Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)
very cool!
On Thu, Sep 22, 2011 at 10:37 AM, J. Oquendo aster...@tormenting.netwrote:
Apologies for cross posting but some of us aren't on the other list
(vice/versa) and thought both groups would benefit.
For those familiar with the VoIP Abuse Project, no need to explain the
gist of this. I got tired of parsing through the alerts (lists) I
receive via email daily. They're long and sometimes I don't have the
time to post them all. So for now, posting VoIP Abuse addresses straight
to Twitter.
So, anyone trying to compromise a pbx, is now autoposted on an hourly
basis to Twitter. Still working on pulling, have about 4 machines linked
up now, will mop em up during the week.
http://twitter.com/#!/voipabuse
Now, you can concoct a quick script off of it, e.g.:
links -dump http://twitter.com/voipabuse;|awk '/attacker/{print
iptables -A INPUT -s $2 -j DROP| sort -u}'
Will get a quickie soon from my Acme's, nCites, etc. when I have time.
For those NOT familiar with it, please Google it as I don't feel like
typing anymore ;) (sorry)
--
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM
It takes 20 years to build a reputation and five minutes to
ruin it. If you think about that, you'll do things
differently. - Warren Buffett
42B0 5A53 6505 6638 44BB 3943 2BF7 D83F 210A 95AF
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x2BF7D83F210A95AF
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Re: [asterisk-users] 10.0.0 better than 2.0.0?
please stop complaining about people complaining and filling up the mailing list. On Wed, Jul 27, 2011 at 8:31 AM, Bryant Zimmerman brya...@zktech.comwrote: Who really cares what the version number is. As long as the new version has new features in it and is more stable than the old one. Please all stop filling the mailing list with useless posts. Live with the new numbering schema and get on with life. Thanks zktech -- *From*: Pezhman Lali l...@lopl.net *Sent*: Wednesday, July 27, 2011 5:00 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] 10.0.0 better than 2.0.0? I think 2.x is more better. may be 10.X is more creative, and has a binary figure.but will not explain the true meaning On Mon, Jul 25, 2011 at 6:24 PM, Adam Moffett adamli...@plexicomm.netwrote: So next is version 11 and then version 100? it has been mentioned that 10 is of course 2 ... think not in base 10 On 22 July 2011 22:26, Matthew J. Rothmr...@imminc.com wrote: Kevin P. Fleming: The versions all go to ten. Look, right across the board, ten, ten, ten and... Asterisk Users: Oh, I see. And most open source projects upgrade to two? Kevin P. Fleming: Exactly. Asterisk Users: Does that mean it's better? Is it any better? Kevin P. Fleming: Well, it's eight better, isn't it? It's not two. You see, most blokes, you know, will be running at two. You're on two here, all the way up, all the way up, all the way up, you're on two on your software. Where can you go from there? Where? Asterisk Users: I don't know. Kevin P. Fleming: Nowhere. Exactly. What we do is, if we need that extra push over the cliff, you know what we do? Asterisk Users: Put it up to ten. Kevin P. Fleming: Ten. Exactly. Eight better. Asterisk Users: Why don't you just make two better and make two be the top number and make that a little better? Kevin P. Fleming: [pause] Asterisk goes to ten. -- Sorry, couldn't resist. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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 -- Pezhman Lali -- _ -- 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] asterisk's SDP
I see, thank you for explaning. The reason for my concern is, we are sometimes having DTMF issues on outbound calls. It seems when the user (Polycom) enters digits, they are not being recognized by the other end. On Thu, Jul 21, 2011 at 5:17 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 07/21/2011 03:54 PM, vip killa wrote: What if asterisk sends telephony events that are not in range of 0-15 though? You are misunderstanding how SDP works; when an SDP offer or answer is sent, that indicates what the sender is willing to *receive*, not what it is going to send. If the Sonus device sent fmtp:101 0-15 in its SDP, then Asterisk should not send 'event 16' events to it. If it does, that's a bug, although standard programming practices would mean that it wouldn't be harmful, it would just be ignored by the Sonus device. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.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-**usershttp://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] asterisk's SDP
I have a call trace of one of these calls...and this seems strange: asterisk sends on INVITE a=fmtp:101 0-16 then 183 Session progress is sent back with: a=fmtp:101 0-16 then asterisk sends 183 Session progress with: a=fmtp:127 0-16 OK is sent back with: a=fmtp:101 0-16 then asterisk sends OK with: a=fmtp:127 0-16 Would the above cause DTMF not to be read on remote end? On Fri, Jul 22, 2011 at 8:12 AM, vip killa vipki...@gmail.com wrote: I see, thank you for explaning. The reason for my concern is, we are sometimes having DTMF issues on outbound calls. It seems when the user (Polycom) enters digits, they are not being recognized by the other end. On Thu, Jul 21, 2011 at 5:17 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 07/21/2011 03:54 PM, vip killa wrote: What if asterisk sends telephony events that are not in range of 0-15 though? You are misunderstanding how SDP works; when an SDP offer or answer is sent, that indicates what the sender is willing to *receive*, not what it is going to send. If the Sonus device sent fmtp:101 0-15 in its SDP, then Asterisk should not send 'event 16' events to it. If it does, that's a bug, although standard programming practices would mean that it wouldn't be harmful, it would just be ignored by the Sonus device. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.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-**usershttp://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] asterisk's SDP
How can we wireshark a trace on the remote end? It is a peer such as Level3 or Dash On Fri, Jul 22, 2011 at 9:15 AM, Jesie Paluca jesie.pal...@gmail.comwrote: Most likely if DTMF is not recognized on the far end, it would be an incompatibility setting of DTMF support or bug on either UAC and UAS. Wireshark trace at both end will help you understand the issue. On Fri, Jul 22, 2011 at 8:12 PM, vip killa vipki...@gmail.com wrote: I see, thank you for explaning. The reason for my concern is, we are sometimes having DTMF issues on outbound calls. It seems when the user (Polycom) enters digits, they are not being recognized by the other end. On Thu, Jul 21, 2011 at 5:17 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 07/21/2011 03:54 PM, vip killa wrote: What if asterisk sends telephony events that are not in range of 0-15 though? You are misunderstanding how SDP works; when an SDP offer or answer is sent, that indicates what the sender is willing to *receive*, not what it is going to send. If the Sonus device sent fmtp:101 0-15 in its SDP, then Asterisk should not send 'event 16' events to it. If it does, that's a bug, although standard programming practices would mean that it wouldn't be harmful, it would just be ignored by the Sonus device. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.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-**usershttp://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 -- _ -- 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] 10.0.0 better than 2.0.0?
I agree, the numbering seems to make no sense. Oh well it's just an arbitrary measurement of non-progress anyway On Fri, Jul 22, 2011 at 3:44 PM, Tony Mountifield t...@mountifield.orgwrote: I read Kevin's piece in asterisk-announce about the new numbering scheme, and saw in svn-commits some tagging of 10.0.0-beta1. Perhaps I'm thick (I hope not!), but I really can't see why calling the next version 10.0.0 is any better than calling it 2.0.0! I'm surprised not to have seen ANY talk in asterisk-users or aserisk-dev about it, since the announcement. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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
[asterisk-users] asterisk's SDP
We have a peer (a Sonus Media Gateway), that sends a=fmtp:101 0-15 Asterisk sends 0-16 back, is there anyway to have asterisk send a 0-15? -- _ -- 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] asterisk's SDP
What if asterisk sends telephony events that are not in range of 0-15 though? On Thu, Jul 21, 2011 at 4:47 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 07/21/2011 03:30 PM, vip killa wrote: We have a peer (a Sonus Media Gateway), that sends a=fmtp:101 0-15 Asterisk sends 0-16 back, is there anyway to have asterisk send a 0-15? No, and it's completely unnecessary. Asterisk is willing to accept telephony-event codes 0 through 16, but the other endpoint is not obligated to send them if it doesn't want to. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.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-**usershttp://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] call paging interrupts call when using Mitel 5224
Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.net wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Tuesday, June 21, 2011 2:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] call paging interrupts call when using Mitel 5224 ** ** Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. ** ** This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- 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] call paging interrupts call when using Mitel 5224
How do you set them to Advanced SIP mode? On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.net wrote: My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. -- *From:* vip killa *Sent:* Wed 6/22/2011 8:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.netwrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Tuesday, June 21, 2011 2:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- 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] call paging interrupts call when using Mitel 5224
Hmm, could be im on old firmware but i don't see SIP Enhanced Mode and i followed instructions in that PDF. would you be able to tell me what firmware you are running? On Wed, Jun 22, 2011 at 9:23 AM, Terry Brummell te...@brummell.net wrote: http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf Page 32 -- *From:* vip killa *Sent:* Wed 6/22/2011 8:59 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 How do you set them to Advanced SIP mode? On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.netwrote: My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. -- *From:* vip killa *Sent:* Wed 6/22/2011 8:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.netwrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Tuesday, June 21, 2011 2:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- 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 -- _ -- 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] call paging interrupts call when using Mitel 5224
i just upgraded to R8.0.08.00.00.04 but i can't find the bootrom upgrade so i'm still running 02.03.02.02 tested and call is still being interrupted when paging it... are you running straight asterisk or is something else handling the dialplan when you test? On Wed, Jun 22, 2011 at 9:58 AM, Terry Brummell te...@brummell.net wrote: R7.2.07.02.00.04 And yes, that is likely the cause. -- *From:* vip killa *Sent:* Wed 6/22/2011 9:36 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Hmm, could be im on old firmware but i don't see SIP Enhanced Mode and i followed instructions in that PDF. would you be able to tell me what firmware you are running? On Wed, Jun 22, 2011 at 9:23 AM, Terry Brummell te...@brummell.netwrote: http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf Page 32 -- *From:* vip killa *Sent:* Wed 6/22/2011 8:59 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 How do you set them to Advanced SIP mode? On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.netwrote: My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. -- *From:* vip killa *Sent:* Wed 6/22/2011 8:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.netwrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Tuesday, June 21, 2011 2:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- 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 -- _ -- 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] call paging interrupts call when using Mitel 5224
Ahh then it makes sense, FreePBX checking to see if the line is in use, then sending busy signal instead of interrupting the call On Wed, Jun 22, 2011 at 10:13 AM, Terry Brummell te...@brummell.net wrote: PIAF with * 1.8.3 My bootrom is 2.3.2.2 also. -- *From:* vip killa *Sent:* Wed 6/22/2011 10:07 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 i just upgraded to R8.0.08.00.00.04 but i can't find the bootrom upgrade so i'm still running 02.03.02.02 tested and call is still being interrupted when paging it... are you running straight asterisk or is something else handling the dialplan when you test? On Wed, Jun 22, 2011 at 9:58 AM, Terry Brummell te...@brummell.netwrote: R7.2.07.02.00.04 And yes, that is likely the cause. -- *From:* vip killa *Sent:* Wed 6/22/2011 9:36 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Hmm, could be im on old firmware but i don't see SIP Enhanced Mode and i followed instructions in that PDF. would you be able to tell me what firmware you are running? On Wed, Jun 22, 2011 at 9:23 AM, Terry Brummell te...@brummell.netwrote: http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf Page 32 -- *From:* vip killa *Sent:* Wed 6/22/2011 8:59 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 How do you set them to Advanced SIP mode? On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.netwrote: My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. -- *From:* vip killa *Sent:* Wed 6/22/2011 8:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.netwrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Tuesday, June 21, 2011 2:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- 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 -- _ -- 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] call paging interrupts call when using Mitel 5224
