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,

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

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

[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

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

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

[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

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

[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

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

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

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

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

[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() ;

[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]

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

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 =

Re: [asterisk-users] GotoIf Problem

Doug Lytle wrote: *snipped exten = _X.,n,GotoIf($[${EXTEN:1} = 9]?not-parked,s,1) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net

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,

[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

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() ;

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.

[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

[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:

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

[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

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 =

[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

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 =

[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

[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

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

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

[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

[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

[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

[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

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

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,

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

[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

[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