Re: [asterisk-users] 1.4.21.1 SIP failing, requiring reboot
Hi David, It may be IAX2 bug, do you use IAX? In my case downgrading back to 1.4.19 did the job. No IAX for me. I don't recall ever having this issue on 1.4.19 so unless I hear any other suggestions as to how to troubleshoot this I'll go back to that version as well. Best regards, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AVM Fritz BRI cards and echo cancellation
Ciao Simon, We are using 2 x AVM Fritz BRI cards with mISDN. The phones are Linksystem SPA922's and we are getting a little echo on the lines.. from what i unserstand, these are passive cards and do not have any onboard echo cancellation, but im wondering if there is anything that can be done software/config wise to help with this? I did find this (http://www.misdn.org/index.php/FAQ): 4) You set another value for tx-gain, -1 for example to prevent echoes. Please set the tx-gain back to 0 for those calls as in 3) (vt0). The point 4 you refer to is related to erroneous fax reception, I don't think it's relevant for you. Tx and Rx gains, though, are important because playing with them you can get around simple echo problems. Those settings are in misdn.conf, and are named txgain and rxgain. If you don't resolve your problem, I recommend installing Octasic SoftEcho for Asteris. I'm not affiliated with them in any way, but on my machines it works perfectly. HTH, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21.1 SIP failing, requiring reboot
David Nedved wrote: Hi David, It may be IAX2 bug, do you use IAX? In my case downgrading back to 1.4.19 did the job. No IAX for me. I don't recall ever having this issue on 1.4.19 so unless I hear any other suggestions as to how to troubleshoot this I'll go back to that version as well. Best regards, David I've just had half a day of wierd SIP stuff on 1.4.21.1 - most of it related to realtime. Sip peers showing up as UNKNOWN, but if you reboot the phone the problem goes away. For a while... PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Video on Hold
Hi, I want to put Video on Hold (3gp or other video file type )for our Asterisk, Is it possble to use music on hold configuration to set up voh ? Is there anybody can help me ? Best regards, PTCHEN ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21.1 SIP failing, requiring reboot
Paul Hales wrote: I've just had half a day of wierd SIP stuff on 1.4.21.1 - most of it related to realtime. Sip peers showing up as UNKNOWN, but if you reboot the phone the problem goes away. For a while... Interestingly enough, I've had my Grandstream suffering from the same problem since I upgraded to 1.4.20, although my config is static rather than realtime. I'd actually written it off to typical Grand-heap-of-$#!+-stream behaviour. :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA hangs up at 30 seconds
Thanks Steve, The reset worked, and now I can access the configuration panel. Can you give more details on how should I handle the 30 seconds issue? How could I manage the dial plan to answer the call? Today it works like this: Call from PSTN comes in the ATA, it picks up the call and hot dial a group number in my asterisk, this group rings several SIP extensions. I pickup a ringing softphone and start the conversation. This converstion then, hangs up at 30 seconds. The other way round: I pick up a softphone extension, have to dial the ATA number (216), it answers automatically and gives me the PSTN tone for dialing, then I have to dial the number. This call also hangs up at 30 seconds. I still have not been able to activate a one stage dialing with this ZOOM ATA. Yesterday the support from Zoom send me some instruction (attached below) on how to configure it, but I still have not been able to apply it to my asterisk server. Any instruction would be nice. Thanks, Felipe ---Instruction from Zoom Support--- Felipe, This is information on how single stage dialing works in regards to ATA and Asterisk Enable this when you enable VOIP to PSTN bridging. Enable Single Stage dialing in ATA in the Voip to PSTN Bridging. You also need to setup your asterisk to support this and these are the options. This is how single-stage dialing works: This feature works by examining the username in the From: header of a SIP INVITE. If the username is different from the username of any account on the ATA, the fxo port will go off hook and automatically dial the number in the username of the From: field. If the user has configured a security code for VoIP to PSTN dialing, the security code is included as a prefix to the number to dial. If the security code matches, the following digits are dialed out the FXO port. If