Re: [asterisk-users] Asterisk sip attack
bayardo.sanc...@gmail.com wrote: This week I was experiencing attacks sip log into my accounts were more than 1,000 requests for records Sip accounts in less than an hour THROUGH deny the ip of my router access list in cisco and my asterisk server to go through the iptables drop ip attacker is a way for an account with another ip can not log into my asterisk server to add some command in my sip.conf for deny register account sip in my asterisk? -- Sent from my BlackBerry® VoIP, Windows/Linux Administration and Network Management US Numbers: 561-886-0664 Nicaragua Mobile: +505.8488.6876 Have a look at fail2ban. Otherwise please add some punctuation to your original question because at the moment I cant tell what you are actually asking for. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not able to join conference
What happens if you put in a 'room' number? Eg: exten = 8080,3,MeetMe(500|MDci) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: 17 September 2010 14:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Not able to join conference Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the conference and meet with the people already in, here is the lines corresponding to conf in the dialplan, that would be a big help if you guys can help diagnose the issue. exten = 8080,1,Answer exten = 8080,2,Wait,1 exten = 8080,3,MeetMe(|MDci) If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Confused about notifyringing in sip.conf
Hello list, I read this in sip.conf : notifyringing = no ; Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: yes) What does this mean ?! Does this mean that when I mark this as yes, a phone that already has taken a call will be send a second and third call ?! I want that if a phone is in use (calling), the phone does not ring on a second call. Is it possible that by setting this as no, my phone's BLF-lamps are not blinking when a phone is ringing ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confused about notifyringing in sip.conf
Hi! notifyringing = no ; Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: yes) Does this mean that when I mark this as yes, a phone that already has taken a call will be send a second and third call ?! No, not directly: This setting is only about SIP subscription information spread via NOTIFY messages, and thus only concerns the hint (BLF) feature of Asterisk. I want that if a phone is in use (calling), the phone does not ring on a second call. The term you are looking for is call waiting; use GROUP() or call-limit= to control the number of calls for a specific phone (or employ DEVICE_STATE()). Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension continues ringing after caller hanged up
Hi, I use asterisk with sip3000 device with sip-aho connected to PSTN and sip-ahi connected to a phone. When call arrives from PSTN, the *phone continues ringing even after caller hanged up*. The dialplan contains the following lines: [from-pstn] ... exten = 99,n,Dial(SIP/sip-ahi,30,g) exten = 99,n,Hangup() The asterisk properly detects hangup of the caller as I see following lines in asterisk -crvv ... Dial(SIP/sip-aho-0003, SIP/sip-ahi,30,g) == Spawn extension (from-pstn, 99, 8) exited non-zero on 'SIP/sip-aho-0003' ... How can I make the phone stop ringing the moment caller hangup? -- Arie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confused about notifyringing in sip.conf
Hi! notifyringing = no ; Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: yes) Does this mean that when I mark this as yes, a phone that already has taken a call will be send a second and third call ?! No, not directly: This setting is only about SIP subscription information spread via NOTIFY messages, and thus only concerns the hint (BLF) feature of Asterisk. I want that if a phone is in use (calling), the phone does not ring on a second call. The term you are looking for is call waiting; use GROUP() or call-limit= to control the number of calls for a specific phone (or employ DEVICE_STATE()). Philipp -- Could you explain this in more detail? I have to use call-limit=2 to enable attended transfers which causes a second call to ring. But I don't want a second call to ring, I want to get a busy state instead. How can I achieve this with GROUP() or DEVICE_STATE()? Thanks, Oliver -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again..
