[asterisk-users] External Extension with Extension
Hi, Wondering if there is any way to set up an external extension in * where the external number requires an extension? So if I have ext. 250 Local/5556667...@outbound, can I add an extension number to that to be automatically dialed somehow? Thanks, Kenny -- _ -- 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 start with php
On 4/2/2010 12:12 PM, Steve Howes wrote: On 2 Apr 2010, at 18:49, salaheddine elharit wrote: thank so much again for your response , i don't understand what shoud i do if you can please give me more information how to do in oreder to excute this script He's damned near written it for you. Try researching the terms he used, and try doing a few of the bits. This is a community of people providing support to each-other, not a bunch of slaves to do your work for free. Steve Really, it's not even a complicated task. It seems you may be in over your head. Hiring a contractor is probably something you should consider. I mean no offense, but as stated, he really couldn't have been more clear, and judging by your responses this doesn't sound like a project you are ready to take on. Luckily for you, there are probably numerous out of work developers right next door that would jump at the chance to help you out! The work you need shouldn't take any more than 1 billable hour. You could even hire a consultant to teach you how to do it, and learn some new things at the same time. Best of luck! Kenny -- _ -- 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] SIP Connection Question
Hi All, I have a question about how a particular situation would work between two PBX systems: If I were to connect my Mitel box to my Asterisk box via a SIP trunk (same rack, same network), and then pass a call from the Mitel to Asterisk to perform some functions (lookups, maybe routing), and then pass the call back to the Mitel to be routed to it's endpoint, would Asterisk stay in that loop after the call was passed back to the Mitel? Or, does the call leave Asterisk completely when passed back? If it does leave/stay in the loop, is there a way to force it to leave/stay based on what my needs are? Thanks, Kenny -- _ -- 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] SIP Connection Question
On 4/1/2010 5:09 PM, Juan E. RodrÃguez wrote: Depends on the configuration you make. For example, if you want to route the call giving the Mitel a new desrination or prefix, you can use Transfer dialplan app. Transfer before answering the call will be redirected with SIP 302. If the call is to be anwered on *, then canreinvite set to yes or directrtp set to yes can help you. Saludos, Juan E. RodrÃguez -Original Message- From: Kenneth Noisewaternoisewater...@gmail.com Date: Thu, 1 Apr 2010 16:50:47 To:asterisk-users@lists.digium.com Subject: [asterisk-users] SIP Connection Question OK, so for instance if I passed a call to Asterisk and grabbed CID info and did some lookups and then transferred it back to mitel to route to a user, then * would be out of the call path (loop, whatever). But, if I were to answer that call in * with an IVR to collect caller input to use and then transferred the call back to the Mitel to route to the endpoint, * would remain in the call. Is that a correct understanding? Also one more question, and please excuse my ignorance (I'm just a developer with pretty limited knowledge on the telephony side of things): When I talk about connecting the Mitel box and the Asterisk box together via a SIP trunk, is that trunk equal to 1 analog line, or channel or whatever, or can I make as many connections as I want on that trunk? Again, my knowledge is a bit limited, and thusfar people have been using a lot of terms interchangably with me to add to my confusion :). This only concerns me because I'm pretty sure we have to buy a license for each SIP trunk with Mitel. It would be really great if I could work out a solution like this, it will allow me to prove Asterisk's worth to my management, and open up a lot of doors for us and our internal apps. The Mitel SDK is unfortunately rather limited, but management is not in any way interested in jumping ship from Mitel to Asterisk. Personally, I say jump, I've had great experience with Asterisk, even in fairly heavy use situations. Anyone have any input on selling Asterisk to higher up's? I know there is the whole enterprise support aspect, but my team manages the Mitel stuff as it is anyway, and I think we'd all much rather be dealing with Asterisk/SER as the core solution. Thanks everyone for your input! Kenny -- _ -- 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 Greetings Will Not Save
extension (from-internal, h, 1) exited non-zero on 'SIP/
200-00fd3150'
Here is voicemail.conf:
[general]
format = wav49|gsm|wav
serveremail = voicem...@edited.com
fromstring = Voicemail System
attach = yes
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 3
emailsubject = Message from ${VM_CALLERID} in mailbox ${VM_MAILBOX}
emailbody = Dear ${VM_NAME}:\n\nThis is a message to inform you of a $
{VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX}
from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it
when you get a chance.\n\n
emaildateformat = %A, %B %d, %Y at %r
[zonemessages]
eastern = America/New_York|'vm-received' Q 'digits/at' IMp
central = America/Chicago|'vm-received' Q 'digits/at' IMp
central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military = Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
european = Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
[default]
202 = 1234,Chad Edited,c...@edited.com,,attach=yes|saycid=no|
envelope=yes|delete=no
300 = 1234,Don Edited,donextr...@edited.com,,attach=yes|saycid=no|
envelope=yes|delete=no
201 = 1234,Danica Edited,dan...@edited.com,,attach=yes|saycid=no|
envelope=yes|delete=no
200 = 3816,Camron Edited,c...@edited.com,,attach=yes|saycid=no|
envelope=yes|delete=no
Let me know if any additional information is needed.
