similar to: Dial timeout with SIP - how to set timeout for INVITE ACK

Hello to all, I have been using an optim with the following call: optim(param_ini,fun_errores2,Precio_mercado=Precio,anos_pagosE2=anos_pagos,control=list(maxit=10000,reltol=1e-16)) depending on the intial values I'm getting the same solution but once I get the convergence message=10 (no convergence) and for the others I get convergence message = 0 Solution1: $par beta1

Hi. Can anyone suggest a simple way to re-sort in R a list of vectors of the following form? input $"1" a b c 1 2 3 $"2" a b c 4 5 6 Output should be something like: "a" "1" 1 "2" 4 "b" "1" 2 "2" 5 "c" "1" 3 "2" 6 I've been futzing with mapply(), outer(), split(), rbind()

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with

Folks, in many recent threads this pattern has been happening: OP: Any suggestions on how to do <something>? Responder 1: Look at <solution1>. Responder 2: Look at <solution2>. Responder 3: Look at <solution3>. Responder 4: It doesn't work for me / I don't like that / it's a stupid solution / etc

I was looking for a (http socket module / mysql module) not using AGI(perl/php/shell) for asterisk in order to do intensive database / web server interactions as needed without performance to much overhead. Is there a real benefit in using : "Asterisk cmd MYSQL" app_addon_sql_mysql , I see it require perl DBD so it is not using Mysql C API, then there must be an instance of the perl

Hi, I would like to monitor FEC usage in order to include it in RTCP EX or calculate MOS estimation, etc. However the Opus codec library does not seem to expose such information. "Was LBRR found and used or was it PLC ?" I saw in WebRTC that they are using a technique to parse the "frame header" WebRtcOpus_PacketHasFec() It this something that is supported ? What would you

I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request? # sip.conf [general] insecure=very permit=207.148.115.10/255.255.255.0 [myproxy] type=friend host=217.118.115.10 context=from-sip # Logging: <--- Reliably Transmitting (NAT) to 207.148.115.10:5060 ---> SIP/2.0 407 Proxy

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On Fri, Jul 20, 2001 at 11:27:16AM +0500, Asif M. Baloch wrote: > Hi guys, > > ICE cast resembles shoutcast, thats for sure and its good that its open > source. But, it has too many probs. I ran it on a dual p3 800 with a T3 comm We are running icecast on linux 2.2.19 serving more than 600gb each months and it has been rock stable so fare. Right now our uptime is around 3 months.

Backgroup: Customer has many old documents(just some scanning pictures) and want to store them into computer,we need to input the doc and relate them to the existing authors in DB. Key works: document;author UI of document input page: left side is the document input form right side is the author search form Problem: How should i arrange the code to deal with it. I have 2 solutions, but i do not

Hi, I intend to change the cdr_mysql-field "uniqueid", which seems not to be used so far, to an (not unique) indexed field and use it later for my own hints and infos. I don't have very much traffic so far, and I wonder, if there will appear problems when asterisk is under high load (100 simultanious calls) and the log table contains 1.000.000 log lines. This would mean, that

On 10/17/19 4:08 PM, Johnny Hughes wrote: > On 9/24/19 2:41 PM, Frank Cox wrote: >> Without wanting to sound too pushy, I'm wondering if there is any update on the status of Mate now that Centos 8 has been released? >> >> I would love to jump on C8 and start playing with it, but the lack of Mate is kind of a showstopper for me at the moment. >> > > Is gnome3

Depends what you want to do! Do you want to do VoIP over that T1 to a provider or IP telephones? Do you want to hook up to the PSTN through that T1 as 24 voice channels, through a T1 card on your asterisk? If you want to use the T1 as 24 voice channels, the Telco is going to have to re-provision the T1 as a voice T1, because currently, presumably it is one big channel of data. You could have

I need for asterisk to call me at a predetermined number once a day at a predetermined time and once connected to me make 5-10 simultanious calls to a DID filling all available channels. What is the best way to do this? Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091015/b97f2d1e/attachment.htm

Hi there, we are looking for an opensource or commercial * based Call Center. Full ACD, call monitoring, multiple queue, IVR, voicemail, management, reporting, CDR, etc is needed. over 100 seat can be the initial target and will grow in a very short time. SIP phones will be used and multiple E1 lines incoming, so to provide full failover a cluster of * machines or some other form of redundancy

YANS: Yet Another Silly Idea The discussion about tags seems to me to be getting a little silly. >From my 'newbie' perspective on tag formats, it seems to me that all tags are arbitrary. I mean, sure, you could add a 'your player should support this tag in this format', but really, it comes down to consistency on the Encoder's (person that encoded the file) part in how to

Hello Rubyists, I''m in the position of being given the job to design and build a mission-critical web facing application for a small but growing enterprise. It is to be used by customers as needed, numbering in the tens up to the hundreds. It will collect operational data, particularly in a table around 50 columns wide, and potentially millions of rows deep, most fields being numerical.

Dear R-users I am trying to understand how Sweave works by running some simple examples. In the example I am working with there is a chunk where the R-commands related to plotting a figure are placed. When running R CMD Sweave ? , pdflatex the output is a portrait figure. I wonder whether it would be possible to change the orientation to landscape (not in the latex file but in Rnw file). Many

On 2016-08-09 10:06, Faheem Muhammad wrote: > trip time and Call Setup time of SIP Requests. > In case of GSM Network with high delay you need to set the T1 timer a > higher value like 1000ms (500 ms default). Similarly you can reduce the > Call setup time by configuring 'T2' upto you choice as per you telephony > network. Configure t1min, timert1 and timerb according to