Re: [asterisk-users] problem with crashing Asterisk 1.8
Hmm disabled Woomera and everything seems stable. Strange! On Thu, Mar 10, 2011 at 11:46 AM, Peter den Hartog wrote: > 1.8.0 :-), Nothing fancy just simple dialing/trunking. > > > On Thu, Mar 10, 2011 at 11:31 AM, --[ UxBoD ]-- wrote: > >> >> -- >> >> My Asterisk 1.8 (with Dahdi/Wanrouter) is crashing every minute or 2. It >> just keeps restarting. >> Any pointers on log files to watch? I tried to debug it but i couldn't >> find a good reason for the crashes. >> Maby the box is just overloaded or something like that but there should be >> a log file telling me that, right? >> >> Thanks, >> Peter >> >> -- >> _ >> -- 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 >> >> Which version of 1.8 are you using ? If you are using call pickup that can >> generate a segfault and crash Asterisk in version 1.8.3. Am hoping 1.8.4 >> will be out soon. >> -- >> Thanks, Phil >> >> -- >> _ >> -- 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] Cisco 7942G IP Phone firmware conversion from SCCP to SIP.
Thanks, Smith. Even keeping the file empty/ touch did not help. Not sure still what we are missing. --- On Tue, 3/8/11, Cassius Smith wrote: From: Cassius Smith Subject: Re: [asterisk-users] Cisco 7942G IP Phone firmware conversion from SCCP to SIP. To: "Asterisk Users Mailing List - Non-Commercial Discussion" Date: Tuesday, March 8, 2011, 12:50 PM Subject: [asterisk-users] Cisco 7942G IP Phone firmware conversion from SCCP to SIP. Hi, The current SCCP image on the 7942 phone is :SCCP42.9-0-2SR1S. We are trying to convert/upgrade the phone to SIP version of the firmware i.e : cmterm-7942_7962-sip.9-0-3 (Firmware is downloaded from the cisco support site). We have unzipped and placed all the files in the /tftp (root directory) of tftp server. Following files are also placed in the tftp directory. The Upgradation / Coversion is not taking place. (In the ethereal can see that the files are getting transferred without any error). Are we missing any other files in the /tftpdirectory? Or the information mentioned in the .xml.cnf / .tlv files incorrect? Your help in this regard is much appreciated. Regards, Srinivas I had best luck with the tlv files being 0 bytes. I.e. Touch the tlv files but leave them empty. HTH Cassius Smith -Inline Attachment Follows- -- _ -- 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] Automatically unpause a paused queue memeber - bad idea?
I have some cases when I want to pause a queue member and automatically unpause the queue member after a specified time. Right now I am doing it by a AMI script, but was thinking if it is possible to add a parameter to PauseQueueMember like, PauseQueueMember([queuename],interface[,options[,reason[,time]]]) where time will be how long (in seconds) the interface will be paused. before brought back. Maybe it is a bad idea, I dont know, what do you think? -- _ -- 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] Exceptionally long voice queue length in asterisk 1.6.2
Hi , I am using asterisk SVN-branch-1.6.2 version when i am making a call from SIP phone i found a warning of "Exceptionally long voice queue length" . When i search it on forum i found that This sounds like issue 15609 which has been resolved newer versions of asterisk https://issues.asterisk.org/view.php?id=15609 is it fixed in asterisk-1.6.2 SVN trunk version or not? thanks, Pankaj Thanks & Regards, Pankaj Pandey +91-9990212758 -- _ -- 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] [1.8] Unable to Register: Registration denied because of contact ACL
Try by reversing the line number of permit & deny -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Thursday, March 10, 2011 6:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL > > Thanks. But Like I said, that's all done. Here's the Endpoint config: > > [authentication] > [basic-options](!) ; a template > dtmfmode=rfc2833 > context=Phones > type=friend > contactdeny=0.0.0.0/0.0.0.0 > contactpermit=172.16.16.0/255.255.255.0 > deny=0.0.0.0/0.0.0.0 > permit=172.16.16.0/24 > host=dynamic > qualify=no > insecure=port,invite > > [natted-phone](!,basic-options) ; another template inheriting > basic-options > nat=yes > directmedia=no > > [555](natted-phone) > secret=$$ecret$$ > disallow=all > allow=ulaw > allow=gsm > > no deal! The irony is that we have a similar configuration at another > place, but we didn't need to put anything there and the phones register regardless! > > Is this broken > Perhaps the contactdeny is taking precedence in 1.8. Try it without the contactdeny - maybe the existence of a contactpermit will imply a contactdeny of "everything else". Cheers, j -- _ -- 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] Automatically unpause a paused queue memeber - badidea?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of magnu...@inputinterior.se Sent: Friday, March 11, 2011 3:59 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Automatically unpause a paused queue memeber - badidea? I have some cases when I want to pause a queue member and automatically unpause the queue member after a specified time. Right now I am doing it by a AMI script, but was thinking if it is possible to add a parameter to PauseQueueMember like, PauseQueueMember([queuename],interface[,options[,reason[,time]]]) where time will be how long (in seconds) the interface will be paused. before brought back. Maybe it is a bad idea, I dont know, what do you think? I don't think it's a "bad" idea; Parking works on a similar basis. -- _ -- 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] SIPAddHeader not working
Hello,
does anyone have a SIP trace for me where the SIPheader "Privacy: id" is
present ?? If so, what Asterisk version ?
