similar to: SIP phone receiving but not transmitting

Hi guys I'm at a loss in getting ./configure to complete successfully with asterisk 1.4.6 on Fedora 7 x86_64, as it complains about no termcap support, even though it is installed (see below). Any ideas where to go next? checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no configure: error: *** termcap support not found [root at mail3 asterisk-1.4.6]# rpm -qv termcap

Hi guys I'm new to this list and I imagine this question has been asked before, so feel free to point me to the correct references. My question is, how do you install asterisk on Fedora Core 3, with all rpm updates, seeing as there is no kernel-source rpm anymore? Thanks for any advice. -- _/_/_/_/ _/ _/ _/ _/ _/ _/ _/ _/_/_/ _/ _/ _/ _/ _/ _/

Hi all I'm fairly new to Asterisk, so be nice :-) I was wondering if anyone has been able to get the Welltech K1000A USB phone working on Linux. I see audio and HID drivers loaded when it is plugged in to my Fedora Core 1 laptop, but that's about all that happens. I've searched all the usual places (FAQ, Google, etc) but not found anything helpful. Asterisk is working fine with

Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY

On Sat, Oct 26, 2019 at 07:08:02PM +1100, Bill Maidment wrote: > On 26/10/2019 12:08 pm, David G. Miller wrote: > >The corresponding system-config-printer rpm from Fedora 28 appears to > >work.? Not the best solution but a solution. > > > > But no longer available that I can find. I presume it must be > version 1.5.11-13 to match the -lib version On my C8 VM I uust

HI , I installed asterisk in fedora core 3 machine perfectly. and i have 10 units of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i use it as extentions in my LAN ? Asterisk PBX alredy there. I didn't try to do any configurations of any files . What are the configurations has to be made with asterisk ? Thanx in advance, Luke. Send instant messages

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

Hello, I tried to configure a cisco 7905 IP phone using the skinny channel but I had not much luck. The relevant portion of skinny.conf is: [cisco1] device=SEP000F3487F8E3 callerid="Alex" <123-456-789> mailbox=500 callwaiting=1 transfer=1 context=default threewaycalling=1 line => 500 ; Dial(Skinny/500@cisco1) I set up the tftp server, and prepared the following

Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzzzzzzzzzzzzzzzzzzzzzzzzzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? Regards Bilal

Hi; Have two grandstream Budge tone...Connected them to the network and able to make call to/from them. But when the coming call answered, I can not hear any voice and also my voice is not heart... I am able to hear voice only if I pressed the hold button and take the call again....This problem also Occurs in calls from x-lite to cisco7940... Does anybody has any idea or documentation

Hi I am using asterisk 1.4 on RHEL4 I am sending this mail to the mailing list , to enquire wheter any one had faced simillar problem which I am facing now I am facing a problem which is not able to solve or understand , the problem is that I cannot log into the mailbox from a VoIP hardware phone , while I am able to login to the mail box using soft-phone for the same users

Hi, I search How To transfer call between my SIP phone. I have an PSTN line (X100P) and 10 grandstream budge tone phone. For example I want : - Reveive an external call and send it to SIP/phone1. At this point no problem. - After my receptionnist want transfert extern call at SIP/phone2... I don't known how to properly transfert call.... Thanks

I finally got my Asterisk all setup and everything seems to be working except for menu interaction between my Grandstream Budge Tone 100 and my Asterisk. I have the SIP phone setup to properly connect when pressing the 'Message' button and that's working perfectly. When the menu starts, it says press 1 to read your messages, but pressing 1 (or any number) fails to send. Does anyone

I'm not sure if this is possible, but I was hoping to find an address book that runs on Windows XP that will allow me to select a phone number and send that to my Asterisk. The Asterisk system would make the call and connect the call to a SIP phone (Grandstream Budge Tone-100). Is there anything out there that can do that? Thanks, Dave -------------- next part -------------- An HTML

We have 3 Grandstream Budge Tone 100 phones which are being very fluid on incoming calls. They are set up as extensions 2501, 2518, and 2536. When calling out to another phone, they always identify themselves correctly. But sometimes they will respond to the wrong incoming calls. (By respond, I mean that the phone rings and if someone picks up the receiver, the call then goes thru.) For

On Mon, 2019-05-27 at 14:05 +1000, Bill Maidment wrote: > > ---> Package libgpg-error.x86_64 0:1.13-1.el7.centos will be an > > update > > ---> Package libgpg-error-debuginfo.x86_64 0:1.13-1.el7.centos will > > be an update > > ---> Package libgpg-error-devel.x86_64 0:1.13-1.el7.centos will be > > an update > > The update for libgpg-error.i686

The corresponding system-config-printer rpm from Fedora 28 appears to work.? Not the best solution but a solution. Cheers, Dave On 10/25/19 5:33 PM, Frank Cox wrote: > On Sat, 26 Oct 2019 10:17:53 +1100 > Bill Maidment wrote: > >> I have also got MATE 1.22.2 running, but I don't have access to Printer >> configuration in MATE. > I think it's missing. On Centos 7

I have a quick question. I am using Asterisk 1.4. I have a user that has changed phones (grandstream budge tone 200 to a polycom 330). I have changed the sip.conf and extensions.conf. I have also unplugged the old phone and plugged in the new phone. I get the ext showing on the phone, but when I do a sip show peer 5000 the old ip address and phone show up. I did a sip reload and a dialplan reload.

On Mon, Dec 23, 2019 at 11:34:32AM +1100, Bill Maidment wrote: > On 23/12/2019 11:06 am, Fred Smith wrote: > > >I can find no way to do it with pavucontrol, nor the default mate > >sound tool. > > In my SL7 Mate system I use the Hardware and Output tabs in > System->Preferences->Hardware->Sound > > Cheers > Bill Thanks bill. But nothing I do there

Hello: Hoping someone might know how to resolve this (probably an easy one). I have one Asterisk PBX with a single NIC and an FXO card with PSTN line attached, and one IP phone (Budge Tone 100) on the LAN. Via the phone I get no dial tone, and dialing 9, <number> doesn't allow me to dial out. I also need the phone to ring when the asterisk PBX is called. I have modestly tweaked the