Do you have BLF working on the Mitel? On Wed, Jun 22, 2011 at 10:36 AM, vip killa vipki...@gmail.com wrote: Ahh then it makes sense, FreePBX checking to see if the line is in use, then sending busy signal instead of interrupting the call On Wed, Jun 22, 2011 at 10:13 AM, Terry Brummell te...@brummell.netwrote: PIAF with * 1.8.3 My bootrom is 2.3.2.2 also. -- *From:* vip killa *Sent:* Wed 6/22/2011 10:07 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 i just upgraded to R8.0.08.00.00.04 but i can't find the bootrom upgrade so i'm still running 02.03.02.02 tested and call is still being interrupted when paging it... are you running straight asterisk or is something else handling the dialplan when you test? On Wed, Jun 22, 2011 at 9:58 AM, Terry Brummell te...@brummell.netwrote: R7.2.07.02.00.04 And yes, that is likely the cause. -- *From:* vip killa *Sent:* Wed 6/22/2011 9:36 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Hmm, could be im on old firmware but i don't see SIP Enhanced Mode and i followed instructions in that PDF. would you be able to tell me what firmware you are running? On Wed, Jun 22, 2011 at 9:23 AM, Terry Brummell te...@brummell.netwrote: http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf Page 32 -- *From:* vip killa *Sent:* Wed 6/22/2011 8:59 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 How do you set them to Advanced SIP mode? On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.netwrote: My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. -- *From:* vip killa *Sent:* Wed 6/22/2011 8:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.netwrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Tuesday, June 21, 2011 2:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] call paging interrupts call when using Mitel 5224
Any chance you could send me (off list) you're example provisioning files (without the SIP credentials and IPs of course)? I can't find them anywhere online. On Wed, Jun 22, 2011 at 11:21 AM, Terry Brummell te...@brummell.net wrote: Yes. -- *From:* vip killa *Sent:* Wed 6/22/2011 10:56 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Do you have BLF working on the Mitel? On Wed, Jun 22, 2011 at 10:36 AM, vip killa vipki...@gmail.com wrote: Ahh then it makes sense, FreePBX checking to see if the line is in use, then sending busy signal instead of interrupting the call On Wed, Jun 22, 2011 at 10:13 AM, Terry Brummell te...@brummell.netwrote: PIAF with * 1.8.3 My bootrom is 2.3.2.2 also. -- *From:* vip killa *Sent:* Wed 6/22/2011 10:07 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 i just upgraded to R8.0.08.00.00.04 but i can't find the bootrom upgrade so i'm still running 02.03.02.02 tested and call is still being interrupted when paging it... are you running straight asterisk or is something else handling the dialplan when you test? On Wed, Jun 22, 2011 at 9:58 AM, Terry Brummell te...@brummell.netwrote: R7.2.07.02.00.04 And yes, that is likely the cause. -- *From:* vip killa *Sent:* Wed 6/22/2011 9:36 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Hmm, could be im on old firmware but i don't see SIP Enhanced Mode and i followed instructions in that PDF. would you be able to tell me what firmware you are running? On Wed, Jun 22, 2011 at 9:23 AM, Terry Brummell te...@brummell.netwrote: http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf Page 32 -- *From:* vip killa *Sent:* Wed 6/22/2011 8:59 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 How do you set them to Advanced SIP mode? On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.netwrote: My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. -- *From:* vip killa *Sent:* Wed 6/22/2011 8:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.netwrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Tuesday, June 21, 2011 2:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
[asterisk-users] DTMF begin ignored
we've been getting complaints that DTMF is not working, i checked full log for a call that they claimed DTMF didnt work, I noticed this: DTMF begin '7' received DTMF begin ignored DTMF end '7' received DTMF end passthrough '7' why is the DTMF begin ignored called? -- _ -- 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] call paging interrupts call when using Mitel 5224
Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. -- _ -- 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 voicemail distribution groups
Is there any to have asterisk record a file then send that file to a distribution list of voicemail boxes? What I'm trying to accomplish is a prompt for a user to record/listen to their message and then choose to send the recording to multiple voicemail box's inboxes -- _ -- 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] asterisk voicemail distribution groups
Or is there anyway to have a message copied from a mailbox to a list of other mailboxes everytime a message is left in it? On Fri, Jun 17, 2011 at 1:54 PM, vip killa vipki...@gmail.com wrote: Is there any to have asterisk record a file then send that file to a distribution list of voicemail boxes? What I'm trying to accomplish is a prompt for a user to record/listen to their message and then choose to send the recording to multiple voicemail box's inboxes -- _ -- 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] fast AGI memory leaks
Could someone point me in the right direction of how to create a Fast AGI script without memory leaks? I was told i need to clear the result set for mysql queries, Im not sure how to do that. My script is a simple perl script of 70 lines doing database lookups and executing dial and voicemail I'd be happy to post the code, thanks in advance. -- _ -- 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] change destination on digit
Is there an easy way to setup diaplan so when someone pushes a digit such as * during a call, they will be transferred to another destination. For example, a caller is hearing ringing while calling a UA, but instead of waiting for the UA to pick up, they can push * and go directly to that UA's voicemail. -- _ -- 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 1.6 - 511 Command not permitted causing high CPU usage
http://pastebin.com/vxGM2n5j We are getting those errors 100x per second in console when AGI set debug is on It is causing extremely high CPU usage, we've tried asterisk version 1.6.1.22 and 1.6.2.18 It seems the problem is worse in 1.6.2.18 Can someone advise how to fix this? Thank you. -- _ -- 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] benefits of asterisk 1.8
digium, eat your dog food! On Sun, Jun 5, 2011 at 11:08 AM, dotnetdub dotnet...@gmail.com wrote: On 3 June 2011 22:41, Hans Witvliet h...@a-domani.nl wrote: On Fri, 2011-06-03 at 09:07 +0100, Ishfaq Malik wrote: Are you suggesting that there are no bugs in 1.4 or 1.6? I presume that you are aware of the fact that it is impossible to prove the absence of bugs in any piece of software You might not have detected them yet. Furthermore behaviour that might have been coded on purpose, can be considered eroneously some time later. Currently there seems to be a fear of 1.8. We're about to put it into production and yes, we've had issues with it, mostly due to the fact we use RealTime, but before you change anything it is always advisable to test the hell out of it. To anyone who is thinking of moving to 1.8 the question is not, 'is it stable?'. The question is, 'have I comprehensively tested it to show that it is suitable for my needs?' If you put it into production, test at least the functions that you are going to use. There might (and probably will) problems in the code, but as long as it does not bother you, so what? See this thread here about Asterisk 1.8 - and Digium's view on the matter. http://nerdvittles.com/?p=743 -- _ -- 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] benefits of asterisk 1.8
can someone explain to me the benefits of upgrading to version 1.8? we are currently running 1.6 I know one benefit of 1.8 is digium supports it also, how stable is version 1.8 compared to 1.6? Thank you for you input. -- _ -- 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] benefits of asterisk 1.8
what do you mean Asterisk 1.8 is not stable enough yet? Can you give specific examples/scenarios? On Thu, Jun 2, 2011 at 9:28 AM, Satish Barot satish4aster...@gmail.comwrote: So many new features have been added in 1.8. Check this...https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 Nope, Asterisk 1.8 is not stable enough yet. [SATISH] On Thu, Jun 2, 2011 at 6:33 PM, Gopal krishnan gopalakrishnan.an@ gmail.com wrote: 1.8 is stable when compared to 1.6, also in 1.8 you will get Long Term support. On Thu, Jun 2, 2011 at 6:31 PM, vip killa vipki...@gmail.com wrote: can someone explain to me the benefits of upgrading to version 1.8? we are currently running 1.6 I know one benefit of 1.8 is digium supports it also, how stable is version 1.8 compared to 1.6? Thank you for you input. -- _ -- 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 -- _ -- 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] disable sip registration
Is there a way to disable all SIP registration and block any requests? The reason I'm asking is this particular Asterisk server will just be originating calls. I've noticed sip attacks where the attacker attempts to register a user 100x per second causing CPU to rise significantly. -- _ -- 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] DB driven voicemail
try using voicemail_odbc On Thu, May 26, 2011 at 2:19 PM, Abdul Basit basit.e...@gmail.com wrote: Have anyone setup voicemail using DB? I am facing problems with asterisk realtime voicemail setup. Asterisk authenticate and saves new voicemail records in mysql with voice file path. /var/spool/asterisk/voicemail/default/337/INBOX/msg0001 When we listen voiemails, app_voicemail deletes old record from voicemail_data and inserts a new one with new file name in Old folder. /var/spool/asterisk/voicemail/default/337/Old/msg Please note that message name also changed. How to handle voicemail in asterisk realtime? -- Regards, Abdul Basit -- _ -- 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] asterisk 1.6.2.17.2 Warning on startup
I get this line several times when starting asterisk in verbose mode: WARNING[3812]: utils.c:1536 __ast_string_field_init: trying to reset empty pool What does it mean and can i fix it? -- _ -- 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] asterisk's zombie processes
I'm sure it's not nagios. I'm not running check_sip and i'm running nagios' NRPE on several other machines that do not have asterisk running. On Wed, May 18, 2011 at 4:43 PM, Alex Balashov abalas...@evaristesys.comwrote: Are you sure it's Asterisk creating the zombie processes, not the check_sip pinger in Nagios? Nagios is extremely bad with high throughput and concurrency, and check_sip is a wrapper around 'sipsak', which means it takes the full Timer T1 * 64 to time out if the Asterisk server is truly not available (about ~30-32 sec). On 05/18/2011 04:40 PM, vip killa wrote: I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -- _ -- 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 -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] asterisk's zombie processes
we are in a production environment and cannot reboot. besides, these zombie processes appear minutes after asterisk starts taking calls. On Thu, May 19, 2011 at 8:59 AM, Satish Patel satish...@hotmail.com wrote: Sometime reboot does help. -- Sent from my iPhone On May 19, 2011, at 8:09 AM, vip killa vipki...@gmail.com wrote: I'm sure it's not nagios. I'm not running check_sip and i'm running nagios' NRPE on several other machines that do not have asterisk running. On Wed, May 18, 2011 at 4:43 PM, Alex Balashov abalas...@evaristesys.com abalas...@evaristesys.com wrote: Are you sure it's Asterisk creating the zombie processes, not the check_sip pinger in Nagios? Nagios is extremely bad with high throughput and concurrency, and check_sip is a wrapper around 'sipsak', which means it takes the full Timer T1 * 64 to time out if the Asterisk server is truly not available (about ~30-32 sec). On 05/18/2011 04:40 PM, vip killa wrote: I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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] asterisk's zombie processes
Actually not sure if it is asterisk generating these zombies... i'm starting to believe it's the enswitch_routed daemon, anybody familiar with enswitch? On Thu, May 19, 2011 at 9:02 AM, vip killa vipki...@gmail.com wrote: we are in a production environment and cannot reboot. besides, these zombie processes appear minutes after asterisk starts taking calls. On Thu, May 19, 2011 at 8:59 AM, Satish Patel satish...@hotmail.comwrote: Sometime reboot does help. -- Sent from my iPhone On May 19, 2011, at 8:09 AM, vip killa vipki...@gmail.com wrote: I'm sure it's not nagios. I'm not running check_sip and i'm running nagios' NRPE on several other machines that do not have asterisk running. On Wed, May 18, 2011 at 4:43 PM, Alex Balashov abalas...@evaristesys.com abalas...@evaristesys.com wrote: Are you sure it's Asterisk creating the zombie processes, not the check_sip pinger in Nagios? Nagios is extremely bad with high throughput and concurrency, and check_sip is a wrapper around 'sipsak', which means it takes the full Timer T1 * 64 to time out if the Asterisk server is truly not available (about ~30-32 sec). On 05/18/2011 04:40 PM, vip killa wrote: I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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
[asterisk-users] asterisk's zombie processes
I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -- _ -- 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] AMI perl daemon
is there a simple way to receive a response from an action such as a Ping
?
