similar to: Asterisk as a "gateway"

Hello list, I'm having a problem with a BRI card, a B200E (PCI-e) card on an Intel server with a riser that I'm guessing is what's making the problem. The output of lspci -vvv is: 07:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Subsystem: Cologne Chip Designs GmbH Device e884 Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop-

Hello everyone, I'm having a hard time configuring my router to forward asterisk traffic correctly. I have the following ports being forwarded to asterisk: 5060, 10000-20000 Now, I can register the accounts when outside the network and I can call every extension that is inside the network. The problem is that I can't ear anything nor can the phones inside the network phone the

Hello list, I've been having a problem for some time now that I can't figure out how to solve it. On a PTP BRI ISDN line, if I have both channels in use and I place a third call from the outside, I'm not getting a busy tone like I should. Instead I get a congestion tone, as if the line was not working/didn't exist. Currently I'm using mISDN 1_1_9.1 and Asterisk 1.4.35 with

Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN and I think the reason is because asterisk is sending the release cause as 0. P[ 3] --> channel:0 mode:TE cause:0

Greetings everyone, I'm trying to compile asterisk for an AVR32 (Atmel NGW100). Buildroot for AVR32 already has the asterisk package, though it has bugs. Firstly it tries to apply a patch for 1.2 on a 1.6, but deleting the contents of the patch file did the trick. Now, the problem is making asterisk. The first error is because asterisk needed to be ./configure:ed. Trying to just do

Hello list, I'm having some issues with some phones that don't stop ringing after the call is answered somewhere else. Basically, a call comes, enters a queue and all the phones in the queue ring. The problem is that when the call is answered, some phones don't stop ringing. I don't know if it is a configuration file, but I don't think so. queues.conf, sip.conf and

Lista aprovados Jequi? da Praia: Tesouro: ANA CLEIDE PEREIRA DE VASCONCELOS, LUAN MARINHO DE SOUZA, FRANCISCO LEANDRO FIUSA DA SILVA, PEDRO HENRIQUE ARAUJO SANTIAGO, JO?O CARLOS MOREIRA DE CARVALHO, DANIEL DA CUNHA MAGALHAES, MARIA IVANI DE ARA?JO, JEANNIE PAULO DE SOUSA. SILVANA GALDINO FERREIRA, AUGUSTO BRUNO PEREIRA DOS SANTOS, LUCIANA ESTELITA LAFAYETTE RABELO, GRACY KELLY DE LIMA MORAIS,

Greetings everyone, I'm having some issues connecting a Asterisk box to a old ISDN PBX. Everything works fine but the undetermined digits rules. For instance, if I have _00X. and I start dialing for instance 0035..., Asterisk just get the 4 first numbers and starts dialing 0035. I've tried adding: stop_tone_after_first_digit = yes append_digits2exten = yes Although I

Brent- There is no good way to say what changing the hardware and PSTN hookup will probably do for the echo problems. I'm not sure if you mentioned (lost in the past history of your post now) what sort of hardware you're using for PSTN connection now- TDMs, X100s, ATA's, etc- but that could also be a potential cause. I've heard tell of aftermarket X100s and certain ATA's

Hi All, We've got msidn configured: Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> childcnt: 2 -------- mISDN_close: fid(3) isize(131072) inbuf(0x8fd5060) irp(0x8fd5060) iend(0x8fd5060) and running on Asterisk 1.4.21.2: pbx*CLI> misdn show stacks BEGIN STACK_LIST: * Port 1 Type TE Prot. PMP L2Link UP L1Link:UP Blocked:0 Debug:0

Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message "Didn't get a frame from channel: SIP/3805-df43", but I can't figure why. asterisk logs:

Rela??o de aprovados Mar Vermelho: ?gua Clara: ANA PAULA RODRIGUES DA SILVA, LUCAS ARAUJO GOMES FROTA, GABRIEL VICTOR BARROS FORTE DA SILVA, QUIT?RIA DA SILVA G?IS, JO?O CARLOS MOREIRA DE CARVALHO, DAYANA MARIA DE SOUSA TAVARES, MARIA JULIENE CORDEIRO, JO?O PAULO DA SILVA. TALITA FERNANDES GONCALVES, BRUNO RAMOS FERNANDES, LUIZ HENRIQUE ALVES DAMASCENO, IAGO DA SILVA NOBRE, RITA ANGELA DA SILVA.

