search for: arruda

Does anyone know if it's possible to set the codecs for a number via an Asterisk command? Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a command that can set the same thing so that it can be done without having to change sip.conf. Essentially I want the user to be able to prefix a code to their dialled number to set their preferred codec for that call.

...K-29040 - res_speech: Assertion on format (Reported by Nickolay V. Shmyrev) * ASTERISK-29001 - chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) * ASTERISK-29034 - Lastpause of realtime members is reseting (Reported by Evandro C��sar Arruda) * ASTERISK-27273 - app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command (Reported by Leandro Dardini) * ASTERISK-29033 - res_pjsip_session: Aggressively terminates session on failed re-INVITE (Repo...

...K-29040 - res_speech: Assertion on format (Reported by Nickolay V. Shmyrev) * ASTERISK-29001 - chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) * ASTERISK-29034 - Lastpause of realtime members is reseting (Reported by Evandro C��sar Arruda) * ASTERISK-27273 - app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command (Reported by Leandro Dardini) * ASTERISK-29033 - res_pjsip_session: Aggressively terminates session on failed re-INVITE (Repo...

...EFERSON GUERRA DA COSTA, PEDRO AMARANTE SILVA FILHO, JO?O CARLOS MOREIRA DE CARVALHO, DANIEL AGUIAR DE SOUSA, MARIA ISABELA DE BRITO MORAIS, JARDEL SOUSA DA SILVA. SILVANA FERREIRA DE ARA?JO, ARTHUR MAURICIO DE SOUZA RODRIGUES, LUCAS RAMOS DE LIMA, GLEYCIANNE M. DE CARVALHO, REBECCA ERONIDES NOJOSA ARRUDA. Moss?medes.

Anyone tried the new PSTN/FXO port in the new SPA 3102 FXO/FXS adapter ? From a quick test (got mine yesterday), seems like it is not recognizing Caller ID from PSTN/FXO port.. Using the same configuration as a Sipura 3000 (to be sent to mother-in-law POP :-), no Caller ID at all, (I've even extended the PSTN delay to give it more time, but no dice). www.voxilla.com forum has a couple

Hi, I need to classify, using Naive Bayes and Bayes Networks, and estimate their performance using cross validation. How can I do this? I tried the bnlearn package for Bayes Networks, althought I need to get more indexes, not only the error rate (precision, sensitivity, ...). I also tried the *e1071* package, but I could not find a way to do cross-validation. Thanks for everyone. Guilherme.

...EFERSON GUERRA DA COSTA, PEDRO AMARANTE SILVA FILHO, JO?O CARLOS MOREIRA DE CARVALHO, DANIEL AGUIAR DE SOUSA, MARIA ISABELA DE BRITO MORAIS, JARDEL SOUSA DA SILVA. SILVANA FERREIRA DE ARA?JO, ARTHUR MAURICIO DE SOUZA RODRIGUES, LUCAS RAMOS DE LIMA, GLEYCIANNE M. DE CARVALHO, REBECCA ERONIDES NOJOSA ARRUDA. Caratinga. Presidente Tancredo Neves e ANA PAULA PEREIRA DA SILVA, LUCAS ARAUJO DA SILVA, GABRIEL LIMA VALENTIM, QUEZIA GOMES SNTOS, JO?O CARLOS MOREIRA DE CARVALHO, DAYA VALE DOS SANTOS, MARIA JOSEANE DE SOUZA, JOAO HUDSON OLIVEIRA NUNES. SU?NIA CONCEI??O SILVA, BRUNO MORAES NASCIMENTO, LUIZ GU...

For work environments where you only get HTTP or HTTPS access, what is the feasibility of doing IAX over HTTP? I have heard of projects such as stunnel. Has anyone tried something like this? I did a quick search but didn't come up with much.

...ike noted by Soren (http://www.ad2.com.br/DTMF.jpg), the main diference between the 2 samples is the time elapsed after the "burst" sign and the first DTMF digit. In my sample, its occurs more quickly... I inspect the asterisk code for days, but nothing found... Thanks all Alexandre de Arruda Paes

How much of an impact can/does local network traffic have on call quality? Would opening large files on local servers affect call quality? We are running QoS on the router but that will only prioritize traffic in/out of the network.