the security code doesn't match, the call is shunted to the local instrument (i.e. to the FXS port). Example I: Device is registered as [EMAIL PROTECTED] INVITE arrives with From: field 2124442121 ATA comes off-hook and dials 2124442121 to the FXO port. It opens a connection between this call and the party that sent the INVITE. Example II: Device is registered as [EMAIL PROTECTED] User has configured security code of 9876 INVITE arrives with From: field 98762124442121 ATA comes off-hook and dials 2124442121 to the FXO port. It opens a connection between this call and the party that sent the INVITE. Example III: Device is registered as [EMAIL PROTECTED] User has configured security code of 9876 INVITE arrives with From: field 67892124442121 ATA connects call directly to the FXS port. Regards ZOom Tech Support Joyce Phillips ---End of instructions from Zoom Support- On Thu, Jul 17, 2008 at 6:52 PM, Steve Totaro [EMAIL PROTECTED] wrote: Generally, when you see a call always hang up at 30 seconds it is because you are not answering in your dialplan before doing other things. As for the reset, you may want to hold in the reset button for like 30 seconds, pull the power plug and plug it back in after 10 seconds while holding down the reset and keep holding it for at least another 30 seconds after you cycle the power. Thanks, Steve T On Thu, Jul 17, 2008 at 4:50 PM, Felipe Trevisan [EMAIL PROTECTED] wrote: My Zoom 5801 ATA hangs up at 30 seconds every call. I do not think it´s an Asterisk issue, as calls on the SIP trunk goes in and out normally. Below is the CLI message. 216 is the extension number assigned to the FXS extension port on the ATA. Another problem that came up while I was trying to solve the first problem, is that I´ve disabled the internal HTTP server from the ATA, and I can no longer access the configuration panel through the browser window. I´ve tried a reset puching the small button on the back, but ot simply won´t do nothing. Any clues? Thanks a lot, Felipe [Kserver*CLI == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/216-b7803460' in macro 'dial' == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/216-b7803460' -- Executing [EMAIL PROTECTED]:1] [1;36;40mMacro [0;37;40m( [1;35;40mSIP/216-b7803460 [0;37;40m, [1;35;40mhangupcall [0;37;40m) in new stack -- Executing [EMAIL PROTECTED]:1] [1;36;40mResetCDR [0;37;40m( [1;35;40mSIP/216-b7803460 [0;37;40m, [1;35;40mw [0;37;40m) in new stack -- Executing [EMAIL PROTECTED]:2] [1;36;40mNoCDR [0;37;40m( [1;35;40mSIP/216-b7803460 [0;37;40m, [1;35;40m [0;37;40m) in new stack -- Executing [EMAIL PROTECTED]:3] [1;36;40mGotoIf [0;37;40m( [1;35;40mSIP/216-b7803460 [0;37;40m, [1;35;40m1?skiprg [0;37;40m) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [EMAIL PROTECTED]:6] [1;36;40mGotoIf [0;37;40m(
[asterisk-users] DID - Panama
I need a low monthly rate DID in Panama. Maybe 2-3 inbound calls a day max. 1-2 outbound calls a day only. Will be rarely used but needs to be very good quality and needs longevity of business (eg this number is going into print so company needs to be around for a while). Please email with rates and details. Cheers, Dean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] How Register to ONE SIP provider with Multi Accounts
jiangtao wrote: I'm using asterisk 1.4.21 and a problem with sip reg server In SIP.CONF register = 07070480800:[EMAIL PROTECTED] register = 07070480801:[EMAIL PROTECTED] register = 07070480802:[EMAIL PROTECTED] register = 07070480803:[EMAIL PROTECTED] register = test1:[EMAIL PROTECTED] ... auth=07070480800:[EMAIL PROTECTED] auth=07070480801:[EMAIL PROTECTED] auth=07070480802:[EMAIL PROTECTED] auth=07070480803:[EMAIL PROTECTED] auth=test1:[EMAIL PROTECTED] and in sip show registry 211.174.52.74:5060 07070480803 360 Auth. Sent 211.174.52.74:5060 07070480802 360 Auth. Sent 211.174.52.74:5060 07070480801 360 Auth. Sent 211.174.52.74:5060 07070480800 345 Registered 117.10.72.27:5060test1 345 Registered when I register one account of 07070480800 ~07070480804 that work ok how can I seting sip.conf let 4 account regist work on ?! thanks I believe you're seeing the issue reported here: http://bugs.digium.com/view.php?id=12005 There is a workaround patch attached to that issue that you can attempt using for the time being. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX + Inidication
Hi all, I have a little problem with my IAX phones... When I pick up the headset I don't get a dialtone, and whenever I dial to a SIP phone I don't get an indication tone... Ideas? Thanks, David Vazquez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Magnetic door locks
Yes, I used a Pap2 adaptor attached door tamperproof video/speaker phone. The model I used had alarm contacts just in case it was removed from the wall you can instant trigger an alarm system. You preprogram the extension it dials and it waits to here a touch tone code that NO NC contacts are activated to do with what you please. I do not remember specifically which door phone I used but here is a link to a exhibit which list several manufacturers. http://www.archiexpo.com/cat/home-building-automation-security/access-contro l-video-and-audio-door-phones-Y-652.html Vince APCN -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of c james Sent: Thursday, July 17, 2008 8:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Magnetic door locks I have an opportunity to interface asterisk with a security system to open their magnetic door locks. The security system needs a dry contact close upon activation to signal the door. Has anyone done this before? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID - Panama
We have got that for $10 USD setup and $25 USD per month If you are interested please email me back Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Friday, July 18, 2008 9:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DID - Panama I need a low monthly rate DID in Panama. Maybe 2-3 inbound calls a day max. 1-2 outbound calls a day only. Will be rarely used but needs to be very good quality and needs longevity of business (eg this number is going into print so company needs to be around for a while). Please email with rates and details. Cheers, Dean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID - Panama
Dean asked for it so he can decide if it's worth it to him but that sounds like the price someone would pay for flatrate and probably not what one would want to pay for 5 calls per day. MARK. Sam Tam wrote: We have got that for $10 USD setup and $25 USD per month If you are interested please email me back Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Friday, July 18, 2008 9:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DID - Panama I need a low monthly rate DID in Panama. Maybe 2-3 inbound calls a day max. 1-2 outbound calls a day only. Will be rarely used but needs to be very good quality and needs longevity of business (eg this number is going into print so company needs to be around for a while). Please email with rates and details. Cheers, Dean /mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID - Panama
Hi Mark, You are correct - way overpriced. Looking for around $5-$10 a month max. I was over estimating the number of inbound and outbound calls as well. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MFH Sent: Friday, 18 July 2008 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID - Panama Dean asked for it so he can decide if it's worth it to him but that sounds like the price someone would pay for flatrate and probably not what one would want to pay for 5 calls per day. MARK. Sam Tam wrote: We have got that for $10 USD setup and $25 USD per month If you are interested please email me back Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Friday, July 18, 2008 9:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DID - Panama I need a low monthly rate DID in Panama. Maybe 2-3 inbound calls a day max. 1-2 outbound calls a day only. Will be rarely used but needs to be very good quality and needs longevity of business (eg this number is going into print so company needs to be around for a while). Please email with rates and details. Cheers, Dean /mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GotoIf Problem
Everybody,
I have a fall though context that, among other things, tests to see if
someone it trying to pick up a non-existent parked call (Defined from 90
to 99). I have the following:
[not-in-service]
exten = _X.,1,Wait(1)
exten = _X.,n,ResetCDR()
; **
; Check to see if the mis-dialed number was a parking
; slot. If so, jump to the not-parked context
; **
exten = _X.,n,GotoIf($[${EXTEN} = 90]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 91]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 92]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 93]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 94]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 95]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 96]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 97]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 98]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 99]?not-parked,s,1)
I'd like to move it to just one line, such as:
exten = _X.,n,GotoIf($[${EXTEN} = 9?]?not-parked,s,1)
But, I'm not finding a way to do this. Any suggestions?
