search for: circuts

Hi everybody, I'm running an asterisk box -cvs version since few monthes, updated it middle of may and a last one on thursday (24 june) Since this one, my H323 calls loose they audio, both sides. Calling directly from Gatekeeper is ok, so problem comes from h323 asterisk channel. I saw few people telling about similar problem begining of month, does they solve their problem? I also grab

Couldn't see the forrest for all the fascinating tree-like applications that are out there: For future reference, see: http://www.voip-info.org/wiki-Asterisk+call+parking :-) -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from

...t dies is not always the same, some times it is server A and some times it is server B. Never have I seen that both ends die, just one. The side that is still connected can make calls to the end that died but not the other way. If you call from the server with the dead IAX2 trunk you here "All circuts are busy now." All networks have static IP addresses and their firewalls are setup to allow UDP 4569 to come in to the Asterisk systems. I have been doing a lot of research into this problem. I found this bug tracker http://bugs.digium.com/view.php?id=5912 that talks about it being an old pro...

Hi I have a system (S) that has a PSTN gateway to accept incoming calls and setup outgoing calls from/to Telco network. In the other hand I have a distant Asterisk box (A) that I would like to connect to (S) using the PRI interface. I understand that the proper way is to order to my Telco two PRI lines one for (S) and another for (A), and configure (S) and (A) to call each other numbers when

...through the pstn trunks through the upstream provider I find this problem. has anyone else ever seen this? Or seen a case where mis-matched dtmf modes across multiple providers causes this problem? minor detail on what I referred to as the 'pstn trunks' I have no analog or digital circuts all handoffs are sip. -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird Saving Lost Packets since 1994 Have you seen this packet? 1010101111010

Hello all, When I call the Broadvoice number all is good. When I try to call out through DISA on my broadvoice line i get the following: Executing Dial("SIP/147.135.0.129-0815bc60", "SIP/16037862111@proxy.bos.broadvoice.com|30") in new stack -- Called 16037862111@proxy.bos.broadvoice.com -- Got SIP response 480 "Temporarily Not Available" back from

...s is not always the same, some times it is server A and some times it is server B. Never have I seen that both ends die, just one. The side that is still connected can make calls to the end that died but not the other way. If you call from the server with the dead IAX2 trunk you here "All circuts are busy now." All networks have static IP addresses and their firewalls are setup to allow UDP 4569 to come in to the Asterisk systems. I have been doing a lot of research into this problem. I found this bug tracker http://bugs.digium.com/view.php?id=5912 that talks about it being an old...

I'm trying to use Asterisk app_parkandannouce to build a global parking pool from within a couple of Norstar PBXes. Right now I can blind transfer calls into the parking lot, but the slot announcement relies on calling back the 'transferee' after the call is parked and I can't pass enough callerid data out from within the PBX to be able to route the call back in (ie. no PRI

Hi list, I have to connect two asterisk box, in this scenario: [asterisk1]----sip----[asterisk2]----PSTN I must use sip, cos we'll use cisco rtp header-compression to save bandwidth. Could you tell me the best way to send calls from asterisk1 to asterisk2, since I cannot use IAX trunking? Thanks in advance Eduardo

asterisk-users-bounces@lists.digium.com wrote: > In the following setup: > call coming from a pstn line -> into FXO card -> asterisk -> SIP > phone > > i get an incredible loud echo in the SIP phone (about 0,5-1s) > (everything i speak into SIP phone microphone i hear in its > speaker). The person calling from PSTN is not getting any echo. Make sure you're not

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure

Hello, I''m wondering If I am missing something here when creating an example that sets an expecation at the top or beginning of an action but requires you to stub / mock everything that follows. Example: I want to test that a certain controller is running a before_filter...thats easy: - controller.should_receive(:require_user) - do_get But now i''ve got to mock / stub