Re: [asterisk-users] Asterisk behind D-Link ADSL router with private IP
You didn't give full details... which port is unreachable? 5060? some random RTP port? Did you forward udp or tcp or both? Also why did you type in "gmail" when outlook asked you for your name? :) Is virtual server the same as dmz option? On Sun, Nov 21, 2010 at 4:26 AM, gmail wrote: > i have this configuration , An Asterisk server connected to my private > LAN 192.168.10.0/24 when i do port forwarding for port 5060 so that i make > a call from Internet into Asterisk wireshark show the message "destintion > port unrechable" > > i configured sip.conf for "nat=yes" and "qualify=yes" and "externip="my > public IP" > > did i forget some other ports to forward otherthan 5060? > > did i forget any other configurations? > > i even tried the "virtual server" function in my D-Link 2640U ADSL router > with no hope > > > appreciate your help > > -- > _ > -- 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 behind D-Link ADSL router with private IP
i have this configuration , An Asterisk server connected to my private LAN 192.168.10.0/24 when i do port forwarding for port 5060 so that i make a call from Internet into Asterisk wireshark show the message "destintion port unrechable" i configured sip.conf for "nat=yes" and "qualify=yes" and "externip="my public IP" did i forget some other ports to forward otherthan 5060? did i forget any other configurations? i even tried the "virtual server" function in my D-Link 2640U ADSL router with no hope appreciate your help -- _ -- 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] Installing Asterisk to it's own directory
On 10-11-19 04:56 PM, Stephen Brown wrote: > I've never tried this before, and before I potentially break something > I'd like to know if it's possible and how to implement it? > $ mkdir -p ~/digium/asterisk/testing $ cd ~/digium/asterisk/testing $ svn co http://svn.asterisk.org/svn/asterisk/branches/1.8 $ cd 1.8 $ ./contrib/scripts/live_ast configure $ make $ ./contrib/scripts/live_ast install $ ./contrib/scripts/live_ast samples $ ./live/asterisk -vc -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge
You need the following modules at least. Maybe more, but in a vanilla config you will have them already loaded. It took me some half an hour to figure out that bridge_softmix is needed. app_confbridge.so bridge_softmix.so extensions.conf (this is the most vanilla conference room you can get. Answer is important here): exten => 32,1,Answer exten => 32,n,ConfBridge(1234) meetme.conf: (I am a bit confused here, as my conference room 1234 is defined only here) [general] audiobuffers=32 [rooms] conf => 1234 On 2010-11-20 19:36, Michael wrote: > Hello all, > > > Can anyone post a full working example of a configuration required to > setup a conference room on Asterisk 1.6.2.x, using ConfBridge? > > > Thank you in advance, > > > Michael > > -- _ -- 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] ConfBridge
Hello all, Can anyone post a full working example of a configuration required to setup a conference room on Asterisk 1.6.2.x, using ConfBridge? Thank you in advance, Michael -- _ -- 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 CDR Update (with set variable) problem.
hello,
First of all i am using Asterisk 1.6.2.9-2
The following problem seem like a bug to me but im not sure.
Any help or comment will be great..
We are trying to implement our own billing software with AGI - Php Scripts.
When a hangup received, i am calling a script to calculate the bill and we
are trying to write the results to cdr database.
We added two fileds to cdr table as rate and cost.
When we call a number and the other party doesnt hang up, rate (the targets
rate according to our rates table) and cost (calculated cost according to
rate and billable seconds) is written to database correctly.(sure cost is 0
because of hangup.)
But, if other party picks up the phone, and makes the call, even at debug of
php, i can see all the data correctly, it doesnt update the cdr fileds
correct.
Here is the php script output and cdr output of the calls.
