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Hi, I am having trouble making a selection when I call a number and need to make a selection to go to an extension with my polycom phones 301. Anybody have an idea how to fix this problem? Thanks in advance. Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax: 972-999-4113 Toll Free: 1-877-801-5511

Under asterisk info: Sip registry 12/12 76.xxx.xxx.xxx D N 5066 UNREACHABLE 11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE 10/10 76.xxx.xxx.xxx D N 5062 UNREACHABLE All these IP phones are behind NAT. What could be the problem? Thanks in advance. Jarga -------------- next part -------------- An HTML

Does anybody know why am getting a lot of static and sometimes dropped calls from my asterisk server. Vitelity is my number provider if it matters. Thank you Jarga Jallow -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071105/3ea1de51/attachment.htm -------------- next part -------------- A

I am getting this error under system info: File Line Command Message common_functions.php 314 file_exists(/proc/scsi/scsi) the file does not exist on your machine Does anybody know how to fix this? Thank you in advance Jarga -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a Linksys sipura phone which does not dial ext 26 only, every other ext works. When I dial ext 26 it show to:0 instead. Does anybody know how to fix this? Thanks in advance. Jarga Jallow -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071121/fd2d139f/attachment.htm -------------- next part

I am trying to achieve a Flash-like menu with OnMouseOver effects on 3 divs below. Unfortunately, the javascript resets when the mouse is rolled over the text and if you go mouse crazy on the divs, the script seems to remember where it cut off, and doesn''t roll down completely afterwards. Is it a good place to ask for help? I couldn''t find any Scriptaculous forum (only

I have a bit of a Xen theory question. I have a Debian 5 Lenny x64 Dom0 currently running two DomUs. One is an x64 built via Xen-tools, and it seems to be running great. The second (known as Alpha) is having some troubles, as I mentioned in my post "Ubuntu 8.04 DomU crash" to this list on 5/12/09. It is an x86/32 Ubuntu 8.04 that started life as an HVM and was rebuilt into a PV where

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

Asterisk 1.0.7 Has anyone got the hint function working, and maybe with the GXP2000. I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment trying to get the LED's to light up. On ext 690, button 1 is setup for ext 691, I did this using both methods 691, and <sip:691@192.168.69.1> On ext 691, button 1 is setup for ext 690, I did this using both methods 690, and

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

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they

I am trying to set up the BLF on a GXP2000. Currently what I have is extensions.conf: [globals] polycom430=SIP/101 [internal] exten => 101,1,Macro(voicemail,${polycom430}) [macro-voicemail] exten => s,1,Dial(${ARG1},10,tT) exten => s,2,VoiceMail(u${MACRO_EXTEN}@default ) exten => s,102,VoiceMail(b${MACRO_EXTEN}@default) [ext-local-custom] exten => 101,hint,${polycom430}

Hey guys, I'm looking for some realworld specs on somebodys machine that will work with the Digium 4-port T1/PRI card and that will support 96 simultaneous calls. Dell is soon to release the PowerEdge 1850: 2U, Dual 3.6Ghz Xenon, 1Gb DDR2 RAM, Dual 36GB Ultra320 SCSI RAID, Hot swap Powersupply, one 64bit 133Mhz PCI and one 64bit 100Mhz PCI for about $3,000. Tack on a 4 port Digium card and

Hello, I configured both asterisk and grandstream 2000 accourding to howtos on the web.. And everything seems working fin. But if i reload asterisk grandstream stops working with BLF. I need to restart the phone to enable BLF again. Any clues??

One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets

I've tried to use a Cisco power injector to supply power over Ethernet to a GXP2000 phone without success. Although when I plugged these phone to a PoE capable Cisco Switch it worked without a problem! Knowing that all these three equipments implement IEEE 802.3af protocol, why doesn't it work with the Cisco power injector? Anyone also had this problem before? Thanks, Ricardo Carvalho.

Guys I just read on the wiki: "2005-07-19 - long awaited extension lights (hint priority) and call pickup on various phones work with newly released asterisk patch digium bugtracker - feel free to test and report findings to the bugtracker to have this commited to cvs." How does this work? And will it work only on certain phones or can it work with the gxp2000?

Hey all First off, ive just switched to using mongrel_cluster and pound - its working a treat! Its far far faster than fast-cgi: operations that took a few seconds on fcgi now are almost instant with mongrel. Excellent! However, I''ve been trying to decide how best to utilise my equipment most effectively. I have a dual pro 3Ghz Xenon with 2GB RAM, and the server will be running a few

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems