Re: [asterisk-users] dialplan problem : not including context

Doug Lytle wrote: Then I don't know. I am still on 1.4.x and my PRI context contains all that I'm matching against (No sub contexts). One thing does come to mind; the inbound call is coming into your s extension and then your doing a gosub, in which case, you might be

Re: [asterisk-users] dialplan problem : not including context

Jonas Kellens wrote: there is only one include-statement Then I don't know. I am still on 1.4.x and my PRI context contains all that I'm matching against (No sub contexts). Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, de

Re: [asterisk-users] dialplan problem : not including context

Jonas Kellens wrote: Does this mean the Return() comes before Asterisk looks into the context [TrunkAccounts] ?? No, I believe the includes are read first, but the order in important. Since you may be matching against another context that may cause failure. For example, I have the following

Re: [asterisk-users] dialplan problem : not including context

Jonas Kellens wrote: Meaning ? Meaning I want to see the dialplan order of that context. I'm guessing that's your inbound context. With includes that also include sub-contexts. Usually, there is something ordered differently then expected. Also, what version of Asterisk? Doug -- Ben

Re: [asterisk-users] dialplan problem : not including context

Jonas Kellens wrote: Everything works fine when including context 'TrunkAccounts' in context 'PROVIDERin dialplan showPROVIDERin Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." --

Re: [asterisk-users] dialplan problem : not including context

Jonas Kellens wrote: I have the following in dialplan : [TrunkAccounts] dialplan show TrunkAccounts Make sure the sort order is what you're expecting. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty

Re: [asterisk-users] Question on system command 1.4.43

Jerry Geis wrote: Any ideas on what might be happening here? If Asterisk isn't running as root, my guess would be a permissions issue. By the way, out of curiosity, what does "usbserial -start" do? I'm guessing, some type of paging busy notification? Doug -- Ben Franklin quote: "Thos

Re: [asterisk-users] Problem w/ PC port on Polycom 335

Mike Diehl wrote: Usually, it just works... Any ideas? I've seen this before. One of our facilities have 'smart or managed' switches that have caused no ends of problems, including preventing computers plugged into the phones not having network access. You may want to review your switche

Re: [asterisk-users] Set Call type in dial plan

Faraj Khasib wrote: Please help, I have tried many things I cannt make it work, when I make an audio call it is converted by asterisk to video call request Not that I can help, since I don't do any video calling. But, if you don't give any information about your system (OS and version, Aste

Re: [asterisk-users] Recompile Asterisk

LL wrote: I need to recompile asterisk to install some apps. Does anyone has a good tutorial on how to do this From the statement above, you're giving conflicting information. You first sounded like the installation was compiled from source, but then you're asking for a tutorial on how to co

Re: [asterisk-users] how to listen on different sip port for a device?

Kevin P. Fleming wrote: If you are using a TCP-capable version of Asterisk and a TCP-capable endpoint, you can try that. Or, You can setup OpenVPN, works very well. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neit

Re: [asterisk-users] Dialing problem with Polycom phones after SIP update

Eric Wieling wrote: Polycom (r) UC Software: Configuration File Conversion Utility\ On the pagehttp://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip560.html And for those of us without Windows, this utility appears to work fine under wine. Doug

Re: [asterisk-users] OT: Does IEEE 801.2q include VLAN trunking?

Ruben Rögels wrote: Does a switch stating to support IEEE 801.2q also supports VLAN trunking? I don't know if you miss-typed or not. But, 802.1q is VLAN. If it was typed incorrectly, then yes. I just recently setup a Linux DHCP to handle multiple networks with VLAN 802.1q trunking. Doug

Re: [asterisk-users] CDR mysql with asterisk 1.4

salaheddine elharit wrote: because this server is very important for me and i can’t stop it This is one of those "schedule for after hours" things then. I don't believe you can do this without a restart of the Asterisk service. But, down time should be minimal. Doug -- Ben Franklin quo

Re: [asterisk-users] Using asterisk with DSP chips

Sazzad wrote: Is it the wrong place for this question? Is it the wrong place for the question? No. Does anybody that has read your message know the answer, probably not. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserv

Re: [asterisk-users] Question about Read() application

Danny Nicholas wrote: The user reported to me that "I punched 1 and it hung up" - in my testing, I found that slow DTMF entry (1 digit every 2 seconds or so) or fast entry (more than 10 digits per second) was most likely to cause the problem. I've never had mine just hangup on a mis-key, but