Respectfully,
Dr. Kenneth Noisewater, Phd
On Apr 11, 2009, at 5:50 AM, Doug Lytle wrote:
Dr. Kenneth Noisewater wrote:
-Asterisk and FreePBX source installs on CentOS 5.4
Without version numbers and console output and samples of your
dialplan,
it'g going to make it very difficult to help.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Temporary Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] Voicemail Greetings Will Not Save
I very probably did build them with ODBC or MySQL support. IMAP I don't think so, but where would I look for configs that tell asterisk to use such support? I'm almost positive I compiled it to support database, but I definitely never configured it for use. Or is this something it does automatically and I need to recompile? Thank you very much for your help. On Apr 13, 2009, at 6:14 PM, Tilghman Lesher wrote: On Monday 13 April 2009 05:54:38 pm Dr. Kenneth Noisewater wrote: Hi All, -My asterisk will not save voicemail greetings when you call in and record them. -It also will not save voicemail messages after emailing them,even though delete=no. -Folder permissions are fine, no errors in asterisk cli. -If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, and then call in and record new unavail message, unavail.wav disappears? Can anyone help point me towards any possible info to fix this, i'm stumped and losing hair! You wouldn't happen to have built voicemail with ODBC and/or IMAP support, would you? That would make the most sense, as both of those engines remove recordings from the directory after having sucked them into the relevant backend storage device. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Greetings Will Not Save
Hi All, Just wanted to post a follow up in case anyone else has the same issue in the future. I recompiled Asterisk and in the makemenu system there is a Voicemail Build Options, in there there is []ODBC Storage and []IMAP Storage. I had ODBC Storage checked on my last compile, I unchecked it, finished building and it all works now. Apparenlty this does not install the option of using ODBC storage, it commits you to ODBC storage without any additional configuration. Tilghman, thanks, your question is what ultimately led me to my solution. Respectfully, Dr. Kenneth Noisewater, Phd On Apr 13, 2009, at 6:14 PM, Tilghman Lesher wrote: On Monday 13 April 2009 05:54:38 pm Dr. Kenneth Noisewater wrote: Hi All, -My asterisk will not save voicemail greetings when you call in and record them. -It also will not save voicemail messages after emailing them,even though delete=no. -Folder permissions are fine, no errors in asterisk cli. -If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, and then call in and record new unavail message, unavail.wav disappears? Can anyone help point me towards any possible info to fix this, i'm stumped and losing hair! You wouldn't happen to have built voicemail with ODBC and/or IMAP support, would you? That would make the most sense, as both of those engines remove recordings from the directory after having sucked them into the relevant backend storage device. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Greetings Will Not Save
Hi All, -My asterisk will not save voicemail greetings when you call in and record them. -It also will not save voicemail messages after emailing them,even though delete=no. -Folder permissions are fine, no errors in asterisk cli. -If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, and then call in and record new unavail message, unavail.wav disappears? -Asterisk and FreePBX source installs on CentOS 5.4 Can anyone help point me towards any possible info to fix this, i'm stumped and losing hair! Respectfully, Dr. Kenneth Noisewater, Phd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
