search for: makevuy

Hy, I have a network with WIFI communication and VHF/ HF channels. I have integrated asterisk in the network using SIP, ZAP and IAX2 channels for WIFI communications, but I don't Know How I could integrate the VHF/ HF channels. I have heard speaking about app_rpt project, but I don't Know very much about this. Could I integrate VHF/ HF channels with this application? if the answer is

...-j TOS --set-tos 0x20 But the TOS doesn''t change (I see this with ethereal) Waht could be the problem? I''ve tried the same with the MARK option: iptables -A PREROUTING -i eth1 -t mangle -p udp -j MARK --set-mark 0x20 and a DSMARK. Regards. -- Sandra Salmerón Ntutumu <makevuy@ehas.org> Tlf. Analog: +34 914888405 / Móvil: 653574298 Tlf. IP desde FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010 Fundación EHAS: Enlace Hispanoamericano de Salud - www.ehas.org Telemedicina rural para zonas aisladas de países en desarrollo

Hello, Has anyone Knowledge about softphone IAX for pocket PC totally free? Tkanks for all. -- Sandra Salmer?n Ntutumu <makevuy@ehas.org> Tlf. Analog: +34 914888405 / M?vil: 653574298 Tlf. IP desde FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010 Fundaci?n EHAS: Enlace Hispanoamericano de Salud - www.ehas.org Telemedicina rural para zonas aisladas de pa?ses en desarrollo

Hy, I would want to give QoS a one wireless network. Do you know if exist any qdisc specific for this??. I know that the most modern qdisc is HFSC, is this true??. Thanks for all and Regards. -- Sandra Salmerón Ntutumu <makevuy@ehas.org> Tlf. Analog: +34 914888405. Ext:10 / Móvil: 653574298 Tlf. IP desde FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010 Fundación EHAS: Enlace Hispanoamericano de Salud - www.ehas.org Telemedicina rural para zonas aisladas de países en desarrollo

Hello everybody, I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip installation. I actually have a problem when a PSTN user calls and hangs up. The disconnect tone is not detected by the SPA, the the call continues and, for example, leaves an empty message on the voicemail before hanging up (because * hangs up). How could resolve this problem?. I set, Detect

...because the comand do the next operation: OLD_TOS + 0x00 = OLD_TOS I would want to know if is possible with iptables, change the TOS of the packets, althought the TOS of the packets that are arriving to my network is difference from 0x00. Thanks ad Regards. -- Sandra Salmerón Ntutumu <makevuy@ehas.org> Tlf. Analog: +34 914888405 / Móvil: 653574298 Tlf. IP desde FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010 Fundación EHAS: Enlace Hispanoamericano de Salud - www.ehas.org Telemedicina rural para zonas aisladas de países en desarrollo

...ing the parameter of qdisc that depends of bandwith, with this variances. In sumarize, an adaptative design with the bandwith of each Wifi link. Is that possible with linux?, what can we do this? or what could be the best strategy? Thanks for all and Regards. -- Sandra Salmerón Ntutumu <makevuy@ehas.org> Tel. +34 914888405 / Móvil: 653574298 Fundación EHAS: Enlace Hispanoamericano de Salud - www.ehas.org Telemedicina rural para zonas aisladas de países en desarrollo

Hy, I would want to "test" the FXO and FXS channels of a TDM400p with the asterisk. For example, when I want to Know if a sip peers is correctly register, I use ChanisAvail, but if a module FXS hasn't a phone connected, How could I Know it?. Note, that I don't want to Know if the channel ZAP is correctly configurated. This information is given for Chanisavail. Thanks for

Hy, I'm writing from Spain. I have the 1.2.4 asterisk version and 1.2.3 zaptel version. I've heart that this asterisk's version detects correctly de hang up of PSTN, but in my case this thing doesn't happen. Moreover, my asterisk sends the next messages in the CLI: Feb 21 15:03:13 WARNING[10363]: chan_zap.c:10876 setup_zap: Ignoring signalling Feb 21 15:03:13 WARNING[10363]:

hello I have an asterisk server with a public IP address and a nat address like alias. I have 20 sip clients with private IP address. I don't Know why, sometimes, when I try to call between 2 phones, y see the next menssage in the astersk console: Can't find address for host 'XX' What could happen?? Thaks for all and regards

Hy, I'm working with asterisk 1.2.4 and zaptel 1.2.4 With these version an the options in zapata.conf: answeronpolarityswitch=yes hanguponpolarityswitch=yes I don't detect polarityswitch. When asterisk reloads I see in CLI: May 10 00:44:27 WARNING[14639]: chan_zap.c:10876 setup_zap: Ignoring answeronpolarityswitch May 10 00:44:27 WARNING[14639]: chan_zap.c:10876 setup_zap: Ignoring