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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[asterisk-users] Asterisk Recordings
Hello all! Is there a GUI for asterisk recordings other than ARI that comes with trixbox?. I am searching for a tool to administer call recordings. Thanks!! Cheers! Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID - Panama
No problem, you know you can always email us again if you have any other requirement in VoIP Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Saturday, July 19, 2008 12:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID - Panama Hi Mark, You are correct - way overpriced. Looking for around $5-$10 a month max. I was over estimating the number of inbound and outbound calls as well. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MFH Sent: Friday, 18 July 2008 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID - Panama Dean asked for it so he can decide if it's worth it to him but that sounds like the price someone would pay for flatrate and probably not what one would want to pay for 5 calls per day. MARK. Sam Tam wrote: We have got that for $10 USD setup and $25 USD per month If you are interested please email me back Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Friday, July 18, 2008 9:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DID - Panama I need a low monthly rate DID in Panama. Maybe 2-3 inbound calls a day max. 1-2 outbound calls a day only. Will be rarely used but needs to be very good quality and needs longevity of business (eg this number is going into print so company needs to be around for a while). Please email with rates and details. Cheers, Dean /mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIf Problem
How about:
exten = _9X,n,Goto(not-parked,s,1)
Doug Lytle wrote:
Everybody,
I have a fall though context that, among other things, tests to see if
someone it trying to pick up a non-existent parked call (Defined from 90
to 99). I have the following:
[not-in-service]
exten = _X.,1,Wait(1)
exten = _X.,n,ResetCDR()
; **
; Check to see if the mis-dialed number was a parking
; slot. If so, jump to the not-parked context
; **
exten = _X.,n,GotoIf($[${EXTEN} = 90]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 91]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 92]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 93]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 94]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 95]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 96]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 97]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 98]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 99]?not-parked,s,1)
I'd like to move it to just one line, such as:
exten = _X.,n,GotoIf($[${EXTEN} = 9?]?not-parked,s,1)
But, I'm not finding a way to do this. Any suggestions?
Doug
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide.
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Re: [asterisk-users] GotoIf Problem
Doug Lytle wrote:
*snipped
exten = _X.,n,GotoIf($[${EXTEN:1} = 9]?not-parked,s,1)
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Re: [asterisk-users] GotoIf Problem
Richard Lyman wrote:
Doug Lytle wrote:
*snipped
exten = _X.,n,GotoIf($[${EXTEN:1} = 9]?not-parked,s,1)
Close:
exten = _X.,n,GotoIf($[${EXTEN:0:1} = 9]?not-parked,s,1)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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[asterisk-users] automon=*, Dial(, , Ww), rfc2833, canreinvite=no, but...
After much checking and puzzling, I cannot get my Polycom 601 to toggle call recording with my Asterisk 1.4.21.1. Via FreePBX, I can set a user to always record, and the recording will show up in /var/spool/asterisk/monitor. But if I try to start recording by toggling in-call, no luck. I can see this in the feature*.conf file set: automon=*1 and I can see a 'Ww' in the logged/traced call to dial(). and I can see the RFC2833 RTP packets going through Asterisk, both with rtp debug and with wireshark. So my questions are: 1) How do I verify that asterisk actually saw the feature code spec upon restart/reload? I can't find any clues. 2) Are there any other parameters that have a bearing on this? 3) Is there anything I haven't thought of? Finally, it might be worth noting that the packet traces show three RFC2833 end events for each DTMF code pressed. This might be perfectly normal, and I even tried fudging the automon string to ***111 just to compensate as an experiment, but it had no effect. If I've done everything necessary to configure enabling the toggle function, then where should I see the failure/refusal to comply in any logs. I'm getting nothing in logs/traces. A side question: freepbx is generating include statements with a leading #, a la C includes - or a la Perl/Shell/et al comments! This is OK? I've floundering with the suspicion that I'm overlooking something really dumb... I would be grateful for some explicit diagnostic suggestions. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIf Problem
Doug Lytle wrote:
Everybody,
I have a fall though context that, among other things, tests to see if
someone it trying to pick up a non-existent parked call (Defined from 90
to 99). I have the following:
[not-in-service]
exten = _X.,1,Wait(1)
exten = _X.,n,ResetCDR()
; **
; Check to see if the mis-dialed number was a parking
; slot. If so, jump to the not-parked context
; **
exten = _X.,n,GotoIf($[${EXTEN} = 90]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 91]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 92]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 93]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 94]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 95]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 96]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 97]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 98]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 99]?not-parked,s,1)
I'd like to move it to just one line, such as:
exten = _X.,n,GotoIf($[${EXTEN} = 9?]?not-parked,s,1)
But, I'm not finding a way to do this. Any suggestions?