No answer:
aftercall.php: [agi_context] => DLPN_WorldcallDial
aftercall.php: [agi_extension] => h
aftercall.php: [agi_priority] => 2
aftercall.php: [agi_enhanced] => 0.0
aftercall.php: [agi_accountcode] =>
aftercall.php: [agi_threadid] => 139906110289680
aftercall.php: >code<: 200
aftercall.php: >result<: 1
aftercall.php: >data<:
aftercall.php: >code<: 200
aftercall.php: >result<: 1
aftercall.php: >data<: 0.04830
aftercall.php: >>rate:0.04830
aftercall.php: >>duration:
-- AGI Script Executing Application: (set) Options: (CDR(userfield)=0)
-- AGI Script Executing Application: (set) Options: (CDR(rate)=0.04830)
-- AGI Script Executing Application: (set) Options: (CDR(cost)=0)
aftercall.php: cost:0
-- AGI Script aftercall.php completed, returning 0
and at the cdr rate is filled as 0.04830 and cost is 0 as it should be.
This is answered state:
aftercall.php: [agi_context] => DLPN_WorldcallDial
aftercall.php: [agi_extension] => h
aftercall.php: [agi_priority] => 2
aftercall.php: [agi_enhanced] => 0.0
aftercall.php: [agi_accountcode] =>
aftercall.php: [agi_threadid] => 139906110289680
aftercall.php: >code<: 200
aftercall.php: >result<: 1
aftercall.php: >data<: 5
aftercall.php: >code<: 200
aftercall.php: >result<: 1
aftercall.php: >data<: 0.04830
aftercall.php: >>rate:0.04830
aftercall.php: >>duration:5
-- AGI Script Executing Application: (set) Options:
(CDR(userfield)=0.0483)
-- AGI Script Executing Application: (set) Options: (CDR(rate)=0.04830)
-- AGI Script Executing Application: (set) Options: (CDR(cost)=0.0483)
aftercall.php: cost:0.0483
But the cdr rate and cost fields are both "0".
And this is the php scripts filed update part.
$agi->exec("set", "CDR(userfield)={$cost}");
$agi->exec("set", "CDR(rate)={$rateArray['data']}");
$agi->exec("set", "CDR(cost)={$cost}");
$agi->conlog("cost:". $cost);
--
_
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[asterisk-users] sip attended transfer beep
Hi All, I see some patches about adding atxfer beep sound in the sip channel, but I'm not clear on when this was implemented in what version? I don't see the added function in chan_sip in 1.2.24 or 1.4.21 or 1.6.0.28? Where is this code implemented, what stable release? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Make call in AMI.
>
> [Nov 19 16:49:56] NOTICE[23371]: chan_local.c:655 local_alloc: No such
>> extension/context 04191028...@intermovel creating local channel
>>
> can you display interMovel context ?
> Is there any entry matching 0419102889 in interMovel context ?
>
Oh! Like I'm stupid. This basic detail went totally unnoticed. Thanks for
the reply and sorry for my fault. The context:
context interMovel {
_00XX[7-9]XXX => { &saidaGSM(${EXTEN:1}); }
h => { &hangupGlobal(); }
}
It worked properly when I added one more zero. Thanks again.
At,
--
Rodrigo Lang
Opening your mind - Just another Open Source
site
Re: [asterisk-users] FFA (Fax For Asterisk) tif file (size) problem
Hi Michael, Use spandsp. It is more relaxed about the file resolution, to avoid this exact issue. Files with a resolution within 5% of 204x196 are accepted. However, if you have really made the image width 1680 pixels, that is wrong and I would be surprised if any FAX software accepts it. Standard sized FAX images are 1780 pixels wide. Steve On 11/20/2010 06:02 PM, Michael wrote: > Hi, > > We played around with the different parameters of the tif files and > found that the issue was with the resolution. > > Most files generated on the PC have a 200x200 resolution, but it seems > that FFA only accepts 204x196 resolution. > > Right now, we added a process to change the file resolution using > ImageMagick, but it would make sense to allow also 200x200. > > Michael > > Original Message > Subject: Re: [asterisk-users] FFA (Fax For Asterisk) tif file (size