Hello, I am wondering if anyone is using the Nortel 2001 2002 or 2004 phones on their asterisk implementation. My local dealer says they are not compatible with any open source implementations. Is there a phone compatibility list somewhere? Cheers Cian

Dear all, I've noticed some significant voice quality deterioration when calling US landline via VoIPjet.com in the last week or so. Before that the quality was pretty good. Has anyone else experienced any voice quality problems with voipjet recently? Thanks, Roman

...ime for compiler clang conditionally. (Reported by Alexander Traud) * ASTERISK-27774 - res_musiconhold: Music on hold restarts after every announcement (Reported by lvl) * ASTERISK-27782 - cdr_mysql: Missing MYSQL_PORT definition (Reported by Evandro C??sar Arruda) * ASTERISK-27614 - res_pjsip_session: SDP origin does not use resolved address (Reported by John M.) * ASTERISK-27740 - chan_sip: New Channel creation from new SIP dialog with Replaces failed to be properly tracked and destroyed (Reported by Shannon Price) * ASTERI...

...K-27806 - BASIC-RETRANS: Implement send (Reported by Benjamin Keith Ford) * ASTERISK-27774 - res_musiconhold: Music on hold restarts after every announcement (Reported by lvl) * ASTERISK-27782 - cdr_mysql: Missing MYSQL_PORT definition (Reported by Evandro C??sar Arruda) * ASTERISK-27614 - res_pjsip_session: SDP origin does not use resolved address (Reported by John M.) * ASTERISK-27776 - res_rtp_asterisk: Add support for sending RTCP feedback messages (Reported by Joshua Colp) * ASTERISK-27740 - chan_sip: New Channel creation from new S...

We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What we're trying to avoid is hardcoding the IP address in the ACL. We were trying to match by TOS set by Asterisk however it seems we've run into a snag where the packet TOS tends to get reset somewhere on our network. Has anyone had this issue? We're running Cisco everywhere inbetween (even the switches). Is

Can anyone point me in a good direction for configuring SIP through a PIX using 1:1 NAT. I have read anything I could get my hands on and tried them all with very little success. I can get it to work through the cheap little cable modem routers, but not this PIX. I -can- make a direct SIP call using the IP address of the * server (ie.exten@ipaddr), but when I do that * still doesn't

We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. This week we rebuilt the entire LAN with Cisco 2950-EI switches and have employed QoS on the switches and router. Still sounds terrible. What we are now finding is that the network card in the PC may be the key to the problem. A Dell Optiplex P4 2.4GHz 512MB

...ime for compiler clang conditionally. (Reported by Alexander Traud) * ASTERISK-27774 - res_musiconhold: Music on hold restarts after every announcement (Reported by lvl) * ASTERISK-27782 - cdr_mysql: Missing MYSQL_PORT definition (Reported by Evandro C��sar Arruda) * ASTERISK-27614 - res_pjsip_session: SDP origin does not use resolved address (Reported by John M.) * ASTERISK-27740 - chan_sip: New Channel creation from new SIP dialog with Replaces failed to be properly tracked and destroyed (Reported by Shannon Price) * ASTERI...

...K-29040 - res_speech: Assertion on format (Reported by Nickolay V. Shmyrev) * ASTERISK-29001 - chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) * ASTERISK-29034 - Lastpause of realtime members is reseting (Reported by Evandro C��sar Arruda) * ASTERISK-27273 - app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command (Reported by Leandro Dardini) * ASTERISK-29033 - res_pjsip_session: Aggressively terminates session on failed re-INVITE (Repo...

...K-27806 - BASIC-RETRANS: Implement send (Reported by Benjamin Keith Ford) * ASTERISK-27774 - res_musiconhold: Music on hold restarts after every announcement (Reported by lvl) * ASTERISK-27782 - cdr_mysql: Missing MYSQL_PORT definition (Reported by Evandro C��sar Arruda) * ASTERISK-27614 - res_pjsip_session: SDP origin does not use resolved address (Reported by John M.) * ASTERISK-27776 - res_rtp_asterisk: Add support for sending RTCP feedback messages (Reported by Joshua C. Colp) * ASTERISK-27740 - chan_sip: New Channel creation from ne...