similar to: Bad Pick up line

Many thanks in advance for anyone that can offer help on the following questions: Asterisk Box Using Asterisk@Home build and updated Asterisk to v2.1 P4, 400 Mhz, 384Mb RAM, 40Gb HD 4 OEM X100P Cards Phones Grandstream GXP-2000 2 * Grandstream BT-100 HandyTone 486 Sipura SPA-3000 Questions 1) When someone calls in to one of the FXO lines, there is a 3-4 second delay before the configured

Yo all, I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P. The call wil sound OK at first, but after 10-20 minutes, the audio will start to crackle. Soon after that, this crackle turns into a continuous noise and the parties won't be able to hear eachother anymore. It also sometimes happens that the party on the TDM400P hears a very loud, short-delay echo of themselves,

All, I know the question has been asked before, but any of the solutions posted in the past have not solved my problem. I have got a Asterisk setup using a P4 1.8 / 512mb server running Redhat Enterprise 3 and 3 grandstream budgetone phones (plus a couple of xten clients on windows) and I'm at advanced stage of testing to see if asterisk will fill our needs as a PBX using voice over IP

https://bugzilla.samba.org/show_bug.cgi?id=9813 Summary: --resume parameter to improve speed of dropped/partial transfers Product: rsync Version: 3.1.0 Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P5 Component: core AssignedTo: wayned at samba.org

I can't seem to make the "Hold" button function on the GS BudgeTone-100. I'm trying a procedure like this: 1) On a call 2) Press "Hold" button 3) Hang up phone What I expect is for the call to go on hold until I pick up the receiver again. (Like my SPA-2000's, except it's a flash, but I can hang up the phone and the call waits there for me to pick it up

We are running asterisk with a single POTS line for local calls and a voip line for long distance. Whenever we receive a call on the POTS line it is more than likely, but not always, going to have significant distracting echo. In addition to that there is occasional heavy crackle or static. I have tried to follow the guidelines at :

Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration ====================== Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am

I have the MusicOnHold feature working great when called from ATA-186 extensions. It's pretty cool. However, when I call from a BudgeTone-100 phone, no music is heard -- instead it continues the ringing feedback and acts like the call is unanswered. At the same time, I can call from (multiple) ATA-186 extensions and hear music as long as I like. How can I debug this? As far as I can tell,

I am setting up a small Asterisk system for use at home. I have one Budgetone 101, one Cisco 7960 and two Xten lite softphones. So far everything is working expect for an issue with the Budgetone. When a call is placed between the Budgetone and any other phone, the call is setup and sounds good. If I hang-up on the Budgetone, everything is ok. If I hang up on the other phone, the Budgetone

Does anyone have X-Lite build 1059 working fully with Asterisk? The GSM Codec works very well now but we have problems when using G711 in that when I setup a ping between the two sites and then watch the latency, it steadily increases and starts at about 150ms and goes up to 2500ms within about 20 seconds. I have not investigated fully but I guess that its sending ever increasing size packets.

Just for the record and to possibly help with others who get BudgeTone phones. My asterisk box is behind NAT, and I use Vonage, NuFone, and iconnecthere for my "POTS backhaul." On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102. The BudgeTone definitely has issues wrt the RTP stream and NATting, although unfortunately I haven't yet been able to dig

I have a strange problem in that I have put a budgetone out on the internet that connects to my * server that's behind a firewall. They can call me I can call them and it works fine. However, I have setup a link to sipdiscount on my * server. If the budgetone user calls via my * box to sipdiscount all the budgetone user hears is silence and the called person hears silence as well when they

I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my

Dear List, I'm currently investigating if the argument dispatch mechanism based on `...` could somehow be "generalized" to scenarios that involve `r` recipients located across `c` calling stack layers *and* combined with the S4 method mechanism (for those interested see

Hi, I have had this issues for ages but have ignored it, but my handsets get faint background crackle on the FXS port. When I connect the handset directly into my PSTN, it goes away complete, so I am generating this from somewhere in my asterisk box. I get it immediately when the phone goes off-hook and it stays through the dialing and call progress. I can't think of any other devices

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

Hi, After a few attempts, I've managed to grab the files from CVS and build it on a rh redora box I've setup especially for Asterisk. Firstly, we're new to the asterisk scene, so please excuse any "lame" questions which may follow.. We're a new voiptalk.org customer. We have purchased the voip phones (budgetone 102's) and set aside a little box to run Asterisk on.

Server A has 4 PRIs coming into it, and provides a SIP connection to Server B. When we call into server B via a DID from one of Server A's PRIs, we get crackly sound on recorded playback only (prompts, IVR, voicemail instructions, etc.), but not on actual live voices once you're talking. Server A provides this same service to other servers and problem does not appear in those cases.

Janko, I'm not entirely sure I understand your proposal. Are you suggesting methods be dispatched based on the *contents* of ... (ie which arguments are in there)? This seems like it would be pretty different from how dispatch behaves now, which is entirely class based. Even the dispatching based on ... via dots methods is class based, having nothing to do AFAIK with the argument names. From

Hi, I'm considering giving the Grandstream BudgeTone-102 phones a try. I've been using Cisco 7960's to date, but the low cost of the Grandstream phones are hard to ignore. I have two questions: 1) Does the message waiting indicator on the BudgeTone's work with Asterisk? 2) The one line 12-digit LDC concerns me a bit. Is the LCD able to display both the CID number and name on