Doug
How about something like this:
exten = _X.,n,GotoIf($[${EXTEN:0:1} = 9]?not-parked,s,1)
You may need to tweak the extension pattern as this will match anything
that begins with 9.
-Dave
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Re: [asterisk-users] GotoIf Problem
Eric ManxPower Wieling wrote: How about: exten = _9X,n,Goto(not-parked,s,1) This works quite well, thank you! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] automon followup
A followup to my own inquiry... pig*CLI feature show Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# # Attended Transfer One Touch Monitor Disconnect Call * * Park Call Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 70 Parking context : parkedcalls Parked call extensions: 71-79 I guess this narrows it down. So presumably, my feature code specs are not finding their way into the process, but why? I'm looking, but comments are most welcome. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Colorado Asterisk User Group Forming
If you have interest in participating in a newly forming Colorado Asterisk User Group, please contact Paul Gregory at BlueModus in Denver.Monthly meetings may begin as early as the Fall 2008 and will be in the metro Denver area. Contact information as follows: Paul Gregory BlueModus t: (303) 759 2100 x319 d:(303) 951 0319 e: [EMAIL PROTECTED] w: www.bluemodus.comwww.Bluemodus.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automon followup
Bill Michaelson wrote: A followup to my own inquiry... pig*CLI feature show Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# # Attended Transfer One Touch Monitor Disconnect Call * * Park Call Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 70 Parking context : parkedcalls Parked call extensions: 71-79 I guess this narrows it down. So presumably, my feature code specs are not finding their way into the process, but why? I'm looking, but comments are most welcome. There is a note in the features.conf sample which may answer this: ; Note that the DYNAMIC_FEATURES channel variable must be set to use the features ; defined here. The value of DYNAMIC_FEATURES should be the names of the features ; to allow the channel to use separated by '#'. For example: ; ;Set(DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3) Did you remember to set the DYNAMIC_FEATURES variable? Something like: exten = blah,n,Set(DYNAMIC_FEATURES=automon) Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] automon follup #2
OK, I had broken the feature.conf fileset, but I just fixed it. Now I can confirm: pig*CLI feature show Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * ** Park Call Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 70 Parking context : parkedcalls Parked call extensions: 71-79 but, still no evidence of recording upon sending *1 through box. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitForSilence Problems
Actually, I thought about it for a while. What I want is something that
will allow me to restart the message if another sound is detected.
Something like this:
exten = answermachine,1,Answer()
exten = answermachine,n,WaitForSilence(1000,2)
exten = answermachine,n,Background(message)
exten = answermachine,n,GotoIf($[${BACKGROUND}=DETECTED]?replay:exit)
exten = answermachine,n(replay),Playback(message)
exten = answermachine,n(exit),Hangup()
But Background() is looking for a DTMF tone and doesn't even work the way I
described up there. Is there a function that looks for any significant
sound (ie, a BP) that will return and not continue the audio?
On Thu, Jul 17, 2008 at 1:43 PM, Julian Lyndon-Smith [EMAIL PROTECTED]
wrote:
This is what we use, with (seemingly) good success:
exten = answermachine,1,Answer
exten = answermachine,n,Wait(5)
exten = answermachine,n,WaitForSilence(1000,2)
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[asterisk-users] Beep on transfer
Hello All, I have a request that I have not been able to figure out as yet. I need to be able to play a beep when a call is transfered via attended transfer. This is exactly what is in the bug tracker at: http://bugs.digium.com/view.php?id=3819 Has any one found a way, elegant ot otherwise, to make something such as this work? Thanks in advance for any help. -- JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitForSilence Problems
Nicholas Blasgen wrote:
Actually, I thought about it for a while. What I want is something
that will allow me to restart the message if another sound is
detected. Something like this:
exten = answermachine,1,Answer()
exten = answermachine,n,WaitForSilence(1000,2)
exten = answermachine,n,Background(message)
exten = answermachine,n,GotoIf($[${BACKGROUND}=DETECTED]?replay:exit)
exten = answermachine,n(replay),Playback(message)
exten = answermachine,n(exit),Hangup()
But Background() is looking for a DTMF tone and doesn't even work the
way I described up there. Is there a function that looks for any
significant sound (ie, a BP) that will return and not continue the
audio?
