similar to: Grandstream, SIP encryption

Since firmware 1.0.3.81, unless I'm imagining things, Voicemail2 seems to be having problems. The Grandstream and sip.conf were set to RFC2833 now with that setting I get extra digits during "Mailbox" and "Password" phases. 222001 instead of 2201 for example. When both are changed to "SIP info" there is no problem. But what is the new setting "DTMF Payload

I have an Asterisk 0.4.0 install working with two grandstream budgetone 100 phones, gnophone, and kphone. This is a private network segment (172.17.x.x), with the PBX configured on my outbound firewall which has a public address (66.x.x.x). - I can make calls between phones - all extensions are working. - I can make IAX calls to IAXTEL. No problems (apparently gsm only) - I can call SIP phone

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

Does anyone have a working example of how to use the "switch" directive to peer two Asterisk PBXes? -- - Ian C. Blenke <icblenke@nks.net> (This message bound by the following: http://www.nks.net/email_disclaimer.html)

I want to create a new array which selects values from an original array based on a function of the indices. That is: I want to create a new matrix Vnew[i,j,k]=Vold[i,j,ks] where ks is a function of the index elements i,j,k. I want to do this WITHOUT a loop. Call the function "ksfunction", and the array dimensions nis,njs,nks. I can do this using a loop as follows: # Loop version:

I''m creating a new domU using ''virsh create <xmlfile>'' without specifying the MAC address within the XML file, under the assumption that the dom0 or hypervisor generates the MAC address for me. Next I''d like to get the MAC address that was assigned to this new domU, and I don''t see a way to get that information from the command line via virsh

Hi All, I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as well as any configuration in asterisk thats needed, on this phone? Thank you, James -------------- next part -------------- An HTML attachment was scrubbed...

I have followed a combination of instructions from this post and have been able to get an Ubuntu paravirtualized instance running under snv_96. Is there a way to "list" the fact that it is paravirtualized? For example, xm list, virsh list, etc. thanks -- This message posted from opensolaris.org

This is important so please read this message very carefully. There is a DoS for Ruby''s cgi.rb that is easily exploitable. The attack involves sending a malformed multipart MIME body in an HTTP request. The full explanation of the attack as well as how to fix it RIGHT NOW is given below. Most of the work was done by Jeremy Kemper and Jamis Buck. They did all the work of building the

On Zed''s suggestion, I caught two new spinning mongrels and sent a SIGUSR2. The code appears to be stuck in read_multipart for both processes: # kill -USR2 6109 ** USR2 signal received. Thu Sep 21 14:55:39 EDT 2006: Reaping 1 threads for slow workers because of ''shutdown'' Thread #<Thread:0x419d7ce0 run> is too old, killing. Waiting for 1 requests to finish, could

Does anybody has got the some problem? The grandstream ATA 486 schould support almost all codecs, but it doesn't work. I get the following message when I force the use of different codec WARNING[9529]: chan_sip.c:2765 process_sdp: No compatible codecs! Feb 3 11:17:15 NOTICE[9529]: chan_sip.c:7395 handle_request: Unable to create/find channel What could I do to see some more detailed

Dear All Among all the readers anybody have ever work on Granstream device GXE2504A which act as ippbx and having GUI to configure and maintain. We are facing one problem with this device, thsi device reply or adding codec like ilbc,G.721 which is not supported by our Asterisk server or our SBC. We want to disable this codecs, but form available GUI we not able to see any option to disble it.

Hi All We have a private network setup (no nat) with three types of phones connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco 7940 IP phones. When we ring polycom to grandstream or grandstream to polycom then both phones can send and receive voice fine and all is well. When we dial any combination of Cisco and either Polycom, or Granstream the Cisco, no voice is being sent

Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as the 102D. And to make matters worse he starts the bid at $90.00 Beware. http://search.ebay.com/search/search.dll?query=sip+phone&ht=1&sosortproperty=1&from=R10&BasicSearch= -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

I think I have something misconfigured regarding voicemails. They work great, I have this setup: Sip.conf [ext1] Context=phones Mailbox=201 Voicemail.conf [home] 201,password,name,email@mail Voicemail delivery and all works great but when I check sip extension ext1 (analog phone using a Granstream ATA 286), the stutter tone signaling message waiting does not work. Anything wrong with

First off: Our clusters are LVS balanced Apache 2.2.3 + mod_proxy_balancer + gem mongrel 0.3.13.3 / mongrel_cluster 0.2 + memcached / gem memcache_client + gem rails 1.1.6 on debian boxen, and a pgcluster backend. On 2 of our deployed clusters, we are getting the "spinning mongrel" problem. As the clusters are very low volume right now, it takes days to collect a spinner, making it

Good evening all. I'm having this strange behavior when dialing two or more simultaneus calls via IAX to other * boxes. Sound starts to have more latency, wich increments until it's almost impossible to talk (6 or more seconds), I try this calling with two grandstreams, one grandstream one tdm410p, one tdm410p and sjphone, sjphone and one grandstream, the result are similar.

Anyone in need of some of these? Garrett Smith Sales Executive garrett.smith@b2llc.com B2 Technologies 454 Sonwil Drive Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales Specializing in New and Used equipment from vendors including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, Granstream, Snom, Mediatrix,

Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten => 0,1,Playback(pls-wait-connect-call) same=> n,SIPAddHeader(Alert-Info:;info=ring3) same=> n,Dial(SIP/310&SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com