On Mon, May 16, 2011 at 4:21 PM, Alex Balashov abalas...@evaristesys.comwrote:
On 05/16/2011 04:17 PM, vip killa wrote:
forgive me for i am very new to asterisk and perl. but how could you
detect if you were disconnected from AMI?
if(defined($mgr_sock)) would evaluate to false. That's all you need to do
with the plain vanilla blocking I/O you're using now.
Down the road, if non-blocking I/O is set, there are other strategies. The
traditional way was an ioctl() FIONREAD that returned 0 for the bytes value,
though the ioctl() call did not fail:
require 'sys/ioctl.ph';
...
my $bytes_waiting = pack(L, 0);
ioctl($mgr_sock, FIONREAD(), $bytes_wating);
$bytes_waiting = unpack(L, $bytes_waiting);
if($bytes_waiting == 0) {
# Far-end disconnected.
close($mgr_sock);
return;
}
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
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[asterisk-users] AMI perl daemon
Would anybody know how to run a perl script as a daemon that would stay connected to asterisk via AMI? Right now, my AMI script connects to the manager interface, originates a call, disconnects. The script will be run maybe 20+ per minute. It would make more sense to me to have the script run as a daemon and have a persistent connection to asterisk's AMI. Thank you in advance for your input. -- _ -- 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] AMI perl daemon
If the script were called each time an extension were dialed in a dialplan for example, wouldn't each new instance of the script need to re-connect to AMI, run command, disconnect? On Mon, May 16, 2011 at 8:16 AM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 08:14 AM, vip killa wrote: Would anybody know how to run a perl script as a daemon that would stay connected to asterisk via AMI? Right now, my AMI script connects to the manager interface, originates a call, disconnects. The script will be run maybe 20+ per minute. It would make more sense to me to have the script run as a daemon and have a persistent connection to asterisk's AMI. Thank you in advance for your input. Well, you would just write the Perl script in such a way as to not close the connection :-), but continue reading from the socket, ideally in an asynchronous manner. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] AMI perl daemon
Thank you, that makes sense but actually I would be invoking the script using the externnotify in voicemail.conf, similar to externnotify = /var/lib/asterisk/scripts/notify.pl I assume externnotify cannot call the FastAGI server...correct? On Mon, May 16, 2011 at 8:23 AM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 08:19 AM, vip killa wrote: If the script were called each time an extension were dialed in a dialplan for example, wouldn't each new instance of the script need to re-connect to AMI, run command, disconnect? Well, yes, if you invoke a new instance of the script each time, that is what would happen. The desired approach is to have some means of communicating with the running daemon to indicate to it that it should originate a call, perhaps via a control socket/API. If your invocation is in the dial plan, the simplest thing to do would be to build a FastAGI server in Perl. This CPAN module can save some work: http://search.cpan.org/~jaywhy/Asterisk-FastAGI-0.02/lib/Asterisk/FastAGI.pm Then have that process either maintain a persistent AMI connection, or open a new one each time if you don't feel like/don't know how to implement the asynchronous approach. When you want to initiate a dial, just call: exten = ...,x,AGI(agi://some.server.ip/your_script) Of course, you could also use call files if the script is executing on the same Asterisk server as the one on which the dials take place. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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
[asterisk-users] AMI check if connection is alive
I'm using a perl daemon i wrote to connect to AMI and perform actions. The daemon connects to asterisk via AMI at start up. Is there anyway to check if the AMI connection is still alive, for example every 2 seconds. if the connection is not alive, re-connect to AMI? Also, does AMI timeout after a certain amount of time of not sending commands? -- _ -- 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] AMI perl daemon
i was able to create a daemon that queries a database every 2 seconds for outbound calls. the daemon originates a call to a destination determined by the database. what i've noticed is, after the originate, the script never does anything else. it seems i have to use Async or the AMI will disconnect, so i tried using OriginateHack=1 but still no dice... any ideas? On Mon, May 16, 2011 at 11:37 AM, Ryan Bullock rrb3...@gmail.com wrote: Alex is pointing you in the right direction. You should want a single daemon running that then gets notified by the voicemail script, either through a FIFO, a socket, or by dropping a file in a watched directory. If you are going to write a daemon, I would suggest looking at : http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.5/ It has integration with event loops and should work well for what you are doing. It also has some features for detecting disconnects and timeouts. On Mon, May 16, 2011 at 5:42 AM, Alex Balashov abalas...@evaristesys.com wrote: On 05/16/2011 08:33 AM, vip killa wrote: Thank you, that makes sense but actually I would be invoking the script using the externnotify in voicemail.conf, similar to externnotify = /var/lib/asterisk/scripts/notify.pl http://notify.pl I assume externnotify cannot call the FastAGI server...correct? That is correct. But you can call a script that notifies the daemon through a FIFO or UNIX domain socket, if local, or network socket if remote. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- _ -- 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] AMI perl daemon
http://pastebin.com/W5h9AMrQ anything else you need to see? On Mon, May 16, 2011 at 2:45 PM, Ryan Bullock rrb3...@gmail.com wrote: A normal Originate over the AMI will block all other actions until it completes. So to do other commands while the Originate is still going you have to call Originate with the Async option. I would suggest using an Originate with the 'Async' option and OriginateHack=1. If that is still not working I would have to see your code. Unfortunately I am not on irc today. On Mon, May 16, 2011 at 11:16 AM, vip killa vipki...@gmail.com wrote: i was able to create a daemon that queries a database every 2 seconds for outbound calls. the daemon originates a call to a destination determined by the database. what i've noticed is, after the originate, the script never does anything else. it seems i have to use Async or the AMI will disconnect, so i tried using OriginateHack=1 but still no dice... any ideas? On Mon, May 16, 2011 at 11:37 AM, Ryan Bullock rrb3...@gmail.com wrote: Alex is pointing you in the right direction. You should want a single daemon running that then gets notified by the voicemail script, either through a FIFO, a socket, or by dropping a file in a watched directory. If you are going to write a daemon, I would suggest looking at : http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.5/ It has integration with event loops and should work well for what you are doing. It also has some features for detecting disconnects and timeouts. On Mon, May 16, 2011 at 5:42 AM, Alex Balashov abalas...@evaristesys.com wrote: On 05/16/2011 08:33 AM, vip killa wrote: Thank you, that makes sense but actually I would be invoking the script using the externnotify in voicemail.conf, similar to externnotify = /var/lib/asterisk/scripts/notify.pl http://notify.pl I assume externnotify cannot call the FastAGI server...correct? That is correct. But you can call a script that notifies the daemon through a FIFO or UNIX domain socket, if local, or network socket if remote. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- _ -- 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] AMI perl daemon
could you suggest a better method where the perl-daemon stays persistently connected to asterisk's AMI ? On Mon, May 16, 2011 at 3:21 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 03:19 PM, Ryan Bullock wrote: You could us a timer to periodically poll your database and do non-blocking originates (with async) with callbacks to catch the response, update the log, and do the delete. http://en.wikipedia.org/wiki/Anti-pattern http://en.wikipedia.org/wiki/Database-as-IPC *ducks* :-) -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] AMI perl daemon
yes, my problem is i would like a persistent connection to AMI because there will maybe be 20+ originates per second. On Mon, May 16, 2011 at 3:37 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 03:35 PM, vip killa wrote: could you suggest a better method where the perl-daemon stays persistently connected to asterisk's AMI ? It is not the AMI connection that is under discussion. The AMI connection will be over a TCP socket regardless, because that is the nature of the interface. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] AMI perl daemon
you are incorrect, the while loop never exits...and i already have async=1 On Mon, May 16, 2011 at 3:41 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 03:38 PM, vip killa wrote: yes, my problem is i would like a persistent connection to AMI because there will maybe be 20+ originates per second. What was wrong with Ryan's original Async suggestion to address that? In other words, what is currently the problem? I looked at your code and it seems to me that your while() loop is going to exit when all rows are retrieved from your 'active' table as part of the current SELECT transaction. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] AMI perl daemon
yes, it's originating the call and never responding. On Mon, May 16, 2011 at 3:47 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 03:44 PM, vip killa wrote: you are incorrect, the while loop never exits...and i already have async=1 I meant this while() loop: while(my @row = $res-fetchrow()) In other words, it is possible for additional rows to be added to the table after the SELECT query has been made and prior to the conclusion of this loop, which will incur a sleep(2) penalty. That's not terribly important, though. So, just to make sure I understand, you're calling the $astman-action with the Originate parameters and it's sitting there and blocking until the Originate completes with a final disposition, despite the presence of the Async option? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] AMI perl daemon
Umm thank you...apparently AMI::Asterisk sucks because that code did
everything i needed in one try. thanks again!
On Mon, May 16, 2011 at 3:58 PM, Alex Balashov abalas...@evaristesys.comwrote:
On 05/16/2011 03:48 PM, vip killa wrote:
yes, it's originating the call and never responding.
This sounds to me like a possible problem with the Asterisk::AMI module,
although I am unsure what the problem is, since I am not familiar with its
internal architecture and have never used it.
To debug, try initiating the AMI connection and issuing the Originate
statement raw:
use IO::Socket;
...
my $mgr_sock = IO::Socket::INET-new(
'PeerAddr' = '127.0.0.1',
'PeerPort' = 5038,
'Type' = SOCK_STREAM,
'Protocol' = 'TCP',
'Timeout' = 5);
print $mgr_sock
Action: login\r\n .
Username: XX\r\n .
Secret: XX\r\n .
\r\n;
sleep(1);
...
print $mgr_sock
Action: Originate\r\n .
Channel: Local/$row[2]\@outbound\r\n .
Context: page\r\n .