On 20 September 2010 05:33, dashy dude dashy_v2...@yahoo.com wrote: Hi, I tried disabling cdr_addon_mysql.so. Still error comes let's say once a day or so. Is there anything else I can do about? rgds --- On Thu, 9/9/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: From: Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de Subject: Re: [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again.. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, September 9, 2010, 2:17 AM Hi! I am running asterisk ver 1.2.4 and have faced this error: Try a downgrade to Asterisk 0.7.1 ;- Philipp Do you honestly think that you are going to get support here for this version of Asterisk? Upgrade to something from the last year or so.. 1.4.xx branch is very stable. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playing Audio To One Channel
Hi I have a call established and I want to play audio to just one channel on that call. Is this possible? If so, how? My google-fu has failed on this one. Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not able to join conference
it's going to put you in conf no 500 without prompting you to enter a conference number I guess, but i don't it's going to solve my issue. actually I'm atill wondering is there a way to debug just Meetme app output or the only way is turn the whole debug thing on? On Mon, Sep 20, 2010 at 4:11 AM, Andrew Thomas a...@datavox.co.uk wrote: What happens if you put in a 'room' number? Eg: exten = 8080,3,MeetMe(500|MDci) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: 17 September 2010 14:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Not able to join conference Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the conference and meet with the people already in, here is the lines corresponding to conf in the dialplan, that would be a big help if you guys can help diagnose the issue. exten = 8080,1,Answer exten = 8080,2,Wait,1 exten = 8080,3,MeetMe(|MDci) If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Abdullah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing Audio To One Channel
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Farmer Sent: Monday, September 20, 2010 7:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Playing Audio To One Channel Hi I have a call established and I want to play audio to just one channel on that call. Is this possible? If so, how? My google-fu has failed on this one. Regards Jon -- Jon Farmer Tel 07795 118140 One option would be to play your audio through a conference; Asterisk seems to have great controls over legs using that infrastructure. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing Audio To One Channel
One way to do it is to use ChanSpy and the whisper option. We use AMI to play
sound bits to one leg of the call.
Something like
Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable: CfMC_WhatToPlay=lyrics-louie-louie
Variable: CfMC_WhoHear=SIP/GXP280_18-0002
ActionID: PlayBack
Async: true
exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_playback,n,Wait(0.3)
exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}
${PLAYBACKSTATUS})
exten = do_playback,n,Hangup()
exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 20, 2010, at 5:51 AM, Jon Farmer wrote:
Hi
I have a call established and I want to play audio to just one channel
on that call. Is this possible? If so, how? My google-fu has failed on
this one.
Regards
Jon
--
Jon Farmer
Tel 07795 118140
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Re: [asterisk-users] Playing Audio To One Channel
On 20 September 2010 14:23, Danny Nicholas da...@debsinc.com wrote: One option would be to play your audio through a conference; Asterisk seems to have great controls over legs using that infrastructure. That is not an option. I am using Asterisk as a media relay and want to play a message to the subscriber when call credit is low. However I don't want the other party to hear the message. Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension continues ringing after caller hanged up
Have you tried removing option 'g' from your Dial command? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-20 7:45 AM, Arie Skliarouk sklia...@gmail.com wrote: Hi, I use asterisk with sip3000 device with sip-aho connected to PSTN and sip-ahi connected to a phone. When call arrives from PSTN, the *phone continues ringing even after caller hanged up*. The dialplan contains the following lines: [from-pstn] ... exten = 99,n,Dial(SIP/sip-ahi,30,g) exten = 99,n,Hangup() The asterisk properly detects hangup of the caller as I see following lines in asterisk -crvv ... Dial(SIP/sip-aho-0003, SIP/sip-ahi,30,g) == Spawn extension (from-pstn, 99, 8) exited non-zero on 'SIP/sip-aho-0003' ... How can I make the phone stop ringing the moment caller hangup? -- Arie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing Audio To One Channel
On 20 September 2010 14:21, Jim Dickenson dicken...@cfmc.com wrote: One way to do it is to use ChanSpy and the whisper option. We use AMI to play sound bits to one leg of the call. Something like Hi I have tried your suggestion however I can't get it to work. When I send the originate via the manager interface the extensions get fired and doing a show channels shows the chanspy and playbacks working but I hear nothing. Any ideas? Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension continues ringing after caller hanged up
Hi, On Mon, Sep 20, 2010 at 16:39, Zeeshan Zakaria zisha...@gmail.com wrote: Have you tried removing option 'g' from your Dial command? Of course, with the same result. -- Arie Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-20 7:45 AM, Arie Skliarouk sklia...@gmail.com wrote: Hi, I use asterisk with sip3000 device with sip-aho connected to PSTN and sip-ahi connected to a phone. When call arrives from PSTN, the *phone continues ringing even after caller hanged up*. The dialplan contains the following lines: [from-pstn] ... exten = 99,n,Dial(SIP/sip-ahi,30,g) exten = 99,n,Hangup() The asterisk properly detects hangup of the caller as I see following lines in asterisk -crvv ... Dial(SIP/sip-aho-0003, SIP/sip-ahi,30,g) == Spawn extension (from-pstn, 99, 8) exited non-zero on 'SIP/sip-aho-0003' ... How can I make the phone stop ringing the moment caller hangup? -- Arie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug with Realtime?
Check the SIP debug and see what is going on. Leif. Hi, I checked the SIP debug. As soon as I issue the RELOAD command, no SIP data gets transferred to the phone. Asterisk output: http://pastebin.com/FB675N16 Any ideas how I can do a SIP reload without losing the Sip Phones registration? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?
Anyone have a AudioCodes with Asterisk ??? 2010/9/18 Olivier CALVANO o.calv...@gmail.com: Hi i have buy a Audiocode Median 2000 VoIP Gateway and connect it on : 1 E1 30 channels 1 Lan Port Anyone use this equipements with asterisk ? because i am search a config sample for AudioCode and for Asterisk (i am new in VoIP). I want that all calls arrives on the AudioCode are sent to the asterisk by SIP (trunk ?) and all outgoing call from Asterisk are sent to the AudioCode. I don't want specify numbers on the audiocode, a +33* = Asterisk. Thanks for your help Olivier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension continues ringing after caller hanged up
Do you mean spa3000 or sip3000? I remember having same problem with spa3000 and the problem was somewhere in the settings of spa3000 that wouldn't stop ringing the phone. I don't remember the details at this moment as it was long time ago, but this much I can tell that it is a config issue with spa3000 device, not asterisk. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-20 11:02 AM, Arie Skliarouk sklia...@gmail.com wrote: Hi, On Mon, Sep 20, 2010 at 16:39, Zeeshan Zakaria zisha...@gmail.com wrote: Have you tried removi... Of course, with the same result. -- Arie Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-20 7:45 AM, Arie Skliarouk s... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug with Realtime?
I am not aware of any way to do that. My question is if you are using realtime, why are you doing a sip reload? If you change the settings on a device in the realtime DB, just prune it and it will grab the new config the next time they re-register. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Monday, September 20, 2010 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bug with Realtime? Check the SIP debug and see what is going on. Leif. Hi, I checked the SIP debug. As soon as I issue the RELOAD command, no SIP data gets transferred to the phone. Asterisk output: http://pastebin.com/FB675N16 Any ideas how I can do a SIP reload without losing the Sip Phones registration? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting 'fname_base' variable doesn't affect 'automon' result file.
Hello List,
Maybe I'm mistaken, but, shouldn't the 'fname_base' variable of
'Monitor' application affect the file name generated through 'automon'
feature?
I initialized this variable with a value as follows:
Set(fname_base=auto-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
a. Should I use 'fname_base' in uppercase (FNAME_BASE)? or...
b. Is this variable independent of the 'automon' feature?
Thanks in advice,
PS.
version: Asterisk 1.4.33.1
OS: Slackware Linux 13.0
--
Jose P. Espinal
http://www.eSlackware.com
IRC: Khratos @ #asterisk / -doc / -bugs
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Re: [asterisk-users] Bug with Realtime?
Can we not do pastebin any more? I just received this:- [PASTEBIN URL REMOVED] has been detected as suspicious URLs,and Quarantine to user's spam folder has been taken on 9/20/2010 8:24:38 AM. Message details: Server: MADRID Sender: d...@keshrcommunications.com; Recipient: asterisk-users@lists.digium.com; Subject: Suspicious URL:Re: [asterisk-users] Bug with Realtime? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?
