search for: liangliang

...he system, the 'online' indication will be blinking and on, if logout with type of meeting, then 'meeting' LED will be on, and etc for other scenarios. I found it is quite common in the traditional PABX, however now with more advanced technology, we lost such features. Regards, Liangliang

My MusicOnHold sound is very soft, but when I hear it directly from mp3 playe on desktop, the loudness is quite ok. Wonder whether there is any configuration to change the loudness of MusicOnHold. Regards, Liangliang

Hi, we are using asterisk 1.2.13. When callbacklogin agent transfer call using SIP phone's transfer feature, he is always in busy status and cannot answer any more incoming call from queue until the transferee hang up the call. -- Regards! Liangliang

...are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? Regards, Liangliang

...Seq: 4 ACK Proxy-Authorization: Digest username="1015",realm="asterisk",nonce="6f0fbf9b",uri="sip:2@192.168.0.20",response="10e7de147cb8dc688049204479c26bd5",algorithm=MD5 User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 -- Regards! Liangliang

...e. They sometimes still can make call, but the quality was quite bad. China Telecom's engineers already checked the cable using some E1 test tools, it works perfect, and they even plug E1 into a Panasonic PABX, it didn't have any quality problem. FYI, The card model is TE412P. -- Regards! Liangliang

...the transfer, and last # key was received by asterisk as part of the extension, I have to remove it in dial plan. I can not just shorten the digit time out, always there are some ppl with bad memory. Then is there any option can be set to let the blind transfer work just as in 1.0? -- Regards! Liangliang

When we use mobile to call certain number, we can hear the message like, " The number you dialed is incorrect", "The customer is currently is unavailible" and etc. But when we use asterisk to call the same number, just busy tone. We found that since version 1.0 it support the standard hangup cause, so we base on the HANGUP_CAUSE to fake the error message, but seems