Re: [asterisk-users] Strange behavior over Zap chennels

Richard Reina wrote: I have a server that is hooked to a channel bank (Adit 600). It has eight lines coming from a T1 through this adit's FXO card. I have a very similar setup, but my Adit 600 is used to provide FAX only. Voice calls are PRI. And, my system hasn't been rebooted in over 2 y

Re: [asterisk-users] Problem with outbound dialing from remote phone

Adam Robins wrote: No change, thanks Well, In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, dese

Re: [asterisk-users] Problem with outbound dialing from remote phone

Adam Robins wrote: Any ideas, suggestions, etc., would be greatly appreciated My guess that the Polycom digitmap isn't being loaded (sip.cfg). I'm sure if she were to dial the phone number and then press 'send' soft key, it'd probably dial. Doug -- Ben Franklin quote: "Those who woul

Re: [asterisk-users] PRI Issues After Upgrade

Stephen H. Gerstacker wrote: Came in this morning to more of the same: Then, if you have the ability, I'd drop 1.2 back into place and see if it's happy. But, my feeling is that you'll need to contact the provider. The other thing that comes to mind is that your PRI card is having issues.

Re: [asterisk-users] PRI Issues After Upgrade

Stephen H. Gerstacker wrote: Is there a big difference between the two? From what I've read, a DMS100 can redirect a call off of your system, meaning that if you have an inbound call and you want to redirect it to a different number, the DMS100 will redirect the call and take your system ou

Re: [asterisk-users] PRI Issues After Upgrade

Stephen H. Gerstacker wrote: I'm just a simple programmer who happens to be the only IT guy in the office. And I'm just an IT guy, that started learning (And still am) about phone systems around 10 years ago. I was just going through the process of elimination, the differences between our

Re: [asterisk-users] PRI Issues After Upgrade

Stephen H. Gerstacker wrote: Anything else I can try? Try switchtype=national just for testing. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." --

Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

linux guy wrote: So, other than a bit of disk space, is there any reason why I shouldn't install KDE when I set it up ? KDE has other associated background services that "may" slow a machine down. If you're looking for a DE, I'd go with something light weight. LXDE is my preferred choice.

Re: [asterisk-users] PRI Issues After Upgrade

Stephen H. Gerstacker wrote: switchtype=dms100 Are you sure that your switchtype is correct and your provider has a dms100? If not, change this to national. If so, the two things I'd try to narrow it down are: 1.) Temporarily remove the AEX410 card and test again 2.) Temporarily disab

Re: [asterisk-users] PRI Issues After Upgrade

Stephen H. Gerstacker wrote: so any help would be appreciated You could start by supplying snippets of your config for the PRI. Telling us what PRI card you're using. For example, my dahdi config for my PRI: cd /etc/dahdi cat system.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzo

Re: [asterisk-users] Polycoms rebooting themselves

Mike Diehl wrote: Any other ideas? They should be writing out logs to your ftp server (If your provisioning them that way). Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." --

Re: [asterisk-users] PRI Problem

Eric Merkel wrote: The error pretty much right away. The first thing that comes to my mind is to check your cable. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- __

Re: [asterisk-users] FAX Issues

Thorolf Godawa wrote: My experience is, that faxing over IAXModem is not reliable. My experience is just the opposite, but I'm not faxing over IP. PRI, over slinear. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve ne

Re: [asterisk-users] Answering machine answers after pickup a phone.

Don Kelly wrote: There are analog phones connected to the same PSTN. And that is why what you want to do won't work. To have it do what you'd like, you'd need to have Asterisk as the only one receiving the call. Then your other analog phones would need to be on something like a ATAs that w

Re: [asterisk-users] Asterisk as a Operator Phone

Nikhil wrote: Any reply on this.. Not really sure what you're asking. My guess would be to search for Flash Operator Panel or FOP Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." --

Re: [asterisk-users] Fail to compile DAHDI under Linux 3.0.0

Shaun Ruffell wrote: You could either run off the tip of the 2.4 branch Thank you very much! Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." --

[asterisk-users] Fail to compile DAHDI under Linux 3.0.0

Just finished installing Kernel 3.0.0 and tried to compile DAHDI Complete 2.4.1.2+2.4.1. Errors below: /home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:95:2: warning: #warning "No CONFIG_BKL is an experimental configuration." CC [M] /home/doug/E

Re: [asterisk-users] Recall: Time zone on phones

motty.cruz wrote: motty.cruz would like to recall the message, "Time zone on phones". Good luck on that one! Exchange feature only. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." --

Re: [asterisk-users] Compact, affordable x86 devices?

