search for: circut

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

...es. {clip} So, the question becomes: How do I structure my extensions.conf to convince app_parkandannounce to play it's message on the incoming channel rather than using dialback? Have I simply missed a switch somewhere, or will I have to use some sort of meetme bridge configuration to short-circut the incoming call with the announcement? {clip}

...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 pr...

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

...ircuit-busy == Everyone is busy/congested at this time -- Executing Busy("SIP/147.135.0.129-0815bc60", "") in new stack == Spawn extension (outgoing, 16037862111, 102) exited non-zero on 'SIP/147.135.0.129-0815bc60' Is this as simple as it seems? Broadvoice is circut busy? Can any one think of any other reason I might get this message? Or do I just need to call BroadVoice and complain? I have tried two different proxy's (ip's in /etc/hosts) and get the same error. in extensions.conf: [outgoing] exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@proxy....

...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...

...isk system. So, the question becomes: How do I structure my extensions.conf to convince app_parkandannounce to play it's message on the incoming channel rather than using dialback? Have I simply missed a switch somewhere, or will I have to use some sort of meetme bridge configuration to short-circut the incoming call with the announcement? I sense that this might be related at least in part to the occasional requests to the list for app_park to pass back the results in a ${PARKEDSLOT} varable. Any suggestions considered... Kris Boutilier Information Systems Coordinator Sunshine Coast Regiona...

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