CallerID: $row[1]\r\n .
Exten: $row[1]\r\n .
Priority: 1\r\n .
Async: 1\r\n .
\r\n;
while(defined($mgr_sock) $_ = $mgr_sock) {
print;
}
sleep(1);
close $mgr_sock;
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
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Re: [asterisk-users] AMI perl daemon
question... how reliable is what you wrote? as long as the daemon is running
will the AMI stay connected?
On Mon, May 16, 2011 at 4:08 PM, vip killa vipki...@gmail.com wrote:
Umm thank you...apparently AMI::Asterisk sucks because that code did
everything i needed in one try. thanks again!
On Mon, May 16, 2011 at 3:58 PM, Alex Balashov
abalas...@evaristesys.comwrote:
On 05/16/2011 03:48 PM, vip killa wrote:
yes, it's originating the call and never responding.
This sounds to me like a possible problem with the Asterisk::AMI module,
although I am unsure what the problem is, since I am not familiar with its
internal architecture and have never used it.
To debug, try initiating the AMI connection and issuing the Originate
statement raw:
use IO::Socket;
...
my $mgr_sock = IO::Socket::INET-new(
'PeerAddr' = '127.0.0.1',
'PeerPort' = 5038,
'Type' = SOCK_STREAM,
'Protocol' = 'TCP',
'Timeout' = 5);
print $mgr_sock
Action: login\r\n .
Username: XX\r\n .
Secret: XX\r\n .
\r\n;
sleep(1);
...
print $mgr_sock
Action: Originate\r\n .
Channel: Local/$row[2]\@outbound\r\n .
Context: page\r\n .
CallerID: $row[1]\r\n .
Exten: $row[1]\r\n .
Priority: 1\r\n .
Async: 1\r\n .
\r\n;
while(defined($mgr_sock) $_ = $mgr_sock) {
print;
}
sleep(1);
close $mgr_sock;
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
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Re: [asterisk-users] AMI perl daemon
forgive me for i am very new to asterisk and perl. but how could you detect if you were disconnected from AMI? On Mon, May 16, 2011 at 4:14 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 04:10 PM, vip killa wrote: question... how reliable is what you wrote? as long as the daemon is running will the AMI stay connected? I don't know, it was kind of off-the-cuff. I would probably throw a while() loop around it to reconnect if the connection is lost. But I see no reason why it should disconnect unless the Asterisk AMI service has some sort of inactivity timeout. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] receive faxes
I would never choose to use it. Our system is built on top of it (before I ever got here) and it would be too great a task to change it not to mention management would not go for a change. On Wed, May 4, 2011 at 6:09 PM, Matt Riddell li...@venturevoip.com wrote: On 5/05/11 3:02 AM, vip killa wrote: Honestly Digium's Asterisk is not a quality project. Though it has lead the way in innovative open-source VoIP, it's a flawed and chaotic project. Hence, I refuse to pay Digium. So why do you use it? -- 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 -- _ -- 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] receive faxes
The majority of open source projects out are NOT run by commercial institutions... they are run by people committed to a better product (not to making money)... they are maintained by people who have the resources to host a repository (which does not take a lot of resources) and a community of dedicated people who want to advance the product (and again NOT make money)... what people do with the product is their decision and if they want to use to it make money that is perfectly acceptable, but commercial support or commercial addons provided by the maintainers (who are obviously chosen and paid by Diguim in this case) of an opensource product is obscene. Digium has a monopoly on asterisk and if you can't get something to work, that means more money for them, if they break something you may end up paying them $$ to fix it. On Thu, May 5, 2011 at 12:29 PM, Andrew Joakimsen joakim...@gmail.comwrote: It isn't any better than the so called t.38 support in Asterisk that only drops calls. Gee I wonder why, maybe so they can sell their fax product? On Wednesday, May 4, 2011, Steve Edwards asterisk@sedwards.com wrote: On Wed, 4 May 2011, vip killa wrote: screw that i just got hylafax to work with IAXMODEM...i refuse to pay digium a dime... supposed to be open-source right? Great attitude. Should be worth about a bazillion bad karma points. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- 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] receive faxes
I've given up on trying T38 because there is no universal support for it...
Can someone recommend another way of faxing without using T38?
On Tue, May 3, 2011 at 5:13 PM, satish patel satish...@hotmail.com wrote:
Enable debug and verbose on CLI ?
Did you enable and also at logger.conf
full = notice,warning,error,debug,verbose,dtmf,fax
--
Date: Tue, 3 May 2011 16:12:06 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have full log.. only thing that stands out are two warnings:
[May 3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.
result=13: Unexpected message received.
[May 3 16:10:40] WARNING[18176] app_fax.c: Transmission failed
On Tue, May 3, 2011 at 4:05 PM, satish patel satish...@hotmail.comwrote:
I'd enable full debug at logger.conf and try to find issue.
-S
--
Date: Tue, 3 May 2011 15:55:51 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
I tried with those settings and without... same error:
WARNING[18090]: app_fax.c:820 transmit: Transmission failed
On Tue, May 3, 2011 at 3:32 PM, satish patel satish...@hotmail.comwrote:
did you set faxdetect=both or incoming
and faxbuffer=?
-S
--
Date: Tue, 3 May 2011 15:28:36 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed
when trying to fax from a POTS line...
On Tue, May 3, 2011 at 3:27 PM, satish patel satish...@hotmail.comwrote:
You need spandsp i guess following is my dialplan is working example
[fax]
exten = 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten = 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten = 9000,n,ReceiveFax(${FAXFILE})
exten = 9000,n,Hangup()
--
Date: Tue, 3 May 2011 15:20:33 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] receive faxes
does anybody know a good tutorial on how to setup asterisk to receive faxes
(no need to send them) ? i've tried using app_fax.so with T38 but i keep
getting Transmission failed
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Re: [asterisk-users] receive faxes
doesn't digium fax cost money?
On Wed, May 4, 2011 at 9:21 AM, Satish Patel satish...@hotmail.com wrote:
Did you try digim fax ?
Also you can record you incoming fax via mxmonitor and analize it.
--
Sent from my iPhone
On May 4, 2011, at 8:50 AM, vip killa vipki...@gmail.com wrote:
I've given up on trying T38 because there is no universal support for it...
Can someone recommend another way of faxing without using T38?
On Tue, May 3, 2011 at 5:13 PM, satish patel satish...@hotmail.com
satish...@hotmail.com wrote:
Enable debug and verbose on CLI ?
Did you enable and also at logger.conf
full = notice,warning,error,debug,verbose,dtmf,fax
--
Date: Tue, 3 May 2011 16:12:06 -0400
From: vipki...@gmail.comvipki...@gmail.com
To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have full log.. only thing that stands out are two warnings:
[May 3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.
result=13: Unexpected message received.
[May 3 16:10:40] WARNING[18176] app_fax.c: Transmission failed
On Tue, May 3, 2011 at 4:05 PM, satish patel satish...@hotmail.com
satish...@hotmail.com wrote:
I'd enable full debug at logger.conf and try to find issue.
-S
--
Date: Tue, 3 May 2011 15:55:51 -0400
From: vipki...@gmail.comvipki...@gmail.com
To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
I tried with those settings and without... same error:
WARNING[18090]: app_fax.c:820 transmit: Transmission failed
On Tue, May 3, 2011 at 3:32 PM, satish patel satish...@hotmail.com
satish...@hotmail.com wrote:
did you set faxdetect=both or incoming
and faxbuffer=?
-S
--
Date: Tue, 3 May 2011 15:28:36 -0400
From: vipki...@gmail.comvipki...@gmail.com
To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed
when trying to fax from a POTS line...
On Tue, May 3, 2011 at 3:27 PM, satish patel satish...@hotmail.com
satish...@hotmail.com wrote:
You need spandsp i guess following is my dialplan is working example
[fax]
exten = 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten = 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten = 9000,n,ReceiveFax(${FAXFILE})
exten = 9000,n,Hangup()
--
Date: Tue, 3 May 2011 15:20:33 -0400
From: vipki...@gmail.comvipki...@gmail.com
To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com
Subject: [asterisk-users] receive faxes
does anybody know a good tutorial on how to setup asterisk to receive
faxes (no need to send them) ? i've tried using app_fax.so with T38 but i
keep getting Transmission failed
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Re: [asterisk-users] receive faxes
meaning asterisk can receive only 1 fax at a time?
On Wed, May 4, 2011 at 9:47 AM, Satish Patel satish...@hotmail.com wrote:
Single channel license is free.
--
Sent from my iPhone
On May 4, 2011, at 9:44 AM, vip killa vipki...@gmail.com wrote:
doesn't digium fax cost money?
On Wed, May 4, 2011 at 9:21 AM, Satish Patel satish...@hotmail.com
satish...@hotmail.com wrote:
Did you try digim fax ?
Also you can record you incoming fax via mxmonitor and analize it.
--
Sent from my iPhone
On May 4, 2011, at 8:50 AM, vip killa vipki...@gmail.com
vipki...@gmail.com wrote:
I've given up on trying T38 because there is no universal support for
it... Can someone recommend another way of faxing without using T38?
On Tue, May 3, 2011 at 5:13 PM, satish patel
satish...@hotmail.comsatish...@hotmail.com
satish...@hotmail.com wrote:
Enable debug and verbose on CLI ?
Did you enable and also at logger.conf
full = notice,warning,error,debug,verbose,dtmf,fax
--
Date: Tue, 3 May 2011 16:12:06 -0400
From: vipki...@gmail.com vipki...@gmail.comvipki...@gmail.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have full log.. only thing that stands out are two warnings:
[May 3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.
result=13: Unexpected message received.
[May 3 16:10:40] WARNING[18176] app_fax.c: Transmission failed
On Tue, May 3, 2011 at 4:05 PM, satish patel
satish...@hotmail.comsatish...@hotmail.com
satish...@hotmail.com wrote:
I'd enable full debug at logger.conf and try to find issue.
-S
--
Date: Tue, 3 May 2011 15:55:51 -0400
From: vipki...@gmail.com vipki...@gmail.comvipki...@gmail.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
I tried with those settings and without... same error:
WARNING[18090]: app_fax.c:820 transmit: Transmission failed
On Tue, May 3, 2011 at 3:32 PM, satish patel
satish...@hotmail.comsatish...@hotmail.com
satish...@hotmail.com wrote:
did you set faxdetect=both or incoming
and faxbuffer=?
-S
--
Date: Tue, 3 May 2011 15:28:36 -0400
From: vipki...@gmail.com vipki...@gmail.comvipki...@gmail.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed
when trying to fax from a POTS line...
On Tue, May 3, 2011 at 3:27 PM, satish patel
satish...@hotmail.comsatish...@hotmail.com
satish...@hotmail.com wrote:
You need spandsp i guess following is my dialplan is working example
[fax]
exten = 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten =
9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten = 9000,n,ReceiveFax(${FAXFILE})
exten = 9000,n,Hangup()
--
Date: Tue, 3 May 2011 15:20:33 -0400
From: vipki...@gmail.com vipki...@gmail.comvipki...@gmail.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
asterisk-users@lists.digium.com
Subject: [asterisk-users] receive faxes
does anybody know a good tutorial on how to setup asterisk to receive
faxes (no need to send them) ? i've tried using app_fax.so with T38 but i
keep getting Transmission failed
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Re: [asterisk-users] receive faxes
screw that i just got hylafax to work with IAXMODEM...i refuse to pay
digium a dime... supposed to be open-source right?
