search for: dickenson

...1 07:48:43.897] == Using SIP RTP TOS bits 184 [2009-07-21 07:48:43.897] == Using SIP RTP CoS mark 5 [2009-07-21 07:48:43.897] == Using SIP VRTP TOS bits 136 [2009-07-21 07:48:43.897] == Using SIP VRTP CoS mark 6 [2009-07-21 07:48:44.004] -- Executing [*9901 at empl:1] Playback("SIP/dickenson-174c2010", "vm-goodbye") in new stack [2009-07-21 07:48:44.120] -- <SIP/dickenson-174c2010> Playing 'vm-goodbye.gsm' (language 'en') [2009-07-21 07:48:45.140] -- Executing [*9901 at empl:2] Answer("SIP/dickenson-174c2010", "") in new...

...hed call using AMI? I have an established call from which I can record either or both legs. I can additionally "spy" on the call. Is there any way I can play a sound file into the call and not loose the ability for the people to continue talking while listening to the sound file? -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/

...f DID numbers, and I can check those and send to appropriate extensions. Thanks all for helping to clarify. I have gotten a couple of users who haven't been able to call out, and wasn't sure if I wasn't "rolling over" the sip lines properly. Best, Eddie Mikell From: Jim Dickenson<dickenson at cfmc.com> Subject: Re: [asterisk-users] Multiple SIP lines. To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID:<EDA8102C-B255-46E0-940D-1EF217566DDF at cfmc.com> Content-Type: text/plain; charset=us-ascii I th...

In one of my center , its not taking root password. Anyways to recover it ? In other terms , I lost the control of server. Any solution or re-installation is the only way left ? I am using CentOS. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090122/ef95ad6e/attachment.htm

.../dahdi/xpp/init_fxo_modes.verified Building modules, stage 2. MODPOST WARNING: could not find /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32. o.cmd for /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o Anyone else seeing this? -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/

...is played can either a PRI call or a SIP call know this without analyzing the audio stream? Are there reasons to prefer the use of PRI over SIP or SIP over PRI? I would like people's opinions as to if one form is better than the other in any meaningful way. Thanks for you feed-back. -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/

...cause this problem to occur after 15 seconds but that did not change the behavior. I am not sure if there is a setting I can set to increase the 30 second timeout that is occurring to some higher value. Does anyone know why the PRI line does not want to ring for more than 30 seconds? TIA -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/

...le to do is allow the phones on desks connected to this server the ability to dial something that would allow the phone to access an asterisk box to be able to do an agent login over their LAN. Is there any way to do this? Can the Lync server have a SIP trunk to connect to an Asterisk box? -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/

Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : alice at pbx.com should be able to call bob at

Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf, but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA -> High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems

...ched id -1??? There are no messages in the full log file before these line since 21:43 on 6/16/2009. I am running asterisk 1.6.0.9, dahdi-linux-2.1.0.4, dahdi-tools-2.1.0.2, libpri-1.4.10 and wanpipe-3.5.2. The PRI line is plugged in to a Sangoma A102de. Any hints would be appreciated. -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/

...0000:nottelling I have searched various web sites and mail lists but I can not find very much documentation about how the updateconfig action is to work. Can anyone point me to additional documentation in addition to offering some ideas as to why the above transaction might not work. TIA -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n

Hello, On an Asterisk 1.4.33.1 in a simple scenario: [test] exten => _X.,1,Dial(SIP/12345 at peer01,,,) exten => i,1,Hangup(${HANGUPCAUSE}) exten => t,1,Hangup(${HANGUPCAUSE}) exten => h,1,Hangup(${HANGUPCAUSE}) I have noticed that no matter what value we set in the Hangup(<cause code>) commands, if the call is not answered by peer01 for any reason, the actual cause code

if i run let's say 1 pbx running on my main linux box and a another on my windows box if a person dial my main number and press lets say 1 are it possible to transfer the call over to my other pbx hope anyone understand -------------- next part -------------- An HTML attachment was scrubbed... URL:

...|| ast_check_hangup(chan)) New: if (res == -1 || ast_check_hangup(chan) || !peer_chanspy_ds) Otherwise, as best I can tell, unless there is some error chanspy never exits unless the channel running the chanspy application hangs up, which I do not particularly want to do. -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/

Hi all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/c193b48d/attachment.html>

Hello. I'm new with asterisk. Can you help me in this: I have cisco sip phone (601) connected to asterisk server, and 1 client number (500). I want to dial from 601 to 500. But get error in cli console: [Apr 27 15:30:15] NOTICE[9650]: chan_sip.c:20059 handle_request_invite: Call from '601' to extension '500' rejected because extension not found. What's wrong?

...xpected the call to get a busy as I have busylevel set to one. I have tried setting busylevel to two as well with the same result. Can someone let me know what I should look at to see why I am not getting a busy instead of ringing? Here are the definitions in sip.conf for the GXP280 and zoiper: [dickenson] type=friend context=empl nat=yes host=dynamic secret=<password> callerid=Jim Dickenson <108> mailbox=108 at ourvm [GXP280] type=friend context=empl host=dynamic secret=<password> callerid=GXP280 <109> mailbox=109 at ourvm busylevel=1 TIA -- Jim Dickenson mailto:dickenso...

...level > set to one. I have tried setting busylevel to two as well with the same > result. > > Can someone let me know what I should look at to see why I am not getting a > busy instead of ringing? > > Here are the definitions in sip.conf for the GXP280 and zoiper: > > [dickenson] > type=friend > context=empl > nat=yes > host=dynamic > secret=<password> > callerid=Jim Dickenson <108> > mailbox=108 at ourvm > > [GXP280] > type=friend > context=empl > host=dynamic > secret=<password> > callerid=GXP280 <109> &g...