On Wed, 6 Jun 2012 16:57:11 -0400 Hai Nguyenwrote: > A calls B. B attended-transfers the call to C using (polycom, cisco) > phones transfer button. C does not answer the call. A gets Bs > voicemail. However, if B blind-transferred the call to C an..
Im new to asterisk, currently working through Asterisk, The Definitive Guide.I have a couple Grandstream 200 phones and a Polycom 501 to play with.The Grandstreams were very easy to configure, but the more capable Polycom almost drove me crazy.Th..
Daniel Seagraves wrote: > The boss wants to move from landline service to VOIP service as a cost-cutting measure. We have one voice line and one fax line. The telco is billing over $100 a month for the two. Were using Hylafax for faxing and a PBX ..
The boss wants to move from landline service to VOIP service as a cost-cutting measure. We have one voice line and one fax line. The telco is billing over $100 a month for the two. Were using Hylafax for faxing and a PBX for the voice line. Our exist..
During testing with the Digium phones I have run into a problem where I try to make a change to the sip device name. I make the device name change in sip.conf then make the matching change to the lines in res_digium_phone.conf. I then do sip reload ..
Kevin, I have two asterisk boxes with the same issues. Box 1: asterisk ver 188.8.131.52 Box 2: Asterisk 184.108.40.206 setup: CDMA PhoneCDMA Media Gateway WCMAsterisk voice mail The calls are SIP Based.DTMF collection is when the user is entering a password ..
Thanks Kevin and Yaroslav, Sorry was out of town. Sorry I forgot to mention that Iam using an VOIP GSM gateway to connect to PSTN. Kevin, I am have decided to use Sangoma CPA. Do you know of any other options that are easier to integrate with? Yarosl..
I have a question regarding DPMA for Digium Phones, if i install the DPMA on my Asterisk Server A, and then, i move the phone to register into another Asterisk Server B, can i install for free another DPMA license for my digium phones on this sec..
I use an SBC to protect my pool of asterisk servers and as trunking endpoint with SIP Telcos. Now Im trying to implement SIP phone registration to be delegated through the SBC, as a proxy. It doesnt work. It just works when I dont use realtime pe..
On 05/10/2012 09:39 AM, Arif Hossain wrote: > I have following sip account : > > Name/username HostDyn > Forcerport ACL Port StatusDescription > demo-alice/demo-alice 192.168.7.47 D > N 1080 Unmonitored > demo-bob/demo-bob 192.168.7.47 D > N 5060 Unmonito..