Rela??o de aprovados Japaratinga: Tapurah: ANA CAROLINE LIMA VASCONCELOS, L?VIO CHAVES D?D?, FRANCISCO JACKSON DE FREITAS SOUZA, PAULO WILLAME ARAUJO DE LIMA, JO?O CARLOS MOREIRA DE CARVALHO, DAMI?O ROCHA DOS SANTOS J?NIOR, MARIA HELENA DE OLIVEIRA SILVA, JARDEL DE LIMA NOBRE. SILVANA BEZERRA DA SILVA, ARIELA DE MACEDO RICARDO, LUCAS PINTO ASSUNCAO, GLAUCIO VALENCA PEREIRA RANGEL, REBECA FARIAS

Lista aprovados Passo de Camaragibe: Chapad?o do Sul: ALINE RAQUEL BEZERRA LEITE, LEANDRO DO NASCIMENTO BATISTA, FERNANDO MONTEIRO DOS SANTOS, NEYLSON CAVALCANTE DOS SANTOS, JO?O CARLOS MOREIRA DE CARVALHO, CLEVILSON PEDRO FONSECA OLIVEIRA, MARIA DE FATIMA ALEIXO PINHEIRO, IVETE CORDEIRO DE SOUZA. SAMILLE DE OLIVEIRA CAVALCANTE, ELIS?NGELA DOS SANTOS SILVA, MAURY DE SIQUEIRA MELO PEREIRA, JOSEFA

Greetings everyone, I've been having some strange issues with my Asterisk box and some snom phones. In some cases, when I talk, the sound in the other end is cut off, I stop earing the background noise - looks like a walkie-talkie. I've tried this between phones in the same network and in all but one this happens. The one where it doesn't happen is the one connected directly to the

http://bugzilla.netfilter.org/show_bug.cgi?id=639 Summary: iptables iprange Product: iptables Version: unspecified Platform: All OS/Version: All Status: NEW Severity: blocker Priority: P1 Component: iptables AssignedTo: netfilter-buglog at lists.netfilter.org ReportedBy: paulo.santos at

Hi All, Has anyone had experiences with Asterisk and ISDN Gateways or cards? I am looking to setup a home phone system using asterisk and am looking to convert my two existing no's to a single isdn2e line. I was wandering if anyone could recommend which is better to use with asterisk - ISDN Card or gateway? also which brands are best with linux/asterisk? I have used eicon BRI cards for

Nomes dos aprovados Maragogi: Alcin?polis: ANAHISA PEDROSA VITALINO, LUCAS BEZERRA DE MESQUITA, GABRIELLE DE MOURA FERREIRA, RACHEL CABRAL MOTA, JO?O CARLOS MOREIRA DE CARVALHO, DEBORA BEZERRA DE BRITO, MARIA KRYSLANE CARDOSO LEMOS, JOAO PEDRO ALEXANDRE SILVA MOTA. TALYTA MARTINS NEVES, CAIO BRITO BARREIRA, MADSON SANTOS MONTEIRO, IAGO FILLIPI PATROCINIO MACEDO, RITA DE CASSIA GADELHA DA SILVA.

Hi Folks, I've Asterisk Bristuffed up and running behind an Auerswald Commander Basic ISDN PBX on the internal ISDN Bus (BRI/PTMP). The HFC Card works marvelleous for outgoing calls (as the parallely installed avm fritzcard with chan_capi does), but when I'm trying to call in, I get a short ring signal and then the connection is terminated. This does not happen with chan_capi and

Lista aprovados Maravilha: Amamba?: ANDERSON BEZERRA MOURAO, LUCAS CAULA ALBUQUERQUE, GEILSON HOLANDA SAMPAIO, RAFAEL FERNANDES, JO?O CARLOS MOREIRA DE CARVALHO, DEBORA CRISTINA SCHNORNBERGER, MARIA MACLENE BEZERRA LIMA, JOAO PEDRO TAVARES MAGALHAES. TAMYRES AMORIM SOUZA, CAIO CESAR FERNANDES LOPES, MAGNUN SANTOS FREDERICO, IAGO SIMOES CALIARI, RITA MARIA SILVA ALMEIDA. Pedreiras. Maravilha,