On Wed, May 4, 2011 at 9:52 AM, Danny Nicholas da...@debsinc.com wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Wednesday, May 04, 2011 8:49 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] receive faxes
meaning asterisk can receive only 1 fax at a time?
On Wed, May 4, 2011 at 9:47 AM, Satish Patel satish...@hotmail.com
wrote:
Single channel license is free.
--
Sent from my iPhone
On May 4, 2011, at 9:44 AM, vip killa vipki...@gmail.com wrote:
doesn't digium fax cost money?
On Wed, May 4, 2011 at 9:21 AM, Satish Patel satish...@hotmail.com
satish...@hotmail.com wrote:
Did you try digim fax ?
Also you can record you incoming fax via mxmonitor and analize it.
--
Sent from my iPhone
On May 4, 2011, at 8:50 AM, vip killa vipki...@gmail.com
vipki...@gmail.com wrote:
I've given up on trying T38 because there is no universal support for
it... Can someone recommend another way of faxing without using T38?
On Tue, May 3, 2011 at 5:13 PM, satish patel
satish...@hotmail.comsatish...@hotmail.com
satish...@hotmail.com wrote:
Enable debug and verbose on CLI ?
Did you enable and also at logger.conf
full = notice,warning,error,debug,verbose,dtmf,fax
--
Date: Tue, 3 May 2011 16:12:06 -0400
From: vipki...@gmail.com vipki...@gmail.comvipki...@gmail.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have full log.. only thing that stands out are two warnings:
[May 3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.
result=13: Unexpected message received.
[May 3 16:10:40] WARNING[18176] app_fax.c: Transmission failed
On Tue, May 3, 2011 at 4:05 PM, satish patel
satish...@hotmail.comsatish...@hotmail.com
satish...@hotmail.com wrote:
I'd enable full debug at logger.conf and try to find issue.
-S
--
Date: Tue, 3 May 2011 15:55:51 -0400
From: vipki...@gmail.com vipki...@gmail.comvipki...@gmail.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
I tried with those settings and without... same error:
WARNING[18090]: app_fax.c:820 transmit: Transmission failed
On Tue, May 3, 2011 at 3:32 PM, satish patel
satish...@hotmail.comsatish...@hotmail.com
satish...@hotmail.com wrote:
did you set faxdetect=both or incoming
and faxbuffer=?
-S
--
Date: Tue, 3 May 2011 15:28:36 -0400
From: vipki...@gmail.com vipki...@gmail.comvipki...@gmail.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed
when trying to fax from a POTS line...
On Tue, May 3, 2011 at 3:27 PM, satish patel
satish...@hotmail.comsatish...@hotmail.com
satish...@hotmail.com wrote:
You need spandsp i guess following is my dialplan is working example
[fax]
exten = 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten = 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten = 9000,n,ReceiveFax(${FAXFILE})
exten = 9000,n,Hangup()
--
Date: Tue, 3 May 2011 15:20:33 -0400
From: vipki...@gmail.com vipki...@gmail.comvipki...@gmail.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
asterisk-users@lists.digium.com
Subject: [asterisk-users] receive faxes
does anybody know a good tutorial on how to setup asterisk to receive faxes
(no need to send them) ? i've tried using app_fax.so with T38 but i keep
getting Transmission failed
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*[Danny Nicholas] *
*You are “Running before you learn to walk”! You can’t make T.38 work
(that’s ok, most other folks can’t either) but you want a free faxing
solution that does multiple channels. Get the Free license and make that
work, then pay Digium the $10
Re: [asterisk-users] receive faxes
Why use T.38 when you can use ulaw ? On Wed, May 4, 2011 at 10:12 AM, Eric Wieling ewiel...@nyigc.com wrote: Does Hylafax support T.38? The free fax works just fine with DAHDI. I've never tried to do T.38 with that since it seems like it would be complicated and not give me much over using DAHDI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, May 04, 2011 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] receive faxes screw that i just got hylafax to work with IAXMODEM...i refuse to pay digium a dime... supposed to be open-source right? -- _ -- 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] receive faxes
Honestly Digium's Asterisk is not a quality project. Though it has lead the way in innovative open-source VoIP, it's a flawed and chaotic project. Hence, I refuse to pay Digium. Digium seems to make a bazillion dollars off of these flaws by selling commercial support/addons anyway... so that should be worth some bad karma points. On Wed, May 4, 2011 at 10:42 AM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 4 May 2011, vip killa wrote: screw that i just got hylafax to work with IAXMODEM...i refuse to pay digium a dime... supposed to be open-source right? Great attitude. Should be worth about a bazillion bad karma points. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] receive faxes
On 4 May 2011, at 16:02, vip killa wrote: Honestly Digium's Asterisk is not a quality project. Though it has lead the way in innovative open-source VoIP, it's a flawed and chaotic project. Hence, I refuse to pay Digium. Digium seems to make a bazillion dollars off of these flaws by selling commercial support/addons anyway... so that should be worth some bad karma points. Don't use Asterisk then. I personally, I would never choose to use Asterisk -- _ -- 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] receive faxes
On Wed, May 4, 2011 at 11:13 AM, Danny Nicholas da...@debsinc.com wrote: Non-T.38 faxing works reasonably well. I have some issues with some things at Digium as well, but I'm not going to bite the hand that feeds me. I assume that Digium takes most of the known bugs out of what they charge folks for. From what I read, if you had to make a living on T.38 faxing, you'd be in the hut in Kenya by Obama's brother. As for Karma, we'll see.. Asterisk doesn't pay me and never will, they only make my job more difficult because it's been in place here for so long... it would be nearly impossible to get away from it. BTW, I'm using iaxmodem with hylafax...works much better than T.38. Obama's brother... damn that ain't right... that's worth some bad karma points -- _ -- 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] receive faxes
does anybody know a good tutorial on how to setup asterisk to receive faxes (no need to send them) ? i've tried using app_fax.so with T38 but i keep getting Transmission failed -- _ -- 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] receive faxes
i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed
when trying to fax from a POTS line...
On Tue, May 3, 2011 at 3:27 PM, satish patel satish...@hotmail.com wrote:
You need spandsp i guess following is my dialplan is working example
[fax]
exten = 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten = 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten = 9000,n,ReceiveFax(${FAXFILE})
exten = 9000,n,Hangup()
--
Date: Tue, 3 May 2011 15:20:33 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] receive faxes
does anybody know a good tutorial on how to setup asterisk to receive faxes
(no need to send them) ? i've tried using app_fax.so with T38 but i keep
getting Transmission failed
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or update options visit:
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Re: [asterisk-users] receive faxes
I tried with those settings and without... same error:
WARNING[18090]: app_fax.c:820 transmit: Transmission failed
On Tue, May 3, 2011 at 3:32 PM, satish patel satish...@hotmail.com wrote:
did you set faxdetect=both or incoming
and faxbuffer=?
-S
--
Date: Tue, 3 May 2011 15:28:36 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed
when trying to fax from a POTS line...
On Tue, May 3, 2011 at 3:27 PM, satish patel satish...@hotmail.comwrote:
You need spandsp i guess following is my dialplan is working example
[fax]
exten = 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten = 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten = 9000,n,ReceiveFax(${FAXFILE})
exten = 9000,n,Hangup()
--
Date: Tue, 3 May 2011 15:20:33 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] receive faxes
does anybody know a good tutorial on how to setup asterisk to receive faxes
(no need to send them) ? i've tried using app_fax.so with T38 but i keep
getting Transmission failed
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Re: [asterisk-users] receive faxes
i have full log.. only thing that stands out are two warnings:
[May 3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.
result=13: Unexpected message received.
[May 3 16:10:40] WARNING[18176] app_fax.c: Transmission failed
On Tue, May 3, 2011 at 4:05 PM, satish patel satish...@hotmail.com wrote:
I'd enable full debug at logger.conf and try to find issue.
-S
--
Date: Tue, 3 May 2011 15:55:51 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
I tried with those settings and without... same error:
WARNING[18090]: app_fax.c:820 transmit: Transmission failed
On Tue, May 3, 2011 at 3:32 PM, satish patel satish...@hotmail.comwrote:
did you set faxdetect=both or incoming
and faxbuffer=?
-S
--
Date: Tue, 3 May 2011 15:28:36 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed
when trying to fax from a POTS line...
