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Hi all of you. I am trying to configure voice mail in asterisk and i am facing problems. I have found following warning message in /var/log/asterisk/messages -------------- No application 'Voicemail' for extension (macro-mainmenu, s, 5) I have configured voice mail accordingly in extention.conf [headoffice] -- ------------ ------------- exten => _63,1,Macro(mainmenu)

...in g?nderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekk?rler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recip...

hello, i have trouble with nat + sip outgoing call.when make an outgoing call to a sip gateway, i have no sound. i have 2 sip gateway, one is asterisk. asterisk is on public ip and private ip other sip gateway is on public ip phone are cisco and grandstream on private ip on the same subnet as asterisk. phone are connected by sip to asterisk (i have try with or without nat=yes) incoming call

...in g?nderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekk?rler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recip...

...in g?nderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekk?rler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recip...

...in g?nderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekk?rler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recip...

Hi; Have two grandstream Budge tone...Connected them to the network and able to make call to/from them. But when the coming call answered, I can not hear any voice and also my voice is not heart... I am able to hear voice only if I pressed the hold button and take the call again....This problem also Occurs in calls from x-lite to cisco7940... Does anybody has any idea or documentation

I have a Cisco 7940G IP Phone. I am trying to load the firmware to SIP 3.2. The Phone just hangs in Defaulting CM to TFTP Server. It doesn't do anything else after that. I also have two other 7940g's that are the Universal Application Loader mode and say Protocol Application Invalid. I need to know what I can do to fix both these problems. I am running Asterisk@Home version 1.0. I have to