Maybe you wanted BackgroundDetect application.
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[asterisk-users] 1.6b9 Audio Issue
I'm noticing in 1.6 Beta 9 that on outgoing calls I get a brief audio drop when the audio starts on the other end of the call. So basically I hear the first word, miss the second word and then hear the rest fine. I've noticed this after calling multiple locations and getting some recording on the other end. The origin of the outbound channel is always SIP but the asterisk to PSTN could be SIP or IAX. Anyone else? MARK. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TOS and security
I'm preparing for a client install of * by doing a fresh one in-house. Unlike my earlier installation that runs asterisk as superuser, my current experimental box runs without such privilege. This is causing it to moan that it can't set TOS. I absolutely don't want to install it on the client LAN without this capability. If need be, I'll set the binary to run setuid root. But I'm looking for something more elegant. While googling, I found a suggestion to use iptables mangle rules to set TOS for all packets going out of the box on ports like 5060 and 1:2. Not a bad hack, but indiscriminate and this box will be handling other traffic besides the RTP. I'd like to do better. I thought of using POSIX access control to enable asterisk to do TOS setting without being root (would this be CAP_NET_RAW?), which sounds perfect, but so far I'm operating with stock ubuntu hardy, and I would like to avoid a kernel build to add this capability. Any other ideas? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6b9 Audio Issue
MFH wrote: I'm noticing in 1.6 Beta 9 that on outgoing calls I get a brief audio drop when the audio starts on the other end of the call. So basically I hear the first word, miss the second word and then hear the rest fine. I've noticed this after calling multiple locations and getting some recording on the other end. The origin of the outbound channel is always SIP but the asterisk to PSTN could be SIP or IAX. Anyone else? MARK. One difference between Asterisk 1.6.0 and previous versions is that when a channel answers, there is a built-in 500 ms delay so that media has time to be set up. This may be what you are experiencing. There was a bug reported recently that was traced back to this delay. In the next 1.6.0 tarball, the delay will behave slightly differently, although I doubt it will be noticeable for the situation you have described. The bug I refer to is: http://bugs.digium.com/view.php?id=12924 Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TOS and security
On Friday 18 July 2008 14:21:14 Bill Michaelson wrote: I'm preparing for a client install of * by doing a fresh one in-house. Unlike my earlier installation that runs asterisk as superuser, my current experimental box runs without such privilege. This is causing it to moan that it can't set TOS. I absolutely don't want to install it on the client LAN without this capability. If need be, I'll set the binary to run setuid root. But I'm looking for something more elegant. While googling, I found a suggestion to use iptables mangle rules to set TOS for all packets going out of the box on ports like 5060 and 1:2. Not a bad hack, but indiscriminate and this box will be handling other traffic besides the RTP. I'd like to do better. I thought of using POSIX access control to enable asterisk to do TOS setting without being root (would this be CAP_NET_RAW?), which sounds perfect, but so far I'm operating with stock ubuntu hardy, and I would like to avoid a kernel build to add this capability. It's actually CAP_NET_ADMIN, and we already keep that privilege, if the configure script detects that the capabilities library is available. Simply set the runuser and rungroup in asterisk.conf, and Asterisk will automatically keep those privileges during startup. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Announcing AstriDevCon 2008!