Gilles wrote: Does "Availability>500" mean that it's just not possible to buy just one item? I read it as they have over 500 in stock. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -

Re: [asterisk-users] Macro issue under 1.8.5

--[ UxBoD ]-- wrote: I back leveled to 1.8.3 and that works fine. What am I missing as app_macro has been installed okay? Macro was depreciated in 1.6 and most likely removed in 1.8.5 I'd check the upgrade.txt to see if there is an entry about it, from the wiki " Macro Implementation.

Re: [asterisk-users] Redirecting call from one E1 to another?

Tony Mountifield wrote: I don't want just to relay the call through to the second box using IAX or SIP or an additional PSTN channel. What I would like to do is to redirect the call in the PSTN so that it ends up connected only to the second box. If I recall correctly, it's only possible if the

Re: [asterisk-users] Meetme not prompting for PIN

Doug Lytle wrote: I've been searching the Jira issue tracker and found a ticket: What I ended up doing was to copy the app_meetme.c out of the 1.4.30 source and compiled it into my current Asterisk setup. I now have PIN prompts. Doug -- Ben Franklin quote: "Those who wou

Re: [asterisk-users] Meetme not prompting for PIN

Steve Totaro wrote: I guess you could do it the old fashioned way until you open a ticket I've been searching the Jira issue tracker and found a ticket: https://issues.asterisk.org/jira/browse/ASTERISK-16747 Not being familiar with the new Jira system, I can't seem to find a patch for the 1.

[asterisk-users] Meetme not prompting for PIN

I've just put into place an updated meetme server. I went from 1.4.20.1 to 1.4.42. In testing, it would seem that dynamically created conferences will not prompt for the PIN. I've read though the readme and even went as far as reading the 1.2 to 1.4 upgrade document. s far as I can see, th

Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

Warren Selby wrote: Not trying to start a war here, That may be, but I have experience with VB. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _

Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

Patrick Lists wrote: With virtualized environments prone to timing issues does this make sense at all? Wouldn't the timing be taken from the PCI device? This is how it's working on my systems now. Also, it's not something I would be put into production until I've hammered it. Doug -- Be

Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

Steve Totaro wrote: Source code? http://download.virtualbox.org/virtualbox/4.1.0_BETA2/VirtualBox-4.1.0_BETA2.tar.bz2 Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- ___

[asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- ___

Re: [asterisk-users] How to without GUI

Danny Nicholas wrote: step by step how to set up a SIP trunk just by editing sip.conf You'll find that most here don't use a GUI. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- __

[asterisk-users] [SOLVED] Re: Asterisk 1.4 func_odbc frustrations

Doug Lytle wrote: exten => s-setup,n,Set(ODBC_CONFERENCE_WRITE(room=${conference.room}\,password=${put-new-password}\,admin=${conference.admin}\,scheduled=${TODAY}\,owner=${conference.owner}\,comments=${conference.comments}) exten => s-setup,n,Set(ODBC_CONFERENCE_WRITE()=${conferenc

[asterisk-users] Asterisk 1.4 func_odbc frustrations

Maybe somebody can help me here. I've finally got another server together, so I can test and upgrade a couple of my older 1.4.x installations. I figured that while I'm at it, I'll give func_odbc a try (have been using the mysql addon), knowing full well that when I finally move over to 1.8.x

Re: [asterisk-users] Cisco IP Phones 7942 and Skinny/SIP in asterisk

bilal ghayyad wrote: from where you got these files? I found the link on http://www.voip-info.org I searched for Cisco 7940 http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary

Re: [asterisk-users] Cisco IP Phones 7942 and Skinny/SIP in asterisk

bilal ghayyad wrote: Dears; The Cisco 7942 worked in SIP and did not work in skinny firmware (in skinny, it register but no voice can be heared). My phones are Cisco 7940s, so it may be a different layout then expected: cat dialplan.xml Doug -- Be

Re: [asterisk-users] Connected Line ID

Arjan Kroon | Mobillion wrote: But is there a patch from version 1.6.2.12? Not that I can see. You could try applying the patches against that version and see if they apply cleanly. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Saf