On Tue, May 3, 2011 at 3:27 PM, satish patel satish...@hotmail.comwrote:
You need spandsp i guess following is my dialplan is working example
[fax]
exten = 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten = 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten = 9000,n,ReceiveFax(${FAXFILE})
exten = 9000,n,Hangup()
--
Date: Tue, 3 May 2011 15:20:33 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] receive faxes
does anybody know a good tutorial on how to setup asterisk to receive faxes
(no need to send them) ? i've tried using app_fax.so with T38 but i keep
getting Transmission failed
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[asterisk-users] music on hold skipping
For some reason our music on hold is intermittently skipping... running Asterisk 1.6.1.22 anybody know what could be causing this? I don't think it's an encoding problem because it plays fine sometimes. -- _ -- 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] Multiple Asterisk
could you send me book? On Fri, Apr 29, 2011 at 9:48 AM, satish patel satish...@hotmail.com wrote: I have sent you book in PM. -S -- Date: Fri, 29 Apr 2011 10:39:56 -0300 From: sf.ri...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple Asterisk Thanks! Would apreciate the book! But i am already researching []'sf.rique On Fri, Apr 29, 2011 at 10:10 AM, Satish Patel satish...@hotmail.comwrote: Don't expect lots of thing because I have just post my basic config and method to integrate openser with asterisk and I did that 3 year ago. http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration. I would say search on google today lots of material are there and I have remembered there is a nice book regarding this. I guess I have PDF version of that book I will search and try to find. -- Sent from my iPhone On Apr 29, 2011, at 8:40 AM, Henrique Fernandes sf.ri...@gmail.com wrote: Can you post later t he link for it ? I read alot that page. []'sf.rique On Fri, Apr 29, 2011 at 9:35 AM, Satish Patel satish...@hotmail.com satish...@hotmail.com wrote: True, we had setup before openser with asterisk and it works great. I have wrote small document on voip-info related my project. -- Sent from my iPhone On Apr 29, 2011, at 8:23 AM, Henrique Fernandes sf.ri...@gmail.com sf.ri...@gmail.com wrote: Thnaks a Lot. So i will look for openser integration with asterisk! []'sf.rique On Fri, Apr 29, 2011 at 3:12 AM, RAJNIKANT VANZA rajniva...@gmail.comrajniva...@gmail.com rajniva...@gmail.com wrote: Hi, If u want to setup for 4500 or more phone then better to user OpenSER + Asterisk. OpenSER easily work for 10,000 calls. You need to setup one server for OpenSER and all phone register on this server. You need to write routing logic in OpenSER server to call connect and if u need to play media then forward to destination asterisk server. 1 OpenSER server + Asterisk server for each location. -- Best Regards, Rajnikant Vanza Software Engineer --- Working On Linux,C/C++,VoIP,Asterisk Technology On Tue, Mar 22, 2011 at 10:47 PM, Henrique Fernandes sf.ri...@gmail.comsf.ri...@gmail.com sf.ri...@gmail.com wrote: No one ? Other thing, i was reading about asterisk realtime, it can be configured to have multiple asterisk conectted to the same database? But how would it know in wich host are the number?? Thanks! []'sf.rique On Wed, Mar 16, 2011 at 5:41 PM, Henrique Fernandes sf.ri...@gmail.comsf.ri...@gmail.com sf.ri...@gmail.com wrote: I am reading about, and some people are saying that openser is better for biger envoriments, and dundi is fine for smal envoriments, does anyone have any info about it ? We have now about 4500 convencional phones and we gonna expand a lot. So, OpenSER vs DUNDi ? I guess i will use Asterisk RealTime also right ? []'sf.rique On Wed, Mar 16, 2011 at 6:20 AM, Rizwan Hisham rizwanhas...@gmail.comrizwanhas...@gmail.com rizwanhas...@gmail.com wrote: Here is a better link for DUNDi http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/ http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/ skip the part which you know already On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes sf.ri...@gmail.comsf.ri...@gmail.com sf.ri...@gmail.com wrote: []'sf.rique On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger pabelan...@digium.compabelan...@digium.com pabelan...@digium.com wrote: On 11-03-15 06:19 PM, Henrique Fernandes wrote: Have many diferenet locations that have convencional phones that need to call others locations with convencional phones. And we can not change this, I was reading and asterisk cannot handle it self this kind of setup, it needs an separated serrver to control and routers the calls to this poins right ? So can you guys give any help ? I guess asterisk with SER could do the job ? I don't believe SER will help you in the setup (see below). So my question is how do i make the 2 PABX with asterisk talk to each other? Do i need only 2 asterisk with digium or i need one server with SER to maki it happen ? There is another program that does what i am looking for ? If you require local hardware for each site, then you can install Asterisk at each location. You can then interconnect them using IAX2 or SIP, additionally you can use DUNDi in your dialplans to share information before the Asterisk boxes. Thanks! I had heard some thing about DUNDi but now i am reading i guess it is what i need! I am guessing i can use both IAX2 and SIP i read something about H.323 So i am gonna see which one is
[asterisk-users] asterisk practices
I just completed building a feature rich asterisk voicemail system using perl, php, and mysql. My only concern is that the system i built will not be able to handle the call volume needed. Let me start by explaining my setup. Incoming call - route.agi (perl - mysql lookup) - AGI - voicemailbox (using mysql odbc) or terminate with wrong number message if a message is left in a voicemailbox the following happens: externnotify - notify.pl (perl - mysql lookup) - up to 2 calls originated (using AMI), up to 4 emails sent out (with up to 2 attachemnts of voicemail) this system may need to handle up to 50 concurrent calls. the notify.pl script may be called several times a second. My question is, will asterisk be able to handle calling the notify.pl script that many times? or is there a better way to handle large volumes of voicemail notification, thank you in advance for your input. -- _ -- 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] send voicemail to multiple emails
I understand now the need for externnotify to run on vm check is to update MWI. But I agree with M Hulber, an extra variable would be nice to tell the script why externnotify was called... On Wed, Apr 13, 2011 at 6:32 AM, M Hulber asterisk.ad...@hulber.com wrote: Instead of picking from multiple scripts, send the action to the script in a variable like the dial status -- _ -- 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] send voicemail to multiple emails
I already am using 'distribution groups', the point is I want 2 or more emails to be sent per mailbox... I've already figured out how i'm going to do this... i ended up modify app_voicemail.c to not run externnotify when someone checks their voicemail, it now only runs when a new message is left... then externnotify will send out the emails and other notifications... On Tue, Apr 12, 2011 at 4:30 AM, Andrew Thomas a...@datavox.co.uk wrote: So why not simply go back to square one and create a ‘distribution group’ e-mail address – and send to that? You’ve probably realised by now that if you want * to do something it doesn’t already do – you have to write that bit yourself. Good luck. -- _ -- 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] send voicemail to multiple emails
We are talking about mailcmd not externnotify I am aware of extennotify, problem is, it runs script when someone checks their voicemail, i need a script to run only when a voicemail is left On Mon, Apr 11, 2011 at 6:32 AM, Andrew Thomas a...@datavox.co.uk wrote: Not quite true. I use a PHP script to do my processing (called from voicemail.conf [externnotify = /usr/local/bin/vmnotify.php]). The main three lines are: $vm_context = $argv[1]; $extension = $argv[2]; $number_of_messages = $argv[3]; Self explanatory really. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: 10 April 2011 05:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] send voicemail to multiple emails I've already taken the steps you described...issue i ran into was there is no variables passed to mailcmd only STDIN... as a result i have to extract variables from STDIN... On Fri, Apr 8, 2011 at 5:09 PM, Warren Selby wcse...@selbytech.com wrote: On Fri, Apr 8, 2011 at 1:18 PM, vip killa vipki...@gmail.com wrote: That does not sound easy... besides these email addresses would be taken from a MySQL database. It's actually what you're going to end up doing, whether you do it on the MTA level or your code it into your script that you execute instead of sendmail -f. Currently, there is no way to natively have asterisk send one voicemail to multiple email addresses. What's probably going to work best for you since you seem to like program your own scripts (and I'm not talking an AGI here, I'm talking either pure bash, php, perl, or whichever you prefer), is to change the mailcmd= option inside voicemail.conf and replace it with a script of your own design. I'm not sure off the top of my head which variables are passed to the command, but you could always write a simple script that just outputs all arguments to see and go from there. My guess is you're going to at the least get the preconfigured email address and the contents of your emailsubject and emailbody options (both of which have the option of passing multiple useful variables). -- Thanks, --Warren Selby, dCAP http://www.selbytech.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 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 -- _ -- 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] voicemail odbc Length is .....
I'm using voicemail ODBC with Asterisk 1.6.2.17.2. Why do I see Length is 186545 or something similar but a different number in Asterisk CLI everytime someone leaves a message? -- _ -- 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] voicemail odbc Length is .....
indeed but why in console and the info is so limited, it doesn't say which message or anything...strange On Mon, Apr 11, 2011 at 10:31 AM, Steven Howes steve-li...@geekinter.netwrote: On 11 Apr 2011, at 15:28, vip killa wrote: I'm using voicemail ODBC with Asterisk 1.6.2.17.2. Why do I see Length is 186545 or something similar but a different number in Asterisk CLI everytime someone leaves a message? Because not all messages are the same length. I'd guess it's length in bytes?.. 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] voicemail odbc Length is .....
Apologies you were correct, i had debug on... Sorry... On Mon, Apr 11, 2011 at 10:45 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Monday, April 11, 2011 9:44 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] voicemail odbc Length is . indeed but why in console and the info is so limited, it doesn't say which message or anything...strange On Mon, Apr 11, 2011 at 10:31 AM, Steven Howes steve-li...@geekinter.net wrote: On 11 Apr 2011, at 15:28, vip killa wrote: I'm using voicemail ODBC with Asterisk 1.6.2.17.2. Why do I see Length is 186545 or something similar but a different number in Asterisk CLI everytime someone leaves a message? Because not all messages are the same length. I'd guess it's length in bytes?.. S *[Danny Nicholas]* *WAG – you’ve got a high debug level on somewhere.* -- _ -- 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] send voicemail to multiple emails
Would be so much simpler if mailcmd acted just like externnotify or externnotify only ran when a message was left but not when someone checks their voicemail... That's pretty much where you're at. What gets passed to STDIN is an email, it's not set up for use by a script. Remember, what you're doing is asking Asterisk to do something out of the ordinary, hence why having an email alias is typically the best solution. -- _ -- 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] send voicemail to multiple emails
I'm not confused about this... Everytime a voicemail is left, I need asterisk to run a script that will query a database, and according to those results perform various actions. These actions include calling a number and connecting it directly to voicemailmain and/or sending out multiple emails... I'm able to accomplish this using mailcmd, reading the STDIN, parsing out what i need such as the mailbox but I'm not sure how stable this method is I'd rather use externnotify because it is designed to a run script On Mon, Apr 11, 2011 at 1:34 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On 4/11/2011 12:30 PM, vip killa wrote: Would be so much simpler if mailcmd acted just like externnotify or externnotify only ran when a message was left but not when someone checks their voicemail... That's pretty much where you're at. What gets passed to STDIN is an email, it's not set up for use by a script. Remember, what you're doing is asking Asterisk to do something out of the ordinary, hence why having an email alias is typically the best solution. -- _ -- 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 I think you're confused about how it works. When someone leaves a voicemail, Asterisk sends a notification email, by piping an email to the defined mailcmd (by default, again, 'sendmail -t')...which is exactly what you're wishing it didmaybe you need to do some more reading? Why do you think it works in a different way? -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- 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] send voicemail to multiple emails
Anyway, i figured out how to accomplish this using externnotify... In app_voicemail.c, in the function vm_execmain i commented out run_externnotify(vmu-context, vmu-mailbox, NULL); Now externnotify is called by asterisk only when there is a new message and not when someone checks their voicemail... -- _ -- 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] send voicemail to multiple emails
I've already taken the steps you described...issue i ran into was there is no variables passed to mailcmd only STDIN... as a result i have to extract variables from STDIN... On Fri, Apr 8, 2011 at 5:09 PM, Warren Selby wcse...@selbytech.com wrote: On Fri, Apr 8, 2011 at 1:18 PM, vip killa vipki...@gmail.com wrote: That does not sound easy... besides these email addresses would be taken from a MySQL database. It's actually what you're going to end up doing, whether you do it on the MTA level or your code it into your script that you execute instead of sendmail -f. Currently, there is no way to natively have asterisk send one voicemail to multiple email addresses. What's probably going to work best for you since you seem to like program your own scripts (and I'm not talking an AGI here, I'm talking either pure bash, php, perl, or whichever you prefer), is to change the mailcmd= option inside voicemail.conf and replace it with a script of your own design. I'm not sure off the top of my head which variables are passed to the command, but you could always write a simple script that just outputs all arguments to see and go from there. My guess is you're going to at the least get the preconfigured email address and the contents of your emailsubject and emailbody options (both of which have the option of passing multiple useful variables). -- Thanks, --Warren Selby, dCAP http://www.selbytech.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] asterisk login to voicemail
Wow, thanks, that worked...
in case anyone is interested this is what i did
[voicemail]
exten = a,1,VoiceMailMain(${MAILBOXID}@${MAILBOXCONTEXT},p)
in AGI...