On Mon, Sep 20, 2010 at 11:48 AM, Olivier CALVANO o.calv...@gmail.com wrote: Anyone have a AudioCodes with Asterisk ??? Yes, but why? Both do the same thing. It would be like me asking 'I have a bike and need to get to work. Can I use the bike with a car?' -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension continues ringing after caller hanged up
On Mon, Sep 20, 2010 at 7:31 AM, Arie Skliarouk sklia...@gmail.com wrote: When call arrives from PSTN, the phone continues ringing even after caller hanged up. I suspect a bug [1] but without a SIP debug, I cannot be sure. [1] https://reviewboard.asterisk.org/r/870/ -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk stops processing SIP UDP messages
Last week I had a couple of outages one machine, the problem was that Asterisk suddly stopped responding to UDP SIP requests. tcpdump show requests arriving on the machine, sip debug log in asterisk doesn't show anything for the UDP peers, TCP functions just fine. In all 3 cases the log is something like below, a +/- 10m gap in any SIP/UDP related traffic in the logs, followed by a bunch of Really destroying SIP dialog messages. Versions of Asterisk affected where 1.6.2.9 and 1.6.2.13 on a Debian/stable machine. There is no NAT involved and reloading/flushing iptables has no effect, one of the first rules is to accept both incoming tcp/udp traffic: Chain INPUT (policy DROP) fail2ban-asterisk-tcp tcp -- 0.0.0.0/0 0.0.0.0/0 multiport dports 5060,5061 fail2ban-asterisk-udp udp -- 0.0.0.0/0 0.0.0.0/0 multiport dports 5060,5061 fail2ban-ssh tcp -- 0.0.0.0/0 0.0.0.0/0 multiport dports 22 ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpts:1:2 ACCEPT tcp -- 0.0.0.0/0 0.0.0.0/0 tcp dpt:5060 ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:5060 ACCEPT all -- 0.0.0.0/0 0.0.0.0/0 state ESTABLISHED Sadly enough I'm not sure wether the RTP streams also stop functioning (forgot to capture all traffic), but since a TCP peer tried to call us (also a TCP peer) and failed to do so my guess is no UDP is working at all. I don't have enough information to make a decent bugreport so my question is if anyone experienced something like this or how to accumilate further information for a better bugreport? == [2010-09-18 14:22:51] VERBOSE[15309] chan_sip.c: --- SIP read from UDP:109.235.33.10:1038 --- - [2010-09-18 14:22:52] VERBOSE[15309] chan_sip.c: Really destroying SIP dialog '194a9bd477ab104d236a1bcb778ff...@109.235.32.137' Method: OPTIONS [2010-09-18 14:34:12] VERBOSE[15309] chan_sip.c: Really destroying SIP dialog '6f71dc6f11b708b733be1ce869353...@109.235.32.36' Method: OPTIONS [2010-09-18 14:34:12] VERBOSE[15309] chan_sip.c: Really destroying SIP dialog '444376ba0f3d42746a9cd51761356...@109.235.32.36' Method: OPTIONS DELETED 50 other Really destroying messages [2010-09-18 14:34:12] VERBOSE[15309] chan_sip.c: Really destroying SIP dialog '43e3a3b678b75cef1a3d362f75188...@109.235.32.36' Method: OPTIONS [2010-09-18 14:34:12] VERBOSE[15309] chan_sip.c: Really destroying SIP dialog '20d68c0d288ef7f6682d0c1f7b5cd...@109.235.32.36' Method: OPTIONS [2010-09-18 14:34:12] VERBOSE[15309] chan_sip.c: --- SIP read from UDP:88.159.80.32:5060 --- NOTIFY sip:voip.pocos.nl SIP/2.0 Via: SIP/2.0/UDP 10.201.0.120:5060;branch=z9hG4bK-6ca6c7ca From: Fax sip:fa...@voip.pocos.nl;tag=613dd9bfc3f8c5co0 To: sip:voip.pocos.nl Call-ID: 1385478b-23141...@10.201.0.120 CSeq: 231 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug with Realtime?
My question is if you are using realtime, why are you doing a sip reload? I said previously:- Let's say I add a new provider to my service and therefore have to add another register= command into sip.conf, I'd have to issue a sip reload which would kill off all the realtime sip phones. Unless I can do register= in realtime too? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?
- Paul Belanger paul.belan...@polybeacon.com wrote: On Mon, Sep 20, 2010 at 11:48 AM, Olivier CALVANO o.calv...@gmail.com wrote: Anyone have a AudioCodes with Asterisk ??? Yes, but why? Both do the same thing. It would be like me asking 'I have a bike and need to get to work. Can I use the bike with a car?' Asterisk is *software* and as far as I can tell, it does not include any sort of provision for working with an E1 out of the box, hence the need for *hardware*. In this case, it appears the OP wants to use his Mediant 2000 gateway for the task. The Mediant interfaces with his E1, then provides service to Asterisk via SIP. So, as far as 'Both do the same thing', that seems outright incorrect. If Asterisk had access to an E1 card (Digium, Sangoma, OpenVox, etc), then yes, they *might* do the same thing, depending on how Asterisk was configured. OP, how about trying to configure it yourself, then ask specific questions about the problems you're having? Audiocodes has decent documentation that should help, and a bit of searching may provide the answers you seek. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug with Realtime?