Re: [asterisk-users] Connected Line ID

Arjan Kroon | Mobillion wrote: And if I can see it, this patch is already included in version 1.6.2.12. Or am I wrong? That I can't answer. I'm still using 1.4.x and am experimenting with 1.8.x. I recall reading that it wasn't supported directly until 1.8 without patches. Doug -- Ben

Re: [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel

bilal ghayyad wrote: But I am afraid it is a bug because I read something this in the below This bug is referring to Zaptel, not dahdi. If things were working fine, and you haven't made any recent changes, in my experience it's always been provider (99%) or cable (1%) issues when I've lost m

Re: [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway!

bilal ghayyad wrote: There is a Yellow Alarm, so what it could be the problem? Experience says you need to call your provider. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- ___

Re: [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco

bilal ghayyad wrote: Any one can suggest a TFTP server to be installed in Fedora The one in the repos should be fine. I'm not Fedora (I'm Mandriva), but both are RPM based. Try: yum install tftp-server. The one that comes with Mandriva works fine with our Cisco 7940/7960 phones. Doug

Re: [asterisk-users] Connected Line ID

Arjan Kroon | Mobillion wrote: But are there also pathes for version 1.6 The last patch available for the 1.6 series was for 1.6.0.1: https://issues.asterisk.org/jira/browse/8824 Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety,

Re: [asterisk-users] Callerid issue

Steve Totaro wrote: For each phone, add "callerid="Joe Smith"<1551212> " no quotes in sip.conf The problem with that solution is that station to station calls will show the same CID and not the extension. I'd vote for the database approach. Doug -- Ben Franklin quote: "Those who wo

Re: [asterisk-users] Connected Line ID

Arjan Kroon | Mobillion wrote: Does anybody have problems with a wrong Connected Line ID with asterisk version 1.6 As far as I know, unless you're applying patches yourself, Connected Line ID is only available for the 1.8 series. I'm running it on 1.4 with patches. Doug -- Ben Franklin

Re: [asterisk-users] PRI hangup request, cause 18

satish patel wrote: We are getting hangup cause 18 http://networking.ringofsaturn.com/Routers/isdncausecodes.php *Cause No. 18 - no user responding.* This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within t

Re: [asterisk-users] why doesn't "s" accept incoming call

sean darcy wrote: Shouldn't "s" work here? S stands for start. Inbound calls via a provider, to your inbound context would match. For you example, it'd have to be: [out] exten => +1xxxyyy,1,NoOp( this is the extension: ${EXTEN}) exten => +1xxxyyy,n,Answer() exten => +1xxxyyy,n(

Re: [asterisk-users] MagicJack quality

Warren Selby wrote: A quick google for "magicjack quality C. Savinovich" Interesting, it would appear that my archives are incomplete. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _

Re: [asterisk-users] MagicJack quality

Steve Edwards wrote: My archives don't go back that far Mine do. No match on Jack, magic or itntelecom.com Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -

Re: [asterisk-users] call files .vbs

Thomas Perron wrote: Can a vb script run somehow on a Linux machine or does it only work on Windows? Visual Basic is Windows specific. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- __

Re: [asterisk-users] OT - Which Android handset with Wifi-only ?

sean darcy wrote: anyone actually used this on Android to connect to an asterisk server? Well, since I can't find a download to test this on my Droid and I don't plan on paying before testing, I haven't. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a

Re: [asterisk-users] Light indicator managed by Asterisk

Andrew Thomas wrote: Cor-wrong (sort of). There is a backport of DevState/Device_State for 1.4 https://issues.asterisk.org/view.php?id=15818 Very cool! I'll have to review this weekend. Thank you! Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a

Re: [asterisk-users] Light indicator managed by Asterisk

Eric Wieling wrote: pbx*CLI> core show application minivmmwi Core show application minivmmwi core show function DEVICE_STATE Both of these must be a 1.6.x or newer, I have neither under 1.4 Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little T

Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

Sebastian Arcus wrote: Cisco phones (at least the 7940) are supposed to be run with a tftp server available at all time That is my experience. But, if you're running tftp under Linux, then it's probably spawned by xinetd and won't be running unless the service is requested. Doug -- Ben F

Re: [asterisk-users] question on ways to activate voicemail light on polycom

Jerry Geis wrote: Is there a way in asterisk to Activate/Clear the blinking light on polycom phones I've always just dumped a msg000.txt in the voicemail directory of that phone and removed it when not needed. Under 1.4, the Polycoms act on it. Doug -- Ben Franklin quote: "Those who woul

