similar to: problems with international dialing

I've got a problem with DTMF, again. My asterisk box is connected with the outside world (PSTN) via a sip proxy. The problem is that for some reason, I need to use rfc2833 for signaling digits to the gateway and inband to accept digits from outside (eg. when someone dials one of our DIDs). It's possible to do this? I've ever tried splitting 'peer' and 'user' part in

Hi, I'm provisioning an office with limited cabling. I'm looking for a desk based wifi phone. Most of the ones I've seen are handsets. Any ideas? Thanks, WILL -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051007/1d2cc49a/attachment.htm

Hi All, I am having a problem with Samba that seems to be a bug. I say this because I read the archives and they described a problem similar to mine and were told that it was a bug in older version of Samba and the latest version should have it fixed. I am running what is essentially a RH 7.3 on a 1.4GHz Athon with 1GB or RAM. I compiled Samba 2.2.5 in /opt/samba-2.2.5 and am using the following

Hi, I'd like to add a feature to my asterisk system that tries to find a user among a couple of locations, and then goes to internal voicemail if the user doesn't pick up. (e,g, an internal extension and a cell phone). The catch is that I want the user to manually accept the call to prevent it from going (for example) to the voice mail on my cell phone. Scenario * Call comes in,

I'm building a dialplan for use with a bunch of GXP2000 desk sets. During testing, we had some user issues surrounding the lack of an on-phone dialplan. Users would hit 9 and sit there waiting for a redial tone, and the GXP would time out, sending just '9' to *, which couldn't do much other than spit back a 404 or play pbx-invalid. I turned on the "early dial" option

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've got Asterisk SVN-trunk-r9059 currently running on Fedora Core 4 w/ 2 eyebeam softphones and 2 Grandstream GXP-2000. At my desk I've got the grandstream and the GXP-2000 I would like to ring both. Using macro-exten-vm and dialparties.agi Macro(exten-vm,200,200-202) the caller is sent to the unavailable voicemail but if I use

Dear All, I am a newbie to Xen and I am trying to set up a Xen VM with more than one guest running on a single machine. As a setup, I have Xen 3.1.0 installed with fc7 core and trying to run debian-4.0 guest image on one of the guest environment. After starting the xm guest domain, the console gets hanged up at DHCP request command, the log is as below: [root@steelbug xen]# xm console

Hi, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it hangs up soon as the other party picks up. I have used different IP phones, VSP's and etc.

BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; } Dear All, I am trying to use the virt-manager to install Xen Guest OS. I got the virt-manager installed and running on fc7 xen dom0 and am trying to install fc9 domU using virt-manager. I have only been able to find the install URL for fc9 (all other links fc6, fc7 etc does not work), but this URL doesnot complete the

Is anyone else having an issue with GXP-2000s and transmit gain? All my other phones are fine on my TDM400P with txgain set at 0, but the GXP-2000 caps at about a third of the scale in ztmonitor. I'm getting people complaining they can't hear me on my GXP-2000s, whereas my Snom 320 and Polycom 301 are great, and my Budgetones are overmodulating. Is there any conceivable fix on the

I started out a few years ago with some SPA-841 sets, because the Grandstream 2000 I thought I wanted was perpetually delayed. The GS had more call appearances, and I didn't want just the 4 max that the SPA offered. As it turns out, with the greater flexibility of VOIP, I don't need 'dedicated' CAs the way I needed them on ISDN previously, so 4 is actually adequate. Along the line,

Hi all, thanks to the all of you. This list is very interesting also for a newby like me. My problem: I just setup my first full working asterisk installation with this config: 1. n.1 GXP-2000 2. n.4 Budgetone 102 3. n.1 TDM400p (3 FXS, 1 FXO) Everything seems to work fine, but the sidetone... it's really annoying! We can hear the sidetone only when we call to the outside (PSTN), it

BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; } Dear All, I was trying to set up a virtual network with fc7 Dom0 and ubuntu 8 (Hardy) & Fedora 9 (etc) as DomU images on AMD Opetron 152 processor based machine. I was able to successfuly establish a virtual network using bridge connection Xen Network configuration. After that I tried to install cpufreq-utils on Dom0 which

Hi, We're having problems dialing out to Asterisk from our Grandstream GXP-200 phones. About 2 of 3 times, when we dial, nothing happens. Looking at the console in max debug mode, there are no messages except the following: Feb 24 10:29:20 WARNING[2475]: chan_sip.c:1208 retrans_pkt: Maximum retries exceeded on transmission 9913b47bcd7 aeb52@192.168.10.100 for seqno 4524 (Critical Response)

Hi When I make a call with the Grandstream GXP-2000 through Asterisk (and SER) to landline using VSP, after I hang up the call the other party are still connected for another 30-40 seconds. I've notice that the SIP BYE is sent to Asterisk, but Asterisk sends no SIP BYE on to VSP. When I use the SPA-941 the call terminates on the other right away soon as I hang up. I have updated the

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my

I have a client with 20 GXP-2000s. Everything seems to be working fine. However, after a couple of weeks of use, the client is having a hard time adjusting to the new IP based phone systems and only misses one feature from their old Lucent system. That is, they had 8 analog lines before and all their old Lucent phones showed a button for each line. So, it was easy for anyone to say,

Hi, We have a page on the wiki just for phone reviews, but I think it needs a bit of format change. Instead of individual reviews for each phone, I think each person should review all phones they have worked with and list the phones they have had access to and rank them in relation to each other. Also each review should have a date so the reader can see how fresh the data is to current.

I'd recommend any of the following, which are all in your price range Snom 300 Polycom IP430 Polycom IP501 Aastra 9112i Linksys SPA-922 Grandstream GXP-2000 Cory Andrews ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Wednesday, November 01, 2006 11:17 AM To: Asterisk

I've finally got a chance to play with 1.0.1.9, and the wiki has been updated. At the moment, I don't know of _any_ bugs with it. I'm yet to play with complex things like early dial, and will update the wiki as I find information. http://aussievoip.com.au/wiki-GXP-2000 I've just been given official permission to offer the firmware as a download, so if you visit