On Mon, Sep 20, 2010 at 11:59:16AM -0400, Dan Journo wrote: Can we not do pastebin any more? No, it's just one user with an excessively paranoid and chatty mailfilter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Authentication best practice
I am working with a simple follow-me-style service: rather than have something that rings several phones in turn, the user dials a number (in the present implementation, unique to that user) to register his presence at a particular extension. What's the standard way to protect this from unauthorised use? Voicemail()-style, where the user has to enter a PIN once the connection is made? With a very long number, so that number and PIN can be integrated in the phone's contact list? With a single central number, where the each user has to enter his own unique identifier _and_ PIN? Roger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?
On Mon, Sep 20, 2010 at 8:58 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Mon, Sep 20, 2010 at 11:48 AM, Olivier CALVANO o.calv...@gmail.com wrote: Anyone have a AudioCodes with Asterisk ??? I use many AudioCode devices with Asterisk. Mostly Mediant 1000s and MP-114s, no Mediant 2000s. I would suggest you contact AudioCodes or your reseller, as AudioCodes has configuration guides that may help you. Here is a quick summary if I remember correctly: - Create a peer in Asterisk for the gateway - Configure the E1 on the Mediant (Provider specific) - Configure the SIP proxy (Asterisk) on the Mediant - Create a Trunk Group on the Mediant for the E1 - Configure IP to Trunk Group Routing to send calls out the Trunk Group If you have problems beyond that, contact whoever sold you the device. For the price they better offer some basic configuration support! You can also purchase support directly from AudioCodes. Yes, but why? Both do the same thing. It would be like me asking 'I have a bike and need to get to work. Can I use the bike with a car?' I would have to disagree with that statement. It is quite common to separate termination, call routing, and media for larger installations or to add some HA. Since termination is only part of the system, a better analogy might be different type of tires on the car. Sure you don't need snow tires, but you might want them when things get slick out! -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Authentication best practice
I implemented a scenario where all extension have to pass trough a kind of policy server (sql+ some script + dialplan) that enable/disable call feature, taht is you can call XXX but you don't YYY. I also added a Voicemail()-style + pin sceanrio to allow extensions to access specific trunks. Hope this can help. Hope this can give you some suggestions. Map On Mon, Sep 20, 2010 at 6:15 PM, Roger Burton West ro...@firedrake.orgwrote: I am working with a simple follow-me-style service: rather than have something that rings several phones in turn, the user dials a number (in the present implementation, unique to that user) to register his presence at a particular extension. What's the standard way to protect this from unauthorised use? Voicemail()-style, where the user has to enter a PIN once the connection is made? With a very long number, so that number and PIN can be integrated in the phone's contact list? With a single central number, where the each user has to enter his own unique identifier _and_ PIN? Roger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting 'fname_base' variable doesn't affect 'automon' result file.
On Monday 20 September 2010 11:01:36 Jose P. Espinal wrote:
Maybe I'm mistaken, but, shouldn't the 'fname_base' variable of
'Monitor' application affect the file name generated through 'automon'
feature?
I initialized this variable with a value as follows:
Set(fname_base=auto-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%
H%M%S)})
a. Should I use 'fname_base' in uppercase (FNAME_BASE)? or...
b. Is this variable independent of the 'automon' feature?
Where did you get the idea that fname_base is even a variable for you
to set? It's always been a parameter to the Monitor application.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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[asterisk-users] Commands needed via AMI to find callerid of inbound call to extension
Hi all, Can anyone help with the logic of which commands to use to say: 1. Extension is 600 2. See if has an ongoing call 3. Check if inbound or outbound to the extension 4. Find callerid of inbound call Been reading http://www.voip-info.org/wiki/view/Asterisk+manager+API Using latest 1.6. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension continues ringing after caller hanged up
You are right, the device is called Sipura SPA-3000. The settings are factory-set, I haven't changed anything beside of SIP registration with the asterisk. How can I enable SIP debug? -- Arie On Mon, Sep 20, 2010 at 18:01, Paul Belanger paul.belan...@polybeacon.comwrote: On Mon, Sep 20, 2010 at 7:31 AM, Arie Skliarouk sklia...@gmail.com wrote: When call arrives from PSTN, the phone continues ringing even after caller hanged up. I suspect a bug [1] but without a SIP debug, I cannot be sure. [1] https://reviewboard.asterisk.org/r/870/ -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting 'fname_base' variable doesn't affect 'automon' result file.
