similar to: asterisk@home and grandstream display

I've searched for a few days now without finding an answer. The release notes for version 1.0.5.16 of the Grandstream firmware says it supports attended transfer using replace but the docs haven't been updated so I can't work out how to enable it, or whether it should Just Work. I'm currently using the # attended transfer patch for * but would like to get back to using the

Does anybody has got the some problem? The grandstream ATA 486 schould support almost all codecs, but it doesn't work. I get the following message when I force the use of different codec WARNING[9529]: chan_sip.c:2765 process_sdp: No compatible codecs! Feb 3 11:17:15 NOTICE[9529]: chan_sip.c:7395 handle_request: Unable to create/find channel What could I do to see some more detailed

Hello guys, I ask you to share your experience with your BudgeTone 100.... I have my asterisk @ work and I've bought a GrandStream BudgeTone (SIP phone) and I usually use X-Lite I have plugged my BudgeTone into my home network because I want to be called even at home. I succeed to register my X-Lite with Asterisk from home but I can't do that with my BudgeTone. (I don't know

Ronald, Grandstream products have a one year warrantee. If you don't have any luck with Pulver, contact us and we can probably get your phones exchanged. Please don't assume that your experience with Grandstream is typical. We sell a lot of these phones and the overwhelming majority of the purchasers are very happy with their units. The quality has improved tremendously over the last

Hi All, I'm trying to connect to a SIP carrier who never connected with Asterisk. I managed to connect with a sipura phone or a grandstream, no problem. When I configure asterisk, I'm able to send out calls to the carrier no problems, however, receiving calls doesn't work, and I keep getting the following messages: <-- SIP read from 69.xx.xx.xx:5060: INVITE

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension

Egad, not again with Broadvoice! Anyhow, I recently installed AAH and configured my TDM11B and got that and some SIP phones working. I still have some issues to work out, etc, but my current problem is Broadvoice. I have checked out all of the online resources, including the recent list exchange about the recent changes made by Broadvoice. However, the one thing I have found to be consitent in

Hi all, Just wondering if anyone's come across this issue, and what might be a fix for it: We've got several BT-101's deployed, and upgraded to firmware v.1.0.5.16. The phone can do proper supervised transfer, but _only_ once. If the user attempts to transfer a second time, it won't work. any suggestions/hints/tips are welcome.. Flynn

voicemail notification? -----Original Message----- From: bam [mailto:bam@cqm.co.uk] Sent: Wednesday, December 24, 2003 12:17 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Grandstream 102 flashing display The phone powers up and I can make calls through my Asterisk gateway to other endpoints. However the four leds under the keypad are permanently illuminated and the backlight

I have just successfully got the TFTP file remapping to work such that I can have unique ringtone files for each and every extension. I added the following to my server_args line in the xinetd configuration for TFTP: -m /home/asterisk/grandstream/ringmap.cfg Now the entire line reads: server_args = -v -s /home/asterisk/grandstream -u asterisk -m /home/asterisk/grandstream/ringmap.cfg (There

Hey everybody, I've been playing around with Asterisk (Current CVS Stable dated: Asterisk CVS-v1-0-11/23/04). I've purchased 2 GS BT102 SIP phones. Both have been upgraded to firmware 1.0.5.18. I've also have installed on my desktop and laptop, X-Lite. I've been using these to learn how to setup Asterisk. I've got a Wildcat X100P on order and will be here next week. My

Hi All We have Grandstream 102's running ver X.18. When hanging up after a call has been made the grandstream seems not to disconnect the call and when you put the handset down the phone rings only to pick it up and be on the same call. This is happening quite often and gets very irritating. Can anyone help with this? Regards Doug

My BT100 won't register with my Asterisk server, it always comes back with a 403. I've included my sip_additional (only one to to have the username 2201) and a portion of the sniffer trace (packets 27 & 28). This has me puzzled as I have my SPA-3K working (incoming and outgoing). On my BT100 I get no dial tone, I can't call it (asterisk says the extension is busy) but I can call

OK, now this should be really simple, but I am a bit of a newbie so please bear with me. I have an *@Home box setup with TDM04B and two POTS lines. On the SIP side, I have GXP2000 phones. Most things seem to work, but the users cannot figure out how to transfer incoming calls from one extension to another. Now I am not sure that I have things setup correctly, but is there something

Wiley, There are a couple of issues that we saw while not using this option. 1) sip authentication failures as Asterisk is not able to reach Polycom phones. A typical problem description is here: http://lists.digium.com/pipermail/asterisk-users/2004-December/079251.ht ml 2) DTMF issues for Transfers, Hold or simply to dial extensions. This problem is more pronounced when you are using

Apologies for asking something that must have been asked many times. I'm running *@home v0.4 and can't get the * time to be local GMT. Tried tzselect etc etc and added ntp server addresses to ntp.conf, * still uses system time of EST so call logs are 5 hours behind. Also, e-mail notifications of vm don't appear to be getting sent, I've set voicemail.conf to include a valid

Hello, it is possible to get MWI working with Grandstream and Asterisk? Thanks. -David

Download V 0.4 here http://sourceforge.net/project/showfiles.php?group_id=123387 burn it to an .iso install into asterisk box (be warned it deletes everything on the hard drive but this is what you want right :) it will automatically install Asterisk AMP FOP and Web Meetme read the FAQ here http://asteriskathome.sourceforge.net/faq.html basically if you are using a X100P all you need to do

Hello. I had grandstream working fine to FWD through my firewall. Now I want it to talk to the asterisk server. Did lots of reading, attempts but I keep getting registration errors even though I can call to/from the sip phone from an analog phone on a tdm400 card. Basically. grandstream = 192.168.1.70 asterisk = 192.168.1.1 The error I see is ;- -- Executing Dial("Zap/2-1",