On September 26-28 in Glendale, Arizona, a group of Asterisk developers will be getting together for three days of hacking, coding, testing, designing and otherwise beating on the Asterisk code base. The event will be hosted at the Renaissance Glendale Hotel and Spa immediately following AstriCon 2008 and will be low-key and open only to serious developers and contributors. We are expecting to keep the attendance to 50 people or less, including many members of the Digium Asterisk development team (currently around 15 people). If you wish to participate, please contact Kevin P. Fleming so he can make arrangements with you. We will need to have the final list of attendees in place by August 15th or so, so that hotel accommodations can be confirmed. You can find accommodation and travel information on the AstriCon website at http://www.astricon.net. Each attendee will be responsible for their own travel, meals and lodging costs; the conference sessions will only have a beverage bar and light snacks. There will be free wireless Internet access in the meeting room and in the guest rooms at the Renaissance. This year we plan to focus our efforts on media stream handling and codec (format) negotiations; at the previous two DevCons we have talked about these topics but not made any significant progress, and it's time to get the work done to improve Asterisk so it can do a better job handling complex media streams and changing codec requirements. If you are interested in attending, send an email application to [EMAIL PROTECTED] including your name, your involvement with Asterisk (or related projects), and who is sponsoring your attendance (if any company or person is doing so). We will accept applications until August 15th, and then make the decisions about who we can accept based on their level of contribution and the space available at the event. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.21.1
Hello, I just upgraded my asterisk to Asterisk 1.4.21.1 I am getting this Notice can any one tell me what i need to see in order to fix this problem. [Jul 18 18:27:08] NOTICE[9779]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '0.0.0.0' [Jul 18 18:27:09] NOTICE[9780]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '0.0.0.0' -- With Best Regards, Faisal A. Ashraf Web www.voip.com.pk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk not converting DTMF from INFO to rfc2833
Hi, I have asterisk 1.4.20 bridging two SIP channels with different DTMF mode set on both. So when one SIP end points send INFO dtmf on channel 1, asterisk is not able to generate rfc2833 dtmf events on the channel 2 bridged to channel 1. The channel 2 dtmfmode is set to rfc2833. I also tried using the SIPdtmfMode() in the dial plan, however even that does not work. Has anyone faced this issue? Regards, Mayur ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] going from 1.4.21 to 1.6 beta 9
1.4 was working fine. I thought I would try 1.6 beta 9 from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept the call. [Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite: Call from 'JJ' to extension 'jj_audio' rejected because extension not found. I changed nothing in the config files. I tried setting debug level to 5 and verbose to 5 all I still get is the one liner above. Has something changed in 1.6 that affects incoming calls (that I have not found) my sip.conf still has the context set to the correct value (as 1.4 did), my extensions.conf still has that context. Thanks for any pointers. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] going from 1.4.21 to 1.6 beta 9
Jerry Geis wrote: 1.4 was working fine. I thought I would try 1.6 beta 9 from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept the call. [Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite: Call from 'JJ' to extension 'mediaport_audio_visual' rejected because extension not found. I changed nothing in the config files. I tried setting debug level to 5 and verbose to 5 all I still get is the one liner above. Has something changed in 1.6 that affects incoming calls (that I have not found) my sip.conf still has the context set to the correct value (as 1.4 did), my extensions.conf still has that context. Thanks for any pointers. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.8:16642 Looking for mediaport_audio_visual in smvoice-mediaport (domain 192.168.1.25) --- Reliably Transmitting (no NAT) to 192.168.1.8:5060 --- SIP/2.0 404 Not Found Jerry I found more information: my sip.conf has: context=smvoice-mediaport The sip debug shows: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.8:16642 Looking for mediaport_audio_visual in smvoice-mediaport (domain 192.168.1.25) --- Reliably Transmitting (no NAT) to 192.168.1.8:5060 --- SIP/2.0 404 Not Found my extensions.conf section: [smvoice-mediaport] exten = public_address,1,Goto(smvoice-mediaport-public-address,s,1) #include /etc/asterisk/express.dnis.conf then express.dnis.conf has: exten = mediaport_audio_visual,1,Goto(smvoice-mediaport-audio-visual,s,1) So its all there, it works in 1.4 but not in 1.6 b9 What gives? Thanks. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip extension compromised, need help blocking brute force attempts