I got confused while reading the documentation.
Tilghman Lesher wrote:
On Monday 20 September 2010 11:01:36 Jose P. Espinal wrote:
Maybe I'm mistaken, but, shouldn't the 'fname_base' variable of
'Monitor' application affect the file name generated through 'automon'
feature?
I initialized this variable with a value as follows:
Set(fname_base=auto-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%
H%M%S)})
a. Should I use 'fname_base' in uppercase (FNAME_BASE)? or...
b. Is this variable independent of the 'automon' feature?
Where did you get the idea that fname_base is even a variable for you
to set? It's always been a parameter to the Monitor application.
--
Jose P. Espinal
http://www.eSlackware.com
IRC: Khratos @ #asterisk / -doc / -bugs
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Re: [asterisk-users] 3rd party app store
That Apple App Store really gets imaginations going, doesn't it? Wouldn't it be great to just publish an AGI script and see even 1% of the Asterisk installed base buy it for the low, low price of $49? Yes, that would be great. But one of the significant components of the moneymaking App Store is platform control. Apple's is the sole legitimate App Store for their platform. Digium gave that sort of control away with the source code to Asterisk. And really, who wants Digium to play middleman in their transaction? We watched with interest when our friends at RedHat launched the RedHat Exchange (RHX). After three years, VP Mike Evans said, We no longer believe that it is productive for Red Hat to try and front end the sale of third-party open source products. It's more effective for them to line up in sales channels with our partners. Source: URL:http://www.linuxplanet.com/linuxplanet/reports/6975/1/ In short, what open source-based ISVs want is not a virtual storefront, but exposure to new customers through existing sales channels and partners. That's why we created AsteriskExchange.com. AsteriskExchange.com provides free listings for free products and services, and paid listings for paid products and services. Digium has already consolidated many of its partnership types into the site, to cross-pollinate them and simplify the programs. The site receives roughly 1/10 the traffic of Asterisk.org and is growing. There's a difference in the visitor as well -- Asterisk newcomers visit Asterisk.org; Asterisk users looking for complementary products and services visit AsteriskExchange.com. Yes, there are fees associated with the AsteriskExchange. You've seen what happens on voip-info and other sites when the barrier is too low. AsteriskExchange is a business project that must earn enough to justify its development and tending. That said, please do get in touch with me if the listing fee is preventing you from joining. We aim to encourage innovative Asterisk applications, even the ones that haven't yet found commercial success. Could we do a better job at getting the word out? Yes, definitely. But we're also cautious about using Asterisk.org and such to promote things like AsteriskExchange. (Also a good reason to discuss this on asterisk-users rather than move it to -biz.) Is it a big deal to see commercial messages on Asterisk.org? We created AsteriskExchange.com as a separate website, but we could just fold it into the project site to raise its visibility and traffic. What waves (good and bad) might this create? What could we do to make the AsteriskExchange more effective? Thanks, rm -- Rod Montgomery Digium, Inc. | Product Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256 428 6267 fax: +1 256 864 0464 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing from zap to DAHDI
On 16 September 2010 15:03, Jerry Geis ge...@pagestation.com wrote: Jerry Geis wrote: below is the results of the command. grep -r ztconfig /etc/. grep: /etc/./httpd/run/asterisk.ctl: No such device or address grep: /etc/./httpd/run/dbus/system_bus_socket: No such device or address grep: /etc/./httpd/run/acpid.socket: No such device or address Jerry This is the new output (ztcfg is no longer mentioned) so I think that issue is fixed. Now its: + initlog -q -c 'unload_module dahdi' execvp: No such file or directory --- Make sure you have the linux-headers for your kernel installed and recompile dahdi. Brian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3rd party app store