Kind regards,
Jonas.
On 03/09/2011 06:43 PM, Bryant Zimmerman wrote:
Jonas
In my systems I have seen the Privacy: id when we do our testing but
it has been several months since I have checked it. I am running some
tests later today with one of our customers and I will enable it and
do a capture to confirm but when we do a CID block our vendors say
they are getting the headers correctly
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
*From*: "Jonas Kellens"
*Sent*: Wednesday, March 09, 2011 9:18 AM
*To*: brya...@zktech.com, "Asterisk Users Mailing List -
Non-Commercial Discussion"
*Subject*: Re: [asterisk-users] SIPAddHeader not working
On 03/09/2011 02:09 PM, Bryant Zimmerman wrote:
*From*: "Jonas Kellens"
*Sent*: Wednesday, March 09, 2011 4:18 AM
*To*: "Asterisk Users Mailing List - Non-Commercial Discussion"
*Subject*: [asterisk-users] SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten => s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
Using Asterisk 1.6.2.16.1
How do I correctly add the Privacy header ?!
Kind regards,
Jonas.
Jonas
Here is the way we add the rfc-3325 privacey header so our vendors
pick it up correctly. This is what we use in 1.6.x and 1.8.x
When I check on my versions the privacy header appears to be there.
exten => rfc-3325-CPN,1,NoOp(Set Call Privacy)
exten => rfc-3325-CPN,n,NoOp(From ${SIP_HEADER(From)})
exten => rfc-3325-CPN,n,NoOp(To ${SIP_HEADER(To)})
exten =>
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(SIP_HEADER(From),@,2)})
exten => rfc-3325-CPN,n,GotoIf($["${l_sipheaderfromip}" != ""]?hasat)
exten =>
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(CUT(SIP_HEADER(From),>,1),:,2)})
exten => rfc-3325-CPN,n,Goto(gotip)
exten =>
rfc-3325-CPN,n(hasat),Set(FROM_IP=${CUT(CUT(CUT(SIP_HEADER(From),@,2),>,1),:,1)})
exten => rfc-3325-CPN,n(gotip),NoOp(Gateway IP is ${FROM_IP})
exten =>
rfc-3325-CPN,n,SIPAddHeader(P-Preferred-Identity:"${CALLERID(name)}"
)
exten => rfc-3325-CPN,n,SIPAddHeader(Privacy: id)
exten => rfc-3325-CPN,n,Set(CALLERPRES()=prohib_not_screened)
exten => rfc-3325-CPN,n,Set(CALLERID(num)=Anonymous)
exten => rfc-3325-CPN,n,Set(CALLERID(name)=Anonymous)
exten => rfc-3325-CPN,n,Return()
I see no great difference. What does
"/Set(CALLERPRES()=prohib_not_screened)/" do ?
How does your INVITE look like ? Does the header "/Privacy: id/"
appears ? Because it does not in my INVITE.
Kind regards,
Jonas.
--
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[asterisk-users] How do you handle queues with AMI?
