similar to: RE: Asterisk-Users Digest, Vol 4, Issue 298

Hi, Anyone knows why there's a limit of 32 callgroup/pickupgroup in * ? It is coded as unsigned int but there's an hardcoded "if( X > 31 )" like line. IMHO, 32 group is very low and I wonder what impact it would have to increase it to 2^16-1 . Anyone?

I'm looking for an enhanced speed dial "dashboard" as DSS (Manager integration) for Operator console integrated in a voip phone (softphone or hardphone, opensource or commercial) to diplay the status of phones (sip, zap, iax...) connected to asterisk. I see in snom site the snom 220 with keypad 220. Can it display the status of internal and external lines (free, busy..) and

Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's "einen Ruf heranholen". It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to

Hi, new to this list, so a little prelude to my issue with Dovecot. We have been using JavaMail against Cyrus for ages, and developed Webtop, a huge Java web collaboration application running on them in production in various installations for all this time. Recently we had to run the same software against Dovecot pre-existing accounts running on Nethesis NethServer solution. After some time of

I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while

Got one weird one and one prob easy one. 1. I have upgraded our BT101's to Program--1.0.5.0 Bootloader--1.0.0.17 HTML--1.0.0.34 VOC--1.0.0.6 after doing this i have some phones on different subnet's ie 255.255.255.248 or .192 or .252 and i am now unable to login to these phones from different subnet's . I have one at home which is on a .248 ( Using an external IP for the phone )

I've searched the lists but I didn't find anything exactly like this. I have two Grandstream BT101 phones connected to an Asterisk. Periodically, for reasons that I can't determine, one or the other (or both) of the BT101s decide(s) to go on permanent busy. Dialing that phone gives: -- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack

I've seen lots of stuff go around about Grandstream firmware levels (in my case specifically the BT101/102). I'm just wondering what the currently accepted 'best' firmware version is to use? After seeing stuff going around about buggy firmware I want to know what I'm getting into before upping past my current 1.0.5.11. It's relatively stable, and the last thing I want

Hi, I'm testing a Polycom IP600. With firmware version 1.1 the phone reboots at any time. With firmware version 1.2, the first reboot was an endless reboot. Then I moved the phone to another lan port, then it worked fine. Then I installed again in the initial lan port and the phone works well. However after some time of inactivity (1 hour?), the IP600 stops to send and receive calls.

Hi everyone, This is off topic and is for GS technical support really but it seems that there are a lot of Budge Tone 100/101/102 users out there. I've got a Budge Tone-100 (101 - without the extra 10base ethernet connetion?) here. I changed the configuration through its web based interface and I clicked the reboot link. But then something went wrong and ever since then it doesn't

Some CVS upgrade in the last day or two has broken the recognition of DTMF eg in Voicemail. I'm running the latest CVS as of now. I'm getting the error... *CLI> -- Executing VoiceMailMain("SIP/phone1-e0dd", "") in new stack -- Playing 'vm-login' (language 'en') **Here I push a button** May 9 18:26:18 WARNING[98311]: chan_sip.c:5027

In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register => number:password@proxy01.sipphone.com extensions.conf: [from-sip] exten => s,1,Dial(SIP/111&SIP/117) exten => 111,1,Dial(SIP/111,20) exten => 117,1,Dial(SIP/117,20) 1. The calling user

Hello All, I am looking for a way to do priority call ringing. That is when a caller places a call to another party, they can indicate that the call is a priority and get a different ring to occur (ring cadence) on the called parties phone. This would be synonymous to an intercom ring on a key system. After some investigation, I have come across the ability of the GS BT101 which will ring

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

Any ideas ????? Jonn -------- Original Message -------- Subject: [asterisk-users] Internal LAN echo problem Date: Wed, 24 Oct 2007 08:34:32 -0500 From: Jonn R Taylor <jonnt at taylortelephone.com> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users

Hello all, This email is intended rather informative than questioning. While developing some script-generated dial plan, we figured out that there are differences between Snom 190's and BudgeTone 101's relating to transfers. It appeared that the 190's will have their own 'Caller ID' set as the 'CALLERID' variable in astersisk when transfering a call, while the

I have used the handytone 488 from grandstream in the past.... However I need to be able to send a number to a unit like the 488 and have it dial out. Is there a unit like this available? Basically a 488 unit that can place a call out. Jerry

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm using a TDM400 with one FXS and one FXO module (developer kit) and I've been testing termination from SIP phones to PSTN and it works fine, but asterisk accounting is doing something strange (for me). Scenario: 1 - extension 1009 (SIP phone - BT101) 2 - Zap/4-1 (TDM400 FXO module) extensions.conf: [dialout] exten =>

Up until yesterday I've had a lot of high pitched noise when connecting a BT101 to the PSTN via the X100P. I was using an Asus A7V133 with raid motherboard and an 850 AMD Duron. Over the weekend I thought I'd try another machine. I had an HP Vectra 400 Mhz PII MMX with 128 Mb RAM available, today no noise at all. Now I must see if the Vectra is up to the job. -- Dave Cotton

I just spent the last hour or so trying to get this firmware to work across a NAT with no success. I have a GS BT101 working through the same NAT, so I don't think it's the NAT itself. I have a STUN setup in * and pointed the UIP200 to it and I tryed several combinations of nat= in the sip.conf and in the config files for this phone. No luck(yes, I did a reload now with each change in