What could we do to make the AsteriskExchange more effective? Rod, I'm not involved with Digium or even Asterisk on a daily basis so I don't know you, I also don't know your intentions but taking you at face value and answering your questions - I suggest you listen to the phone call and understand the reason why it was essential that Digium provide a central transaction point (re-use of Digium code and licensing server). Feel free to give me a call if you have any questions. Cheers, Dean -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Rod Montgomery Sent: Monday, 20 September 2010 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 3rd party app store That Apple App Store really gets imaginations going, doesn't it? Wouldn't it be great to just publish an AGI script and see even 1% of the Asterisk installed base buy it for the low, low price of $49? Yes, that would be great. But one of the significant components of the moneymaking App Store is platform control. Apple's is the sole legitimate App Store for their platform. Digium gave that sort of control away with the source code to Asterisk. And really, who wants Digium to play middleman in their transaction? We watched with interest when our friends at RedHat launched the RedHat Exchange (RHX). After three years, VP Mike Evans said, We no longer believe that it is productive for Red Hat to try and front end the sale of third-party open source products. It's more effective for them to line up in sales channels with our partners. Source: URL:http://www.linuxplanet.com/linuxplanet/reports/6975/1/ In short, what open source-based ISVs want is not a virtual storefront, but exposure to new customers through existing sales channels and partners. That's why we created AsteriskExchange.com. AsteriskExchange.com provides free listings for free products and services, and paid listings for paid products and services. Digium has already consolidated many of its partnership types into the site, to cross-pollinate them and simplify the programs. The site receives roughly 1/10 the traffic of Asterisk.org and is growing. There's a difference in the visitor as well -- Asterisk newcomers visit Asterisk.org; Asterisk users looking for complementary products and services visit AsteriskExchange.com. Yes, there are fees associated with the AsteriskExchange. You've seen what happens on voip-info and other sites when the barrier is too low. AsteriskExchange is a business project that must earn enough to justify its development and tending. That said, please do get in touch with me if the listing fee is preventing you from joining. We aim to encourage innovative Asterisk applications, even the ones that haven't yet found commercial success. Could we do a better job at getting the word out? Yes, definitely. But we're also cautious about using Asterisk.org and such to promote things like AsteriskExchange. (Also a good reason to discuss this on asterisk-users rather than move it to -biz.) Is it a big deal to see commercial messages on Asterisk.org? We created AsteriskExchange.com as a separate website, but we could just fold it into the project site to raise its visibility and traffic. What waves (good and bad) might this create? What could we do to make the AsteriskExchange more effective? Thanks, rm -- Rod Montgomery Digium, Inc. | Product Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256 428 6267 fax: +1 256 864 0464 Check us out at: http://digium.com http://asterisk.org -- __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3rd party app store
On 20/09/10 3:06 AM, Kevin P. Fleming wrote: There is no fee to list free products on AsteriskExchange. The main problem is the fee required to list non free products. If the fee was a percentage of the sale price then I'm sure it would work much better. Otherwise it becomes a catch 22. Nobody promotes the store because they can't afford to put their products on there, so nobody sells their products when listed on the store, so nobody list their products etc etc. If everybody who had products available was listing the products there, and Digium was taking a percentage cut, you'd see much better success from it, because people would redirect there. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk News Accepting Submissions
Hi all, Sorry for the crosspost but I assume this may be of interest to both businesses and users. The Daily Asterisk News (running since 2004) is now accepting article submissions. Basically I've created a submission form where you specify whether your post is commercial or non commercial and I'll be reviewing each article to check what it falls under. You'll be able to specify whether you want to hide commercial posts or not. What we're looking for is anything cool you're doing with Asterisk or any products you've created that work with Asterisk. If you have any ideas or suggestions, feel free to mail me on them. Oh, and we've moved the Daily Asterisk News web server to Dallas, TX, so it should be a bit quicker for those of you in the states - well, anywhere except New Zealand really :) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3rd party app store