Re: [asterisk-users] best current version and motherboard/CPU compatibilities

|| dave cantera Mobile wrote: I had several AMD athlons 64bit My Myth server is this and I have an AMD X2 for my desktop. Todays' chips from AMD are great. It's just those K6 & K6-2s Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary S

Re: [asterisk-users] best current version and motherboard/CPU compatibilities

|| dave cantera Mobile wrote: why are you shaking!?!? The AMD K6 and the AMD K6-2 were (At that time) cheaper then what Intel had to offer, I built many systems based on both. Sorry, but that memory, along with the memories of running a BBS, just made me shudder. Doug -- Ben Franklin

Re: [asterisk-users] receive faxes

Steve Edwards wrote: Should be worth about a bazillion bad karma points +1 Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- __

Re: [asterisk-users] receive faxes

Eric Wieling wrote: Does Hylafax support T.38? Not at this time, no. And, I have no need for it. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- ___

Re: [asterisk-users] best current version and motherboard/CPU compatibilities

C F wrote: model name : AMD-K6(tm) 3D processor *shudder* Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ --

Re: [asterisk-users] dial from voicemail

Kelly Opal wrote: I would like my customers to have a choice of dialing 1 to get my cell phone while in voicemail I use a combination of the follow me feature, along with the 'a' extension: [s-NOANSWER] exten => s,1,Gosub(mailbox_exist,s,1) exten => s,n,Set(VoiceMailCount=${VMCOUNT(${ARG1}@si

Re: [asterisk-users] Flite issue

Satish Patel wrote: from where I get Header file of flite Just a guess: yum install flite-devel Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _

Re: [asterisk-users] If voice mail not found dialplan

Satish Patel wrote: want it say "person doest have vm setup yet." smthing like that. How should I handle this in my dialplan ? Here is a snippet of what I do: [s-NOANSWER] exten => s,1,Gosub(mailbox_exist,s,1) [mailbox_exist] exten => s,1,Set(_direct_vm=${ARG1}) exten => s,n,MailboxExist

Re: [asterisk-users] Asterisk FOP

Flavio Miranda wrote: Hi, FOP 1 OS Debian Lenny Asterisk 1.6 All my installs are under Mandriva, running op_panel-0.30.tar.gz. Googling was inconclusive, varying from install FOP2 to having an old swf in your browser cache. Maybe your perl is messaged up. Give it a shot on a different m

Re: [asterisk-users] Asterisk FOP

Flavio Miranda wrote: I am truing to set up FOP but I getting the following log: What version of FOP? 1 or 2, what OS? What version of Asterisk? Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Saf

Re: [asterisk-users] agi voicemail callback

vip killa wrote: I'm wondering if there is a simply way to perform a voicemail callback feature using AGI I don't have an AGI, but I do have dial-plan code. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Libert

Re: [asterisk-users] hello

Bradley D. Thornton wrote: WTF does that have to do with Asterisk? It's called spam. And either he doesn't know what non-commercial discussion means or he signed up just to send this. Doubtful we'll see him again. Doug -- Ben Franklin quote: "Those who would give up Essential Libe

Re: [asterisk-users] Back-to-back asterisk PRI issue

satish patel wrote: group = 0,24 Granted, I'm still running 1.4.x, but I don't believe the above is valid. My guess is it should be: group = 0 Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety

[asterisk-users] Mail list issues?

I've seen 5 since 4am EST Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- Bandwidth and Colocation Provided by http:

Re: [asterisk-users] wrong time retrieved from system command

aster...@ck-lee.com wrote: It is about the same. Then my guess is that the machine's timezone is incorrect. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _

Re: [asterisk-users] wrong time retrieved from system command

asterisk asterisk wrote: ${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)} I use the above command to get the system date and time I don't specify timezone when I'm working with EPOCH, what happens when you remove it? Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to pur

Re: [asterisk-users] 1.8.3 - IAX - echo - jitterbuffer

sean darcy wrote: BTW, if you haven't, try iaxagent on your phone. Sean, I'm probably one of the very few that have an Android and refuse to get a Google account. Does the developer have a site where iaxagent can be downloaded? I wouldn't mind giving it a try. Doug -- Ben Franklin quot

Re: [asterisk-users] Loudness of recorded wav-audio

Danny Nicholas wrote: In sip.conf, add rxgain=-4.0 to the peer. The last I knew, rx/tx gains are only for dahdi/zaptel devices. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- __