Hey all, I'm in the process of writing a few applications that are going to either monitor the queue (number of calls, positions, etc) or respond to answering a queue call (if you answer, a window pops up with info about caller, hold time, etc.). I'm writing this in C# but language isn't important. I'm not looking for a hand out on code, what I'm really interested in is theory or logic. How are other people watching the call come into the queue and watch it from there. What events are you watching? I've already got the app to recognize the "packets" of information from the AMI so I can handle them accordingly. I know how to action off of the AgentConnect part but what I'm missing is how to tie that back into the call (Caller ID, etc.). I know the first response will be use the Uniqueid for the call but how? What are your methods for tracking it? How do you know it even entered the queue? Also, as I'm writing this, if anyone would like to help out or share code I'm up for it. I'll make my code available to all interested in doing this in C# (it's pretty painless). Thanks! Louis -- _ -- 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] How do you handle queues with AMI?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis Carreiro Sent: Friday, March 11, 2011 9:17 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How do you handle queues with AMI? Hey all, I'm in the process of writing a few applications that are going to either monitor the queue (number of calls, positions, etc) or respond to answering a queue call (if you answer, a window pops up with info about caller, hold time, etc.). I'm writing this in C# but language isn't important. I'm not looking for a hand out on code, what I'm really interested in is theory or logic. How are other people watching the call come into the queue and watch it from there. What events are you watching? I've already got the app to recognize the "packets" of information from the AMI so I can handle them accordingly. I know how to action off of the AgentConnect part but what I'm missing is how to tie that back into the call (Caller ID, etc.). I know the first response will be use the Uniqueid for the call but how? What are your methods for tracking it? How do you know it even entered the queue? Also, as I'm writing this, if anyone would like to help out or share code I'm up for it. I'll make my code available to all interested in doing this in C# (it's pretty painless). Thanks! Louis If you look through your CDR, you'll see the information you need to develop this methodology. Keep in my that (as I understand it), when an agent picks up a call, the uniqueid will change just like the call had been transferred. -- _ -- 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] How do you handle queues with AMI?
Thanks Danny! I took a look at the CDR data through AMI but it only throws an event when the call is hung up. About the Unique ID... it looks like it stays the same through the bridge. I've pasted the AMI output from a queue call below. The only part that changes is the decimal increments. Event: NewAccountCode Privilege: call,all Channel: SIP/Lync-000c Uniqueid: 1299861572.25 AccountCode: OldAccountCode: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: Local/1173@from-queue-ba2a;2 Channel2: SIP/Lync-000c Uniqueid1: 1299861572.24 Uniqueid2: 1299861572.25 CallerID1: 1424 CallerID2: 1173 Event: Unlink Privilege: call,all Channel1: SIP/1424-000b Channel2: Local/1173@from-queue-ba2a;1 Uniqueid1: 1299861572.22 Uniqueid2: 1299861572.23 CallerID1: 1424 CallerID2: Event: VarSet Privilege: dialplan,all Channel: Local/1173@from-queue-ba2a;2 Variable: BRIDGEPEER Value: SIP/Lync-000c Uniqueid: 1299861572.24 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, March 11, 2011 10:21 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How do you handle queues with AMI? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis Carreiro Sent: Friday, March 11, 2011 9:17 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How do you handle queues with AMI? Hey all, I'm in the process of writing a few applications that are going to either monitor the queue (number of calls, positions, etc) or respond to answering a queue call (if you answer, a window pops up with info about caller, hold time, etc.). I'm writing this in C# but language isn't important. I'm not looking for a hand out on code, what I'm really interested in is theory or logic. How are other people watching the call come into the queue and watch it from there. What events are you watching? I've already got the app to recognize the "packets" of information from the AMI so I can handle them accordingly. I know how to action off of the AgentConnect part but what I'm missing is how to tie that back into the call (Caller ID, etc.). I know the first response will be use the Uniqueid for the call but how? What are your methods for tracking it? How do you know it even entered the queue? Also, as I'm writing this, if anyone would like to help out or share code I'm up for it. I'll make my code available to all interested in doing this in C# (it's pretty painless). Thanks! Louis If you look through your CDR, you'll see the information you need to develop this methodology. Keep in my that (as I understand it), when an agent picks up a call, the uniqueid will change just like the call had been transferred. -- _ -- 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] How do you handle queues with AMI?
