search for: gxp280

I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for testing. In addition I register a zoiper SIP soft phone. For the Grandstream I have busylevel=1 in sip.conf. If I place a call from the GXP280 to zoiper and then put that call on hold from the zoiper side and then call GXP280's extension, asterisk indicates the phone is ringin...

I sent this email a few days ago but did not see any responses to it: > I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for > testing. In addition I register a zoiper SIP soft phone. > > For the Grandstream I have busylevel=1 in sip.conf. > > If I place a call from the GXP280 to zoiper and then put that call on hold > from the zoiper side and then call GXP280's extension, asterisk...

...it worked again. I reinstalled 1.6.0.3-rc1 and it stopped. Looks like a problem in the software to me. Following the same steps using the same code for the AMI and conf files for * I get bad behavior in 1.6.0.3-rc1 and good behavior in 1.6.0.1. I have this action: Action: Originate Channel: SIP/GXP280-18 Application: queue data: tqe ActionID: callandqueue Callerid: 1111122222 Async: true And I get this response: ---->Response: Error ---->ActionID: callandqueue ---->Message: Channel not specified Is anyone else seeing anything like this? -- Jim Dickenson mailto:dickenson at cfmc.c...

Hi All, I have working asterisk version 1.4.24. I have a blind transfer issue with grandstream bt200. I have updated the latest firmware to the phone. The phone is sending the *refer* to asterisk but asterisk is not able to connect with the another call that i have checked in sip debug. I am using transfer button of the grandstream phone. Can anybody provide help for this issue? Thanks in

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! During tests with a Grandstream GXP280 phone, I found that pressing 'hold' button, the other extension (Qutecom softphone) is put on hold but without music. Then, when resuming the conversation, I listen the other user again but he/her no longer listen to me. When from softphone the same test is realised, it does not happen thi...

Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue

Can anyone who has used both comment on the pros and cons ? Need to buy about 30 of these, for a small company with limited IT support. Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n

I'm trying to understand why vendors keep making 100Mbps integrated 1-port switches in their hardware SIP phones. Even the recently-announced D40 and D50 Digium phones are limited to 100Mbps. Only the more expensive models (like the D70) can run at 1000Mbps. However, you can't expect a firm with hundreds of extensions to buy the most expensive model... And gigabit speed is important when