem i must use external library,wich library i can use ? can you show me some ?HowTo or book or site

Hi, I am trying to connect the outbound dialing with mysql with the following code, exten => 88,1,MYSQL(Connect connid hostname username password dbname) exten => 88,2,GotoIf($["${connid}" = ""]?error,1) exten => 88,3,MYSQL(Query resultid ${connid} SELECT\ phone\ FROM\ <tablename>\ WHERE\ phone =${a}) exten => 88,4,MYSQL(Fetch fetchid ${resultid} ph\ sa)

Hi all, This is Jagan.i have been provided a task of analyzing sales data of a company in R programming...Just wanted to know,how can I pull Tally 9.1 software data into R programming dataframe. Waiting eagerly for your inputs. With Regards,Jagannathan Krishnan Sent from Yahoo Mail on Android On Tue, Aug 22, 2017 at 10:47 AM, jagan krishnan<jagan.krishnan at yahoo.co.in> wrote: Hi

hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is

On Thursday, August 24, 2017, 1:50:13 AM EDT, David Winsemius <dwinsemius at comcast.net> wrote: > On Aug 22, 2017, at 11:31 PM, jagan krishnan via R-help <r-help at r-project.org> wrote: > > Hi all, > This is Jagan.i have been provided a task of analyzing sales data of a company in R programming...Just wanted to know,how can I pull Tally 9.1 software data into R

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. -------------- next part -------------- An HTML attachment was scrubbed... URL:

> On Aug 22, 2017, at 11:31 PM, jagan krishnan via R-help <r-help at r-project.org> wrote: > > Hi all, > This is Jagan.i have been provided a task of analyzing sales data of a company in R programming...Just wanted to know,how can I pull Tally 9.1 software data into R programming dataframe. > Waiting eagerly for your inputs. > With Regards,Jagannathan Krishnan > Sent

Hi All I don't know whether i am asking on a correct list or not i have a query that how to know that what system call are called when packet enters the system or when we send a packet from a system? Thanks

Hi, Inline below. > On Aug 24, 2017, at 5:22 AM, John Kane via R-help <r-help at r-project.org> wrote: > > > IIt might help to read the material at one or both of these links http://stackoverflow.com/questions/5963269/how-to-make-a-great-r-reproducible-example > <snip> In this case, had Jagan spent about 30 seconds Googling for the application developer's

Hi , Is it possible to have two meetme room in asterisk 1.6 which each one have a different language? I mean, one room the annoucement is in Portuguese an another in english? Today I can change over the sip.conf and it is valid for all room. regards! Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part -------------- An

Hi, I was wondering if anyone here had thought about FLAC support on the Android. It would be great for FLAC adoption. There's a bug open for it<http://code.google.com/p/android/issues/detail?id=1461>on which I've posted some details about how I think it would need to be implemented. It appears to me that it would need to be added to the OpenCore component, which is written in C.

Hello, I'm trying to conect two 1.6.2.13 Asterisk server with IAX. This is my configuration: Asterisk A: iax.conf register => coiax:pass1 at 69.164.207.166 [smiax] type=friend host=dynamic trunk=yes secret=pass2 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.207.166/255.255.255.255 qualify=yes Console: iax2 registry 69.164.207.166:4569 N coiax 69.164.197.105:4569

IIt might help to read the material at one or both of these links http://stackoverflow.com/questions/5963269/how-to-make-a-great-r-reproducible-example Reproducibility ? Advanced R. | | | | Reproducibility ? Advanced R. | | | On Thursday, August 24, 2017, 6:19:25 AM EDT, John Kane <jrkrideau at yahoo.ca> wrote: On Thursday, August 24, 2017, 1:50:13 AM EDT, David