$AGI-set_variable(MAILBOXID, $options);
$AGI-set_variable(MAILBOXCONTEXT,4);
$AGI-set_context(voicemail);
$AGI-exec(VoiceMail, $options);
now the question is how to I get the VoiceMailMain to not ask for Mailbox
and already know which mailbox and just prompt for Password
On Thu, Apr 7, 2011 at 6:44 PM, Dan Journo d...@keshercommunications.comwrote:
Unfortunately, that solution will not work for me... The user must be
able to hit * during the greeting of any voicemail and be prompted for the
Password to that particular mailbox looks like i got a lot of
programming to do to create a work around for this... thanks for your
help...
Forgive me if i'm wrong, but you guys seem to be over complicating things.
Taken from: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
during the prompt if the caller presses:
'*' - the call jumps to extension 'a' in the current voicemail context.
*Example:*
Exten = a, 1, VoicemailMain(@default)
Exten = a, 2, Hangup
When using the star '*' it's important to note that the context you placed
the application voicemail in is irrelevant, it's the context for the
voicemail box that we're looking for in the dialplan for the jump to the 'a'
extension.
So this is what i do...
Before passing the call to the voicemail app, i set ${MAILBOXCONTEXT} to
the correct context, and i set ${MAILBOXID} to the mailbox name.
Then, in extensions.conf, I added this:-
[voicemail]
exten = a,1,Playback(astcc-please-enter-your)
exten = a,n,VoicemailMain(${MAILBOXID}@${MAILBOXCONTEXT})
When the user presses *, they are passed to the 'a' extension above and
into VoicemailMain.
I'm sure you can turn this into AGI easily enough if needed.
Dan Journo
Kesher Communications (UK)
Business Phone Systems http://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
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Re: [asterisk-users] asterisk login to voicemail
SIP/8.224.32.2-AGI Rx SET VARIABLE MAILBOXID 7167435000
SIP/8.224.32.2-AGI Tx 200 result=1
SIP/8.224.32.2-AGI Rx SET VARIABLE MAILBOXCONTEXT 4
SIP/8.224.32.2-AGI Tx 200 result=1
SIP/8.224.32.2-AGI Rx SET CONTEXT voicemail
And the mailbox 7167435000 does exist
On Fri, Apr 8, 2011 at 8:29 AM, Dan Journo d...@keshercommunications.comwrote:
now the question is how to I get the VoiceMailMain to not ask for
Mailbox and already know which mailbox and just prompt for Password
If its asking you for the mailbox, then the ${MAILBOXID}@${MAILBOXCONTEXT}
values aren't correct.
Use the CLI to debug and make sure ${MAILBOXID}@${MAILBOXCONTEXT} is
correct and the mailbox exists.
Dan Journo
Kesher Communications (UK)
Business Phone Systems http://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
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Re: [asterisk-users] asterisk login to voicemail
can you explain how this can be done simpler?
On Fri, Apr 8, 2011 at 10:10 AM, Satish Patel satish...@hotmail.com wrote:
Why are you using agi for this ? They are inbuild features of asterisk.
Or may be I am missing something
--
Sent from my iPhone
On Apr 8, 2011, at 8:26 AM, vip killa vipki...@gmail.com wrote:
Wow, thanks, that worked...
in case anyone is interested this is what i did
[voicemail]
exten = a,1,VoiceMailMain(${MAILBOXID}@${MAILBOXCONTEXT},p)
in AGI...
$AGI-set_variable(MAILBOXID, $options);
$AGI-set_variable(MAILBOXCONTEXT,4);
$AGI-set_context(voicemail);
$AGI-exec(VoiceMail, $options);
now the question is how to I get the VoiceMailMain to not ask for Mailbox
and already know which mailbox and just prompt for Password
On Thu, Apr 7, 2011 at 6:44 PM, Dan Journo d...@keshercommunications.com
d...@keshercommunications.com wrote:
Unfortunately, that solution will not work for me... The user must be
able to hit * during the greeting of any voicemail and be prompted for the
Password to that particular mailbox looks like i got a lot of
programming to do to create a work around for this... thanks for your
help...
Forgive me if i'm wrong, but you guys seem to be over complicating things.
Taken from: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
during the prompt if the caller presses:
'*' - the call jumps to extension 'a' in the current voicemail context.
*Example:*
Exten = a, 1, VoicemailMain(@default)
Exten = a, 2, Hangup
When using the star '*' it's important to note that the context you placed
the application voicemail in is irrelevant, it's the context for the
voicemail box that we're looking for in the dialplan for the jump to the 'a'
extension.
So this is what i do...
Before passing the call to the voicemail app, i set ${MAILBOXCONTEXT} to
the correct context, and i set ${MAILBOXID} to the mailbox name.
Then, in extensions.conf, I added this:-
[voicemail]
exten = a,1,Playback(astcc-please-enter-your)
exten = a,n,VoicemailMain(${MAILBOXID}@${MAILBOXCONTEXT})
When the user presses *, they are passed to the 'a' extension above and
into VoicemailMain.
I'm sure you can turn this into AGI easily enough if needed.
Dan Journo
Kesher Communications (UK)
Business Phone Systems http://www.keshercommunications.com/ | Hosted
PBX http://www.keshercommunications.com/hostedpbx.html
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[asterisk-users] send voicemail to multiple emails
Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -- _ -- 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] send voicemail to multiple emails
That does not sound easy... besides these email addresses would be taken from a MySQL database. The easiest way would be to set up an alias in your MTA configuration. That way, you could configure the mailbox for the alias email address and copies would be sent to all addresses in the alias definition. -- _ -- 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 login to voicemail
Is there a way to login to a voicemail box when someone pushes '#' during greeting? -- _ -- 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] asterisk login to voicemail
i'm afraid my setup is more complex than that [inbound] exten = _X.,1,agi(route.pl) after some logic using mysql, route.pl then does: $AGI-exec(VoiceMail, $options); at that point, I would like the caller to be able to push '#' and be prompted for Password for that particular mailbox On Thu, Apr 7, 2011 at 2:58 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Thursday, April 07, 2011 1:56 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] asterisk login to voicemail Is there a way to login to a voicemail box when someone pushes '#' during greeting? *[Danny Nicholas] * *Here is one way:* *[greeting]* *Exten = s,1,background(greeting)* *Exten =s,n,hangup* *Exten = #,1,voicemailmain(100@default)* -- _ -- 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] asterisk login to voicemail
I'm sorry I'm new to AGI programming but i did this:
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
and in extensions.conf i have:
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = #,n,VoiceMailMain(${EXTEN}@4)
I keep getting 603 declined when i call the number...
On Thu, Apr 7, 2011 at 3:09 PM, Danny Nicholas da...@debsinc.com wrote:
If you add the exten = #,1 line to the end of the inbound context, that
should do it for you. If not, change $AGI-exec(“VoiceMail”,$options) to go
to a context instead of running Voicemail directly.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 2:04 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
i'm afraid my setup is more complex than that
[inbound]
exten = _X.,1,agi(route.pl)
after some logic using mysql, route.pl then does:
$AGI-exec(VoiceMail, $options);
at that point, I would like the caller to be able to push '#' and be
prompted for Password for that particular mailbox
On Thu, Apr 7, 2011 at 2:58 PM, Danny Nicholas da...@debsinc.com wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 1:56 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] asterisk login to voicemail
Is there a way to login to a voicemail box when someone pushes '#' during
greeting?
*[Danny Nicholas] *
*Here is one way:*
*[greeting]*
*Exten = s,1,background(greeting)*
*Exten =s,n,hangup*
*Exten = #,1,voicemailmain(100@default)*
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Re: [asterisk-users] asterisk login to voicemail
Ok, i have this now...
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = *,1,VoiceMailMain(${callednum})
in AGI i have:
$AGI-set_variable(callednum, $options);
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
I'm getting a busy signal and this error
pbx.c:4413 __ast_pbx_run: Channel 'SIP/8.224.32.3-0002' sent into
invalid extension 'XX' in context 'voicemail', but no invalid
handler
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Re: [asterisk-users] asterisk login to voicemail
Actually the mailbox is 7167435000...
in this case the two variables are the same and the mailbox 7167435000 does
exist
I want someone to call 7167435000 reach mailbox 7167435000 (VoiceMail), if
they push * during greeting, then i went them to prompted for a PIN for that
mailbox (VoiceMailMain)
I'm totally lost as to how to get this done... any suggestions?
On Thu, Apr 7, 2011 at 3:47 PM, Danny Nicholas da...@debsinc.com wrote:
As I see it, callednum and vmbox should not be the same. Vmbox is a
“good” mailbox you’re going to reach if the user doesn’t hit #, callednum is
the “fallback” number that you are going to use and should be an established
mailbox (3-4 digits) not a full number (10 digits you have indicated).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 2:41 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
Ok, i have this now...
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = *,1,VoiceMailMain(${callednum})
in AGI i have:
$AGI-set_variable(callednum, $options);
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
I'm getting a busy signal and this error
pbx.c:4413 __ast_pbx_run: Channel 'SIP/8.224.32.3-0002' sent into
invalid extension 'XX' in context 'voicemail', but no invalid
handler
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Re: [asterisk-users] asterisk login to voicemail
Unfortunately, that solution will not work for me... The user must be able
to hit * during the greeting of any voicemail and be prompted for the
Password to that particular mailbox looks like i got a lot of
programming to do to create a work around for this... thanks for your
help...
On Thu, Apr 7, 2011 at 4:18 PM, Danny Nicholas da...@debsinc.com wrote:
Here’s your solution
[vmtest]
exten = s,1,background(vm-Family,3)
exten = s,n,waitexten(3)
exten = s,n,Voicemail(${callnum}@default)
exten = *,1,VoicemailMain(${callnum}@default)
exten = #,1,VoicemailMain(${callnum}@default)
exten = i,1,Voicemail(${callnum}@default)
exten = t,1,Voicemail(${callnum}@default)
vm-family plays when you come in. you have 3 seconds to hit * or #. If
not, you go to regular voicemail. If so, you go to admin and get prompted
for the password.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 2:52 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
Actually the mailbox is 7167435000...
in this case the two variables are the same and the mailbox 7167435000 does
exist
I want someone to call 7167435000 reach mailbox 7167435000 (VoiceMail), if
they push * during greeting, then i went them to prompted for a PIN for that
mailbox (VoiceMailMain)
I'm totally lost as to how to get this done... any suggestions?
On Thu, Apr 7, 2011 at 3:47 PM, Danny Nicholas da...@debsinc.com wrote:
As I see it, callednum and vmbox should not be the same. Vmbox is a “good”
mailbox you’re going to reach if the user doesn’t hit #, callednum is the
“fallback” number that you are going to use and should be an established
mailbox (3-4 digits) not a full number (10 digits you have indicated).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 2:41 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
Ok, i have this now...
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = *,1,VoiceMailMain(${callednum})
in AGI i have:
$AGI-set_variable(callednum, $options);
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
I'm getting a busy signal and this error
pbx.c:4413 __ast_pbx_run: Channel 'SIP/8.224.32.3-0002' sent into
invalid extension 'XX' in context 'voicemail', but no invalid
handler
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Re: [asterisk-users] asterisk login to voicemail
Indeed, that is what i would do except many users will not have a greeting.
so those without a greeting will not be able to login unless i generate a
canned greeting which i think i will have to do.