What we do is just before the call to queue we do a userevent that has the uniqueid and the channel and any other information we care about. You can hold on to this information and match it when you get the agentconnect event. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 11, 2011, at 7:21 AM, Danny Nicholas wrote: > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis Carreiro > Sent: Friday, March 11, 2011 9:17 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] How do you handle queues with AMI? > > Hey all, > > I’m in the process of writing a few applications that are going to either > monitor the queue (number of calls, positions, etc) or respond to answering a > queue call (if you answer, a window pops up with info about caller, hold > time, etc.). I’m writing this in C# but language isn’t important. I’m not > looking for a hand out on code, what I’m really interested in is theory or > logic. How are other people watching the call come into the queue and watch > it from there. What events are you watching? > > I’ve already got the app to recognize the “packets” of information from the > AMI so I can handle them accordingly. I know how to action off of the > AgentConnect part but what I’m missing is how to tie that back into the call > (Caller ID, etc.). I know the first response will be use the Uniqueid for the > call but how? What are your methods for tracking it? How do you know it even > entered the queue? > > Also, as I’m writing this, if anyone would like to help out or share code I’m > up for it. I’ll make my code available to all interested in doing this in C# > (it’s pretty painless). > > Thanks! > Louis > > If you look through your CDR, you’ll see the information you need to develop > this methodology. Keep in my that (as I understand it), when an agent picks > up a call, the uniqueid will change just like the call had been transferred. > -- > _ > -- 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] Is H323 supported when installing Asterisk from Digium Yum repository?
Thanks for that Vladimir, Despite being installed it doesn't seem to be recognized by asterisk: *server55667*CLI> reload set show* *No such module 'set'* *No such module 'show'* *server55667*CLI> reload set show ooh323* *No such module 'set'* *No such module 'show'* *No such module 'ooh323'* *server55667*CLI> ooh323* *No such command 'ooh323' (type 'core show help ooh323' for other possible commands)* *server55667*CLI> core show channeltypes* *TypeDescription Devicestate Indications Transfer* *-- --- --- --- * *DAHDI DAHDI Telephony Driver w/PRI & SS7 & MFC no yes no * *Phone Standard Linux Telephony API Driver no yes no * *IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes * *USTMUNISTIM Channel Driver no yes no * *MGCPMedia Gateway Control Protocol (MGCP)yes yes no * *SIP Session Initiation Protocol (SIP)yes yes yes * *Agent Call Agent Proxy Channel yes yes no * *Bridge Bridge Interaction Channel no no no * *Skinny Skinny Client Control Protocol (Skinny) yes yes no * *Local Local Proxy Channel Driver yes yes no * *--* *10 channel drivers registered.* *[root@server55667 ~]# rpm -qa | grep asterisk16-addons-ooh** *asterisk16-addons-ooh323-1.6.2.3-1_centos5* Does ooh323 have to be loaded in some config file after the rpm install? Thanks, On Fri, Mar 11, 2011 at 12:09 AM, Vladimir Mikhelson wrote: > Bruce, > > Forgot to mention. > > ooh323.conf -- configuration file. > > -Vladimir > > > > On 3/10/2011 9:32 PM, Bruce B wrote: > > But even with *asterisk16-addons-ooh323.x86_64* I don't see any of the > command for h323 in CLI to work. So, I am missing something still. > > On Thu, Mar 10, 2011 at 10:23 PM, Bruce B wrote: > >> I see this in the Digium repository: >> >> *asterisk16-addons-ooh323.x86_64* >> >> Wouldn't that just do it without having to re-install from the source? >> >> Thanks >> >> >> On Thu, Mar 10, 2011 at 10:17 PM, Bruce B wrote: >> >>> Can you please provide link to the RPM? >>> >>> Thanks >>> >>> >>> On Thu, Mar 10, 2011 at 9:58 PM, Vladimir Mikhelson >>> wrote: >>> Alternatively, you can use OOH323 which is available with yum. I am using it for couple years with no major problems. Developer is very responsive. Just finished 1.8 related adjustments to OOH323, should be available in 1.8.4. -Vladimir On <3%2F10%2F2011>3/10/2011 2:29 PM, Jose P. Espinal wrote: > Bruce B wrote: >> Hi everyone, >> >> Installed asterisk from yum repository but I think H.323 is not >> supported as I tried commands like this and they don't work: > > [snip] > >> >> >> Of course I can't go to source since I am using the repository. How >> can I install H.323. Is that OH323 I should look for? > > a. About the first question: > As Danny N. said on his response, H.323 is not available on yum, you > would have to do it from source. > > b. About OH323 and H.323: > You could try OH323, but personally (from past experiences) I would go > with H.323. > > Several weeks ago I posted some SlackBuilds (Script for making > slackware binary packages) of Asterisk to the list. You could download > the scripts and pickup some tips in order to compile the latest > version of H323plus (new name for H.323 project). > > IMO, H323plus project team has made a great job working on H.323 > implementation. > > The scripts are located at: > http://packages.eslackware.com/slackbuilds/asterisk/ > > If you run into any difficulties about any detail on the scripts, you > are free to contact me off-list (of course, for free. No charge, not > fee, etc) > > > -- _ -- 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 UNSUBSC
Re: [asterisk-users] How do you handle queues with AMI?