If you are trying to reject an IP address to connect to asterisk, there is no need to run iptables. Each SIP definition in sip.conf can have: deny=0.0.0.0/0.0.0.0 permit=192.168.135.1/255.255.255.0 just set these values and it wont accept anything from that IP. On Mon, Jul 7, 2008 at 7:37 PM, Dovid B [EMAIL PROTECTED] wrote: - Original Message - From: spectro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 01, 2008 8:02 PM Subject: Re: [asterisk-users] sip extension compromised,need help blocking brute force attempts On Tue, Jul 1, 2008 at 11:19 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Fix your logger.conf, then. -- Tzafrir Cohen What am I missing? [EMAIL PROTECTED] ~]# cat /etc/asterisk/logger.conf ; ; Logging Configuration ; ; In this file, you configure logging to files or to ; the syslog system. ; ; For each file, specify what to log. ; ; For console logging, you set options at start of ; Asterisk with -v for verbose and -d for debug ; See 'asterisk -h' for more information. ; ; Directory for log files is configures in asterisk.conf ; option astlogdir ; [logfiles] ; ; Format is filename and then levels of debugging to be included: ;debug ;notice ;warning ;error ;verbose ; ; Special filename console represents the system console ; ;debug = debug ;console = notice,warning,error ;console = notice,warning,error,debug ;messages = notice,warning,error full = notice,warning,error,debug,verbose ;syslog keyword : This special keyword logs to syslog facility ; ;syslog.local0 = notice,warning,error ; [EMAIL PROTECTED] ~]# The script seems to run off the messages log. Uncomment the messages line and the reload the logger in asterisk (logger reload from the CLI). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TOS and security
I'm preparing for a client install of * by doing a fresh one in-house. Unlike my earlier installation that runs asterisk as superuser, my current experimental box runs without such privilege. This is causing it to moan that it can't set TOS. I absolutely don't want to install it on the client LAN without this capability. If need be, I'll set the binary to run setuid root. But I'm looking for something more elegant. While googling, I found a suggestion to use iptables mangle rules to set TOS for all packets going out of the box on ports like 5060 and 1:2. Not a bad hack, but indiscriminate and this box will be handling other traffic besides the RTP. I'd like to do better. It is possible for an iptables filter/rule to match packets in the OUTPUT chain based on the UID or GID of the process which created them, if you have the owner module loaded. You should be able to add a rule to the OUTPUT chain of the mangle table which will set the TOS properly for any and all outbound packets generated locally by the non-root user ID which you're using to run Asterisk. Come to think of it, I think I need to do this myself. I'm using the ultimate Linux traffic conditioning configuration (modified very slightly) to prioritize my system's outbound traffic into multiple queues by TOS, and it's probably mis-queueing the RTP traffic because my Debian install of Asterisk is running under a non-root UID. I thought of using POSIX access control to enable asterisk to do TOS setting without being root (would this be CAP_NET_RAW?), which sounds perfect, but so far I'm operating with stock ubuntu hardy, and I would like to avoid a kernel build to add this capability. Any other ideas? Seems like iptables -t mangle -A OUTPUT -m owner --uid-owner $ASTERISK would be along the lines of what you want? Mark the packets with the TOS you want... and then consider using the Linux traffic-shaping system to make sure that they really do get transmitted ahead of non-urgent packets: http://tldp.org/HOWTO/Adv-Routing-HOWTO/lartc.cookbook.ultimate-tc.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT Astricon/Digium Beach Ball Mailing
Just an FYI for Digium. I received a mailing today from you guys which was nice. The price of mailing was ~$1.60 and inside was an inflatable beach ball. Cool, but I tried to blow up the beach ball and the the seam where the part opens to inflate the ball was not connected to the ball whatsoever, so it went right in the trash. I wonder if the sick heat had anything to do with it, was mine just bad, or should Digium get a refund from the promotion company for providing garbage? Anyone else get one? Was it OK or junk? I post this not to put down Digium, the thought was nice, I wish I could play with my Digium beach ball, but Digium should know about it if it was common. Postage alone was costly. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changinf Polycom-501 config server from remote?
Hello, Our Polycom-501 phones are set to retreive their config for the server by a static configuation defined at the phones (boot servers). Is there any way to change it remotely? I found no relevant field in the internal WEB browser, nor anything in the configuration files (sip.conf and phone1.conf). Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