On Thu, Apr 7, 2011 at 4:33 PM, Danny Nicholas da...@debsinc.com wrote:
One more thought – assuming that your users all have greetings recorded,
you could change vm-family to
/var/spoo/asterisk/voicemail/default/${callnum}/busy and keep the same plan.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 3:25 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
Unfortunately, that solution will not work for me... The user must be able
to hit * during the greeting of any voicemail and be prompted for the
Password to that particular mailbox looks like i got a lot of
programming to do to create a work around for this... thanks for your
help...
On Thu, Apr 7, 2011 at 4:18 PM, Danny Nicholas da...@debsinc.com wrote:
Here’s your solution
[vmtest]
exten = s,1,background(vm-Family,3)
exten = s,n,waitexten(3)
exten = s,n,Voicemail(${callnum}@default)
exten = *,1,VoicemailMain(${callnum}@default)
exten = #,1,VoicemailMain(${callnum}@default)
exten = i,1,Voicemail(${callnum}@default)
exten = t,1,Voicemail(${callnum}@default)
vm-family plays when you come in. you have 3 seconds to hit * or #. If
not, you go to regular voicemail. If so, you go to admin and get prompted
for the password.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 2:52 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
Actually the mailbox is 7167435000...
in this case the two variables are the same and the mailbox 7167435000 does
exist
I want someone to call 7167435000 reach mailbox 7167435000 (VoiceMail), if
they push * during greeting, then i went them to prompted for a PIN for that
mailbox (VoiceMailMain)
I'm totally lost as to how to get this done... any suggestions?
On Thu, Apr 7, 2011 at 3:47 PM, Danny Nicholas da...@debsinc.com wrote:
As I see it, callednum and vmbox should not be the same. Vmbox is a “good”
mailbox you’re going to reach if the user doesn’t hit #, callednum is the
“fallback” number that you are going to use and should be an established
mailbox (3-4 digits) not a full number (10 digits you have indicated).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 2:41 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
Ok, i have this now...
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = *,1,VoiceMailMain(${callednum})
in AGI i have:
$AGI-set_variable(callednum, $options);
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
I'm getting a busy signal and this error
pbx.c:4413 __ast_pbx_run: Channel 'SIP/8.224.32.3-0002' sent into
invalid extension 'XX' in context 'voicemail', but no invalid
handler
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asterisk
Re: [asterisk-users] agi voicemail callback
What about a executing an AGI script with: [general] externnotify = /some_agi_script.agi Would that work? On Wed, Apr 6, 2011 at 3:20 AM, Thorsten Göllner t...@ovm-group.com wrote: Am 05.04.2011 18:50, schrieb vip killa: I'm wondering if there is a simply way to perform a voicemail callback feature using AGI. For instance, a caller leaves a voicemail, the voicemail will then call the owner of the voicemailbox determined by a database look up. One possibility: look via cron job, if there is a new message and if so, you can drop a call file in /var/spool/asterisk/outgoing. -- _ -- 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] agi voicemail callback
are you using Asterisk::AMIhttp://search.cpan.org/~greenbean/Asterisk-AMI/lib/Asterisk/AMI.pm for this script? On Wed, Apr 6, 2011 at 10:04 AM, Danny Nicholas da...@debsinc.com wrote: Yes – I do it that way because I run the module this is included in on about 10 different Asterisk servers. -- _ -- 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] voicemail call back loop
I have externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl so that when someone is left a voicemail it will call the person's mobile phone and prompt them with the new message. The perl script simply originates a call to a persons mobile phone and connects it to their voicemail using VoiceMailMain. Problem is when user hangs up from checking their messages, it runs the externnotify again causing an infinite loop. Has anybody encountered this problem or is there an option to not have it run externnotify after checking messages? -- _ -- 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] voicemail call back loop
What happens if there are more than 1 message and the user does not listen to all messages though? On Wed, Apr 6, 2011 at 1:00 PM, Steven Howes steve-li...@geekinter.netwrote: On 6 Apr 2011, at 17:46, vip killa wrote: I have externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl so that when someone is left a voicemail it will call the person's mobile phone and prompt them with the new message. The perl script simply originates a call to a persons mobile phone and connects it to their voicemail using VoiceMailMain. Problem is when user hangs up from checking their messages, it runs the externnotify again causing an infinite loop. Has anybody encountered this problem or is there an option to not have it run externnotify after checking messages? Look at the docs. Externnotify sends mailbox + mailbox count. Make your script exit if it's 0 messages. 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] voicemail call back loop
does mailcmd send any variables or data to script? I need a way for script to identify which mailbox was left a message. On Wed, Apr 6, 2011 at 3:11 PM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 5 Apr 2011, Steve Edwards wrote: Use 'mailcmd' in voicemail.conf. On Wed, 6 Apr 2011, vip killa wrote: I have externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl so that when someone is left a voicemail it will call the person's mobile phone and prompt them with the new message. The perl script simply originates a call to a persons mobile phone and connects it to their voicemail using VoiceMailMain. Problem is when user hangs up from checking their messages, it runs the externnotify again causing an infinite loop. Has anybody encountered this problem or is there an option to not have it run externnotify after checking messages? Mailcmd? Also, storing programs that aren't AGIs in the AGI directory doesn't sound like a 'best practice' candidate to me. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] agi voicemail callback
I'm wondering if there is a simply way to perform a voicemail callback feature using AGI. For instance, a caller leaves a voicemail, the voicemail will then call the owner of the voicemailbox determined by a database look up. -- _ -- 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] agi create mailbox
Is it possible to create a voicemail box using AGI? How does asterisk know about mailboxes when using Asterisk with pure AGI? -- _ -- 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] call parking issues in asterisk 1.6.2.16.2
We have a problem of no MoH when parking calls running asterisk 1.6.2.16.2. Also, the parked call never goes back to the parker. We have comebacktoorigin = yes and parkingtime = 180 in features.conf Anybody know why this isn't working? -- _ -- 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] asterisk and fail2ban
Back to the original question, for those of you using Fail2Ban, Does it take an unusually high amount of break-in attempts before attackers are banned? I have it set to 5 attempts in fail2ban but usually, the attacker is able to make over 100 attempts before fail2ban bans them. I've tried this using asterisk's /var/log/asterisk/messages and /var/log/messages with same results. Perhaps someone else is experiencing this or has resolved it, thank you. On Thu, Mar 31, 2011 at 4:05 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Wed, 30 Mar 2011, Terry Brummell wrote: Yah, sounds simple, how do you set it up to do this? Fail2Ban was pretty easy, if it's that easy, why was F2B even created? It's easy for me because I read an undestand how things work, and deal with Linux firewalling in a daily basis. Fail2ban is an (almost) drop-in solution which requires minimal thinking - just a few lines in a config file to edit. (and python which I don't have installed on my systems) Gordon -- _ -- 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] asterisk and fail2ban
I'm afraid you are incorrect, fail2ban reads the log once every second. On Thu, Mar 31, 2011 at 8:52 AM, Terry Brummell te...@brummell.net wrote: Your delay is due to the amount of time the F2B script takes to read the log file, and due to how often it is called. I do not believe it is a realtime event. Say, every minute it's called to read the log and act. I'm not sure of the exact numbers, but you get the idea -- _ -- 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] asterisk and fail2ban
Yes, I see in the log that most of these attacks only last 2 seconds before fail2ban bans them On Thu, Mar 31, 2011 at 11:13 AM, Warren Selby wcse...@selbytech.comwrote: On Thu, Mar 31, 2011 at 7:17 AM, vip killa vipki...@gmail.com wrote: Back to the original question, for those of you using Fail2Ban, Does it take an unusually high amount of break-in attempts before attackers are banned? I have it set to 5 attempts in fail2ban but usually, the attacker is able to make over 100 attempts before fail2ban bans them. I've tried this using asterisk's /var/log/asterisk/messages and /var/log/messages with same results. Perhaps someone else is experiencing this or has resolved it, thank you. Check your log files. With the current generation of SIP attack scripts, I've seen hundreds of attacks come in within one second, especially if you've got decent bandwidth. I've seen fail2ban logs that state between 60-250 failed attempts for asterisk. I think it's just the nature of the speed of the attacks. -- Thanks, --Warren Selby, dCAP http://www.selbytech.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] asterisk and fail2ban
so does anyone use fail2ban w/ asterisk or most people use sshguard? On Wed, Mar 30, 2011 at 6:57 AM, Andrew Thomas a...@datavox.co.uk wrote: Just to respond to the IP range approach. My ISP recently changed my external IP and now it appears that I am in New York (when I am actually static in Manchester, England). I've also been in Birmingham, Motherwell and Nottingham [UK] aswell! So, although banning certain ranges may be a good idea for you - it's not a good idea for everyone (we have 'road warriors' that do, indeed, travel to the Far East and Middle East). I suppose the only 'real' way to invoke security (on any system) is to have very strong passwords - maybe 1234 is not the way to go :p -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: 30 March 2011 10:08 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk and fail2ban On Wed, 30 Mar 2011 01:45:20 +0300, Ioan Indreias indre...@gmail.com wrote: Just to provide an alternative to sshguard: you could use BFD[1] Thanks Ioan. I'll give it a shot. -- _ -- 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 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 -- _ -- 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] asterisk and fail2ban
could you please elaborate on how you have iptables setup to work that way? On Wed, Mar 30, 2011 at 4:11 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Wed, 30 Mar 2011, Terry Brummell wrote: I think you will find Fail2Ban the defacto standard. I don't use fai2ban. Never have, never will because I simply don't need it. Standard iptables are good enough if you can be bothered to use them to their full abilities. No need for anything else as iptables can do connection tracking and blocking against time - just like fail2ban does. More than X connections a second/minute/hour from a given IP address? Yes, iptables can detect and block that. Works for all protocolls too - SIP, IAX, POP, SSH, etc. Gordon -- _ -- 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] asterisk and fail2ban
Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in maxretry in jail.conf For example, I get an email saying: The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts against ASTERISK. when maxretry = 5 in jail.conf Perhaps someone else is experiencing this or has resolved it, thank you in advance for your time. -- _ -- 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] asterisk and fail2ban
Yes I followed directions on that page Running Asterisk 1.6.1.22, anybody else experiencing this? On Mon, Mar 28, 2011 at 8:32 AM, Andrew Latham lath...@gmail.com wrote: On Mon, Mar 28, 2011 at 9:20 AM, vip killa vipki...@gmail.com wrote: Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in maxretry in jail.conf For example, I get an email saying: The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts against ASTERISK. when maxretry = 5 in jail.conf Perhaps someone else is experiencing this or has resolved it, thank you in advance for your time. If you fixed the logging issue discussed here http://www.fail2ban.org/wiki/index.php/Asterisk then I would assume your logging has problems. -- ~~~ Andrew lathama Latham lath...@gmail.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
