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Hello I Installed Asterisk on RedHat 9. I am currently try to configure minimum with two softphone talking to each other over the LAN. I am using X-Lite softphones from xten.com site. I defined 3 phones in sip.conf and also specifies in extensions.conf file. I am able to ring other computer but there is no voice exchange ( i can't hear any think except ring). Here is the portion of sip.conf

Hello I am trying to setup Asterisk on 2 servers PBX300 and PBX200. PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap device. Softphone from PBX200 can talk to softphone on PBX300 but no outgoing call from PBX200. I can call from PBX300 outside but I am unable to configure soft Phone defined in PBX200 to dial out side using PBX300 Zap devices. I am geting error

Hi @all, does anyone know how many ip adresses as aliases and vifs may assign to a single DomU? At the moment it seems for me that ~ 30 ip adresses are max. Kong regards -- - Dominique ''NetAndroid'' Schramm - Twitter: @NetAndroid_BY - Blog: http://schwarz-weiss.cc/ Hinweis: PM''s bitte ausschließlich an dominique.schramm@piraten.it! _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hello I am trying to configure Asterisk on RedHat Linux 9 with one X100P one port card and one USB one port FXS card. I can modprobe wcusb but ztcfg always return ZT_CHANCONFIG failed on channel 2: No such device or address (6) error message. Also unable to config outgoing call using SIP SoftPhone. Any working examples of configuration files is highly appreciated. I mentoned followin lines

Hello I am trying to setup Asterisk on 2 servers PBX300 and PBX200. PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap device. Softphone from PBX200 can talk to softphone on PBX300 but no out going call from PBX200. I can call from PBX300 outside but I am unable to configure soft Phone defined in PBX200 to dial out side using PBX300 Zap devices. I am geting error

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. If i pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if i dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. When I pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if I dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

Hello I am currenly trying to setup Citel Link with Asterisk. So far i don't have any luck in able to assign IP address to Citel Link channel but still thought not a bad idea and get some one's view about Citel channel bank. Please let me know your experience while setting up Citel and Asterisk. I will appreciate if you tell me the way to talk to Citel Link using Hyper Terminal. Thanks

Hello I am trying to setup ChanIsAvail function in the extensions.conf file so that user should use the available channel to call out, but immediately after the function like, zap channel hangup. Here is the copy of my extensions.conf file and messages display on consol while making the call. Please help me to fingure out this issue. Thanks Deepak Extension.conf : exten =>

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. If i pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if i dial any digit than there is no ring on Analog phone after hangup. Log's looks like this