Thanks, Dean. I was able to listen to that conference live. Digium's current licensing server has some limitations that make it unsuitable for general use. We are investigating options to improve the licensing platform, but have nothing to announce today. Even if we did, it would be only one missing component to a one-stop Asterisk software store. We'd also need a universal packaging format. AsteriskNOW (currently on CentOS 5.5) is happy with yum-installable RPM packages. It would be clean and simple for everyone to develop on that uniform image, but there is a lot of variety out there. The initial release of AsteriskNOW was on rPath Linux, which is marvelous for building software appliances, but unfamiliar to, well, everyone. Unlike a strictly controlled iPhone environment, there is no one solution that would work well for Asterisk developers. It would also be useful to have a ton of end-user information like iTMS gathered for years before the launch of the App Store. Part of the genius is that the transactional barrier is so low: millions have trusted Apple with payment details for music purchases, and need only tap Install to charge another payment for an iPad app. There must be hundred of thousands of installed Asterisk systems, but we only know the ones that become Digium customers. Also, there are a number of ways to build something marketable with Asterisk. Custom channels or resources, clever dialplan, AGI scripts, AMI-speaking services... it's often easier to incorporate Asterisk as a dependency into a purpose-built software appliance than to assume that Asterisk is at the center of the application's world. We cannot be all things to all people, especially when so many ecosystem partners are providing a service rather than a software product. Last but not least, Asterisk-based apps are not high-volume consumer content. I just don't see many telephony apps selling at a pace similar to music, movies, and games. Then I look to the RHX example I mentioned earlier, in which our friends at RedHat (and Novell before them) tried to become a hub of commerce around their flagship platform. And they failed. Customers didn't want a middleman. Customers wanted to be introduced to great products and services, and to do business directly with those third-party vendors. That's why AsteriskExchange is more a directory than a storefront. As a product manager, I can dream up a situation that imagines Digium as the all-controlling Apple of the Asterisk world, and conjures a ridiculously lucrative App Store that hauls in cash for talented and lucky developers that align with us. I even have a couple of black turtlenecks. But I am not convinced that more than a few want to use our current licensing mechanism. I am not convinced that the market wants Digium to be a central transaction point. I am not convinced that Digium should aspire (or stoop?) to that level of control. I am, however, convinced that ecosystem partners want to be visible to the Asterisk community. As Digium balances our goals of being a good sponsor of Asterisk and a profitable company, we tread very carefully on Asterisk.org. Perhaps keeping the goals apart is not as important as we make it out to be. It clearly has its negatives: keeping AsteriskExchange separate from Asterisk.org also separates it from the heavier visitor traffic. Does anyone reading this have an opinion on whether commercial listings for complementary products and services should appear directly on Asterisk.org? rm -- Rod Montgomery Digium, Inc. | Product Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256 428 6267 fax: +1 256 864 0464 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digest Username/auth name mismatch
Hi Thanks for help. I will try to help. But others might know more. What SIP client are you using - a softphone, a hardphone? It looks like the client is sending the full at 192.168.0.1 instead of just as the username. Sebastian That's right.hardphone is sending at 192.168.0.1 for Proprietary specification. ※Digest usrname can't change with SIP Client. so I would like to solve this hardphone issue with asterisk. thanks From: kein0...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 15 Sep 2010 03:19:55 + Subject: [asterisk-users] Digest Username/auth name mismatch Hi I'm sorry. I mailed the same question again. because, it cannot be yet solved. any ideas with asterisk? [Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username mismatch, have , digest has a...@192.168.0.1[aug 20 14:40:12] NOTICE[29315]: chan_sip.c:20479 handle_request_register: Registration from ' ' failed for '192.168.0.2' - Username/auth name mismatch [] type=friend username= secret= context= canreinvite=no host=dynamic disallow=allallow=ulaw The error seems that UAC set different username of digest. But UAC cannot send same username of digest and from for specification. *Digest username set a...@192.168.0.1 So I want to know how to solve with Asterisk. Register From: ;tag=644056924 To: Call-ID: 2457796...@192.168.0.2 CSeq: 125 REGISTER Contact: Authorization: Digest username=a...@192.168.0.1, realm=asterisk, nonce=3e635209, uri=sip:192.168.0.1, response=ec89ab3c90316e05d83774630488c61a, algorithm=MD5 Max-Forwards: 70 Expires: 3600 thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3rd party app store
On Mon, Sep 20, 2010 at 10:41 PM, Rod Montgomery rmontgom...@digium.com wrote: Does anyone reading this have an opinion on whether commercial listings for complementary products and services should appear directly on Asterisk.org? Personally, I would like to see less commercial marketing on http://asterisk.org. I count 5 separate marketing ads on the download page alone. This is just my opinion. However, on http://www.asteriskexchange.com, no problems. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