Thanks Jim! That's actually a great idea! I'm looking into that now! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: Friday, March 11, 2011 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How do you handle queues with AMI? What we do is just before the call to queue we do a userevent that has the uniqueid and the channel and any other information we care about. You can hold on to this information and match it when you get the agentconnect event. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 11, 2011, at 7:21 AM, Danny Nicholas wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis Carreiro Sent: Friday, March 11, 2011 9:17 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How do you handle queues with AMI? Hey all, I'm in the process of writing a few applications that are going to either monitor the queue (number of calls, positions, etc) or respond to answering a queue call (if you answer, a window pops up with info about caller, hold time, etc.). I'm writing this in C# but language isn't important. I'm not looking for a hand out on code, what I'm really interested in is theory or logic. How are other people watching the call come into the queue and watch it from there. What events are you watching? I've already got the app to recognize the "packets" of information from the AMI so I can handle them accordingly. I know how to action off of the AgentConnect part but what I'm missing is how to tie that back into the call (Caller ID, etc.). I know the first response will be use the Uniqueid for the call but how? What are your methods for tracking it? How do you know it even entered the queue? Also, as I'm writing this, if anyone would like to help out or share code I'm up for it. I'll make my code available to all interested in doing this in C# (it's pretty painless). Thanks! Louis If you look through your CDR, you'll see the information you need to develop this methodology. Keep in my that (as I understand it), when an agent picks up a call, the uniqueid will change just like the call had been transferred. -- _ -- 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] dnsmgr_lookup
I am using 1.8.3 and changed enable=no on dnsmgr.conf - however I am still getting log messages for dnsmgr_lookup. I wasnt expecting that. I have a server and a couple dedicated machines just running ALSA connections. I dont need any dns lookups for anything - who do I disable it? Thanks jerry Asterisk Ready. *CLI> *CLI>> doing dnsmgr_lookup for 'mndemo' quit No such command 'quit' (type 'core show help quit' for other possible commands) *CLI> core stop now Beginning asterisk shutdown Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (0). cat dnsmgr.conf [general] enable=no ; enable creation of managed DNS lookups ; default is 'no' ;refreshinterval=1200 ; refresh managed DNS lookups every seconds ; default is 300 (5 minutes) -- _ -- 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 1.8 AGI error ast_carefulwrite: write() returned error
Hey Guys,
We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't
working We have allpage.agi script for paging system on all polycom 501 but
after upgrade it broke. Any idea what is this error ?
extension.conf
exten => 7770,1,agi(allpage.agi)
exten => 7770,2,meetme(7770,dq)
exten => 7770,3,playback(beep)
exten => 7770,4,hangup
following is agi debug
-- Executing [7770@from-sip:1] AGI("SIP/7657-0015", "allpage.agi") in
new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/allpage.agi
AGI Tx >> agi_request: allpage.agi
AGI Tx >> agi_channel: SIP/7657-0015
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1299876046.29
AGI Tx >> agi_version: 1.8.2.3
AGI Tx >> agi_callerid: 7657
AGI Tx >> agi_calleridname: iPhone
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 7770
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: from-sip
AGI Tx >> agi_extension: 7770
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >> agi_threadid: -1345438864
AGI Tx >>
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
AGI Rx << VERBOSE "Found extension (None) in use." 1
allpage.agi: Found extension (None) in use.
AGI Tx >> 200 result=1
AGI Rx << VERBOSE "Found extension 7657 in use." 1
allpage.agi: Found extension 7657 in use.
AGI Tx >> 200 result=1
[Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned
error: Broken pipe
AGI Rx << VERBOSE "Adding extension 7527 to call list" 1
allpage.agi: Adding extension 7527 to call list
AGI Tx >> 200 result=1
[Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned
error: Broken pipe
AGI Rx << VERBOSE "Adding extension 7623 to call list" 1
allpage.agi: Adding extension 7623 to call list
AGI Tx >> 200 result=1
[Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned
error: Broken pipe
AGI Rx << VERBOSE "NOT Adding extension 7657 to call list" 2
== allpage.agi: NOT Adding extension 7657 to call list
AGI Tx >> 200 result=1
[Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned
error: Broken pipe
AGI Rx << VERBOSE "Doing 7527" 0
allpage.agi: Doing 7527
AGI Tx >> 200 result=1
[Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned
error: Broken pipe
-- AGI Script allpage.agi completed, returning 0
-- Executing [7770@from-sip:2] MeetMe("SIP/7657-0015", "7770,dq") in
new stack
-- Created MeetMe conference 1023 for conference '7770'
-- Hungup 'DAHDI/pseudo-729745277'
== Spawn extension (from-sip, 7770, 2) exited non-zero on 'SIP/7657-0015'
--
_
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[asterisk-users] Anyway to monitor SIP debug from originator and terminator separate of each other on two screens?
