similar to: Grandstream GXP2020 + Asterisk 1.4.11

Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on .... not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) Thanks! Kind Regards, Erik

Hi, I'm using a GXP2000 (that's sharing the same GXP2020 firmware file) with the latest 1.1.5.10 beta release. It's working since a week and seems working very well. Before I was using the 1.1.5.3 and I had no problem. 1.1.4.xx versions, instead, are not performing like that one (audio, deadlocks and other minor issues). You can find a lot of info and old firmware versions at this

Thx John !! Hmm I found now on voip-info.org a lot of Beta releases which should fix my problems... Kind of strange whats going on with Grandstream devices and their firmware ... If you install the latest "official" release you can expect a few troubles with Asterisk 1.4.11 (one way audio --> randomly, dropped calls). So you have to install the BETAS whether you want or not...

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Hi list, I'm wondering if there's a way for multiple users to share the same voicemail box and have their BLF flashing when voicemail comes, i.e. in a home phone system where there's a general vm for everyone. I'm using couple Grandstream GXP2020. Any suggestions? Kelvin Chan | Positronics Ent. Product Development | | unit 272

Hi, Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors (or more) ? This could be very useful to support extended presence, for instance. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm

Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where the phone did not send rtp keepalives when on mute (resulting in disconnect from tech support hold and concalls) A side effect seems to be that Asterisk pops the following warning on the console... Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short Grandstream say they are not sure what it is but

Hi all, Im using Asterisk 1.4.11 and I want to proceed some time and date operations in my dial plan. (for a time shifted callback). Should look like: CURRENT TIME + x minutes. Of course it should increase the hours for example in this case: 10.59 + 5 minutes = 11.04 I guess I've to use the math function in 1.4 but how can I manage easily the time operations? Kind Regards, Erik

Can anyone who has used both comment on the pros and cons ? Need to buy about 30 of these, for a small company with limited IT support. Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

These is my scenario. Asterisk 1.4.16 Zaptel 1.4.8 Grandstream BT200 Grandstream GXP2020 Grandstream GXP2000 For some reason the end user ask to configurate son direct access like *01,*02,*03 thru *78. After they began to use these direct access, I cant place a 3 way CONFERENCE. I remove the direct access, but i dont know if one of them block the CONFERNCE. Do you know if i can make

Hi, Is there a possibility to control sound levels (higher / lower) in Asterisk (so the codecs). Somebody asked me to evaluate that but I didn`t found any documentation about. I have the opinion that for these (audio) things the end user client is the only part where I can tune around. Problem is for example a (Austria) ISDN --> Asterisk --> SIP / IP ---> (Romania) Asterisk

I call the firm from my portable at home (zoiper softphone). I have internal extension 60, and I call the internal SIP-client 10 at the firm via an IAX-connection over internet. My colleague at phone 10 answers my call. I ask him to transfer me with my colleague at extension 50. He then presses "transfer" on the grandstream GXP2020 (I get music) and dials the number 50. Phone 50

Hey list ! I'm getting the feedback of a customer that a conversation is like half duplex : when he talks, the other end of the call is no longer heard. What could be the cause of these drop-outs ? A call that is coming in from the PSTN is routed through an IVR-system to the correct internal SIP-phone (Grandstream GXP2020). Where do I start searching for this problem ? -------------- next

Hi, I used Grandstream (gxp2000, gxp2020) and Snom (370) SIP Phones, but with 2 extensions BLF status does not work correctly. have someone ever tested a Sip Phone with more then 60 BLF without problems? Can someone suggest me model and brand? Thanks, bye. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, We are in the process of splitting our phone system into two separate logical systems for our two departments. One of the goals of this switch is to restrict members of one department from transferring calls to the other, but not restrict them from calling that department themselves. So what I need to know is how to detect whether a call from a member of that department is a transfer or

Hi All Sorry to be a bother again but seems like I just cant get away from the problems. This time my problem is that *sometimes* a user cant transfer a call from one extension to another, I have narrowed down the problem to it only happening to calls from outside the internal system. The wierd thing about the problem is that it comes and goes one moment the user can transfer, and the next

I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13. Is anyone else getting the following error in the asterisk console: [Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short every couple of seconds when a handset is in a call? I didn't notice this happening when I was using an older GXP2000 with the same firmware (doesn't mean that it

Hi List, I?m having problems compiling chan_misdn: asterisk:/usr/src/chan_misdn-beta-0.0.3-rc4 # make install cc -ggdb -Wall -D_GNU_SOURCE -Wno-missing-prototypes -Wno-missing-declarations -fPIC -I/usr/src/asterisk/include -DAST_CONFIG_DIR=\"/etc/asterisk/\" -I/usr/src/mISDNuser/include -I/usr/src/linux-2.6/include -I/usr/src/mISDNuser/i4lnet/ -Wall -c -o chan_misdn.o chan_misdn.c

Hi, Im sure somebody out there had the same "problem before. IF a call comes in with suppressed caller id (Call Centers, etc.) 'asterisk' is shown as CallerID. Can I change somewhere this behaviour to display like ' Unknown' ? Thanks! Kind Regards, Erik