Re: [asterisk-users] mySQL connection testing

Danny Nicholas wrote: Not to mention that Asterisk is "developmental" and "a moving target". And that's why I'm still on 1.4. And, I have no experience with AGI, nor have I had the time to tackle it in the last 6 months. When I finally do move over to 1.8 series, I plan on looking into ODBC

Re: [asterisk-users] mySQL connection testing

Andrew Thomas wrote: MYSQL_STATUS??? Is this documented anywhere (as I can't seem to find anything about this variable)? core show application mysql hylafax*CLI> -= Info about application 'MYSQL' =- [Synopsis] Do several mySQLy things [Description] MYSQL(): Do several mySQLy things Sy

Re: [asterisk-users] mySQL connection testing

Andrew Thomas wrote: The wait is there as a test. This gives the 'tester' the option of hanging up before the disconnect or not. And the purpose for that would be to share available connections? I've always considered it bad to leave a connection open and have always closed them down aft

Re: [asterisk-users] mySQL connection testing

Andrew Thomas wrote: exten => _X.,n,MYSQL(Clear ${resultid}) exten => _X.,n,Wait(10) ; just for fun exten => _X.,n,MYSQL(Disconnect ${connid}) You should be doing the wait after the disconnect. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a litt

Re: [asterisk-users] Question on Asterisk 1.8 and Wait()

Jerry Geis wrote: Your correct. it was a syntax change. the above works. I've always used Wait(#) in my 1.4.x dial plans. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- ___

Re: [asterisk-users] Handle in dialplan user disconnection

Luiz Gustavo Chiaretto wrote: > only need identify when user's hangup in a Read/Background/Dial > application. If you're going to be doing a read/background/dial, then just before any of them, set a variable. For example: exten s,1,Set(_Flag=Read) exten s,n,Read(get-admin-password|enter-pas

Re: [asterisk-users] Handle in dialplan user disconnection

Sidarta Aguiar de Oliveira wrote: only need identify when user's hangup in a Read/Background/Dial application. Set a flag for those cases and then trap it in the H extension Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deser

Re: [asterisk-users] Dialplan end of pattern matching question

Gabriel Ortiz Lour wrote: exten => _6XXX,1,NoOp(test1) exten => _,1,NoOp(test2) exten => _,2,NoOp(test3) Being in the same context, it would look for the next priority that would match. _,2 would be that priority. I would suggest: exten => _6XXX,1,NoOp(test1) exten => _6XXX,n,

Re: [asterisk-users] Voicemail email attachment as MP3, with tags containing sender name, number, message number

Michelle Dupuis wrote: If anyone wants to try it out let me know! I'd also be interested in having a look! Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- __

Re: [asterisk-users] Paging a message. How?

Russell Brown wrote: I can do a normal page (phones auto-answer on speaker) with SipAddHeader but that doesn't let me play a pre-recorded message. Any suggestions? Dial plan: [polycom-page] ;* ;* Play the previously recorded page file over t

Re: [asterisk-users] dialplan announcements

ERIC HERRON wrote: The conditions are that I want the recorder to dial a feature code that prompts him to record the message. He then presses 1 to accept. This gets saved as announcement.wav. You'll need to supply some of your own audio prompts, I recorded my own, here is the code I have: [

Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer

Mike wrote: I was hoping to use this Queue not for professional agents in a call center, but for reception. When the receptionist (lowest penalty) is not at the desk, then some junior sales person can pick up those calls. We have our receptionist setup in a front-desk queue that has 2 phon

Re: [asterisk-users] Polycom 500 / MWI

Brian C. Huffman wrote: Does anyone know how to setup this phone to work with asterisk so that the indicator light comes on when there's a new message and goes off quickly (less than a minute) after the message is deleted? My phone.cfg for extension 4221 and the voicemail extension of 4200 lo

Re: [asterisk-users] Music on Hold not working?

Tzafrir Cohen wrote: Is that really an issue? open() and all others would normally just reduce '//' to a '/'. That, I really wouldn't know. I'm not a programmer. I noted the differences between mine and his. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to

Re: [asterisk-users] Music on Hold not working?

James Miller wrote: From the command you suggested to enter: Class: default File: /var/lib/asterisk/moh//reno_project-system You have 1 too many // in the directory structure. It should be: /var/lib/asterisk/moh/reno_project-system Not /var/lib/asterisk/moh//reno_project-system

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