Hi Everyone, In order to make life easier and to do debugging easier I want to observe "sip set debug originator" and "sip set debug terminator" on two different putty screens. Trick is that originator calls the terminator. I can of course put two separate calls and get sip debugs at different times but that's not what I want to do. I want both to spit out on my two screens at the same time. Unfortunately once in Asterisk CLI when doing any verbose or sip debug it applies to all SSH opened instances and of course to asterisk log...etc... Anyway you can think of that would give two different outputs? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyway to monitor SIP debug from originator and terminator separate of each other on two screens?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Friday, March 11, 2011 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Anyway to monitor SIP debug from originator and terminator separate of each other on two screens? Hi Everyone, In order to make life easier and to do debugging easier I want to observe "sip set debug originator" and "sip set debug terminator" on two different putty screens. Trick is that originator calls the terminator. I can of course put two separate calls and get sip debugs at different times but that's not what I want to do. I want both to spit out on my two screens at the same time. Unfortunately once in Asterisk CLI when doing any verbose or sip debug it applies to all SSH opened instances and of course to asterisk log...etc... Anyway you can think of that would give two different outputs? Thanks This probably won't work but you could try it For peer 1 Asterisk -rx "sip set debug peer peer1" | tee log1.txt For peer 2 Asterisk -rx "sip set debug peer peer2" | tee log2.txt -- _ -- 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 1.8 AGI error ast_carefulwrite: write() returned error
On Fri, 11 Mar 2011, satish patel wrote: We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? [Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned error: Broken pipe Without source code, I'd guess you are violation the AGI protocol. What language are you using? which AGI library are you using? Can you reduce your source code to a simple application that reliably reproduces the error. Can you post the source to the simplified application? -- 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
Re: [asterisk-users] Asterisk 1.8 AGI error ast_carefulwrite: write() returned error
Thanks for reply Steve, I am not in office so i can't post script right now but will so once reach home. By the way that script working great in asterisk 1.2 my production machine. But now I'm testing on 1.8.x and having issue which I mentioned before. This script is perl script and it going to grab all active sip extension and using manager to call all poycom phone via Ring Anwer sipheader. If you want to take a look at script I have following URL where someone already doing discusion. My script is pretty similer but I am grabbing all active extension via asterisk CLI commands not statically hardcoded. http://www.freepbx.org/forum/freepbx/tips-and-tricks/delayed-paging -- Sent from my iPhone On Mar 11, 2011, at 4:58 PM, Steve Edwards wrote: On Fri, 11 Mar 2011, satish patel wrote: We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? [Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write () returned error: Broken pipe Without source code, I'd guess you are violation the AGI protocol. What language are you using? which AGI library are you using? Can you reduce your source code to a simple application that reliably reproduces the error. Can you post the source to the simplified application? -- 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] Asterisk 1.8 AGI error ast_carefulwrite: write() returned error
Un-top-posting... On Fri, 11 Mar 2011, satish patel wrote: We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? [Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned error: Broken pipe On Mar 11, 2011, at 4:58 PM, Steve Edwards wrote: Without source code, I'd guess you are violation the AGI protocol. Can you reduce your source code to a simple application that reliably reproduces the error. Can you post the source to the simplified application? On Fri, 11 Mar 2011, Satish Patel wrote: I am not in office so i can't post script right now but will so once reach home. If you want to take a look at script I have following URL where someone already doing discusion. My script is pretty similer but I am grabbing all active extension via asterisk CLI commands not statically hardcoded. http://www.freepbx.org/forum/freepbx/tips-and-tricks/delayed-paging If you are referring to the allpage.agi script posted about 40% down the page... It is not an AGI. Note that it does not use any AGI library and that it does not read the AGI environment from STDIN -- which violates the AGI protocol. The allpage script connects to Asterisk via TCP using the AMI protocol. In your dialplan, if you change 'agi(allpage.agi)' to 'system(allpage.agi)' does it behave as you expect? Can you execute the script from a shell command line? -- 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
