similar to: Asterisk Wish List - Can We do it? Can you add to it?

Folks! Netweb Group, Inc. fully supports connectivity to any Asterisk PBX systems you have and can provide A-Z termination with immediate effect. Any volume is good enough for us, even an amount as small as $1.00 a day will do for us. We will provide connectivity from our Softswitch IP 216.162.116.46. If anyone is interested, add this to your Asterisk IPBX and then email me for setting up a

Hello Guys, After adding money into my sysmaster phone account I am able to make calls outside.thnx _____ From: Mani Nair [mailto:mnair at nvloisp.com] Sent: Friday, September 07, 2007 9:16 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Sysmaster and Asterisk Hello Guys, I am unable to make calls to outside number from some of my extensions.

Hi, I am trying to post this again as I am getting no answers and the support@digium.com bounces... (I have searched the whole list and can't find the answer either) I have installed a 5 user license for G.729 and want to route calls through Asterisk from my G.729 phone to Cisco AS5300 also using G729. Both Cisco and the phone connect using this codec if I do not force the call to go

Seshu, Whats with people who work at Morgan Stanley? You on the one hand bash the opensource software, and then on the other hand a few weeks later, ask for assistance from the open source community for assistance??????????????????????????????????????????? Today, May 11, you request assistance in installing the Areski Calling Card platform, after posting a couple of weeks ago, April 20, this

I've been using it in a test environment with no problems. However, I haven't used it in production yet. I'm doing some voice broadcasting with a PRI and so far I'm content with the performance. -MC -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Akpome Akpoguma Sent: Friday, February 03,

Why don't u attach the setup page of the phone ? Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di FaberK Inviato: gioved? 13 ottobre 2005 17.56 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] PA168S/AT320P Right now, but nothing changed. 2005/10/13,

Good day everyone, A Solution now exists for those of you still looking for the first model of Quad T1/E1 Interface Cards - the T400P and E400P. The T400P and E400P were manufactured by Digium until a few months ago, and were designed and released under the GNU Public License as the Tormenta II by Zapata Telephony. Varion, Inc. is now in full production manufacturing these boards. With the

who has purchased a V400 card from Varion ? I need some help . please help me . thanks a lot

For the benefit of everyone, having H323 Configuration problem due to H245 Tunnel, check the h323 Config embeded at the end. Comment the offending line as under: ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; -----Original Message----- From: Tola Ogunsan [mailto:tolaniye@hotmail.com] Sent: Wednesday, May 25, 2005 1:03 PM To: Kanuri, Seshu (Company IT) Subject: RE: oh323 problems

Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '0426000000' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension

Hello all, I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper. PSTN ---->AS5300 ---->Gatekeeper ---->Asterisk I set up a conference room on the Asterisk sever (Room No 1234). I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper. I manage to get to the start of the conference

Hello, Has anyone experienced a segmentation fault in asterisk for using G729 against an AS5300 in SIP ? I'm having this problem. Also, any other codec I use gives me incompatible media (can be read in SIP DEBUG messages). AS5300 configured for multiple codecs, so is Asterisk. Tried G711u/A G723 and G.729. Any clues ? Regards, Jorge A. Info: Asterisk ver 1.0.7 stable Using AMPortal

Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 "Bad Extension" back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-0000004d == Spawn extension (dialin, 065939191, 2) exited non-zero on

Wiley, There are a couple of issues that we saw while not using this option. 1) sip authentication failures as Asterisk is not able to reach Polycom phones. A typical problem description is here: http://lists.digium.com/pipermail/asterisk-users/2004-December/079251.ht ml 2) DTMF issues for Transfers, Hold or simply to dial extensions. This problem is more pronounced when you are using

I've previously been using G711alaw on both the AS5300 and the phones but feel the need for a less bandwidth hungry codec for those users that are connected behind ADSL and so was investigating G.729 but .. Firstly I found that on my AS5300 I have either G.729r8 or G.729br8 and on the 7940 phones I have G.729a, I'm not sure which interoperate the best with each other and so was wondering

Given the following.. Foo has_many :bars Bar has_many ::widgets Widget has_many :gadgets Gadget has_many :parts All of the following are now possible... @foo.bars @bar.widgets @part.gadget.widget.bar.foo However, I can''t just do the opposite of that last one... @foo.bars.widgets.gadgets.parts and get a full list of every Part associated with @foo. I know there''s several

Good time of day all! 1) I am trying to use as5300 and asterisk. As5300 sends calls to me. I get the following in * console: -- IAX2/magrathea/6 is making progress passing it to OH323/R27464 Sep 8 10:57:59 NOTICE[1140046640]: chan_oh323.c:1159 oh323_indicate: Ignoring PROGRESS indication. As5300 user does not hear anything, just silense instead of dial tones. My config is oh323.conf

Hi folks, The obligatory newbie disclaimer: "Hi, I'm new to Asterisk and I have a couple questions..." OK, now that that's over with: I've just started working for a small CLEC, and I'm trying to sell * to my boss as a replacement for some expensive/inflexible/closed-source software he's been using to provide residential dialtone with for a couple years now.

My SIP-provider sends my a SIP-invite like this : INVITE sip:329298yyy6 at 80.XX.XX.69:5060 SIP/2.0 Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c Max-Forwards: 70 From: <sip:321445xxx6 at 80.XX.XX.69>;tag=f395877e02bf8eb2fd8f5a0e To: <sip:329298yyy6 at 80.XX.XX.69> Call-ID: f395877e02187250fd8f5a0f at 80.XX.XX.68 CSeq: 1 INVITE User-Agent: SysMaster VoIP

On Mon, Aug 10, 2020 at 3:34 AM David Chisnall <David.Chisnall at cl.cam.ac.uk> wrote: > > Thanks, > > On 07/08/2020 23:28, Kees Cook wrote: > > On Fri, Aug 7, 2020 at 1:18 AM David Chisnall > > <David.Chisnall at cl.cam.ac.uk> wrote: > >> I think it would be useful for the discussion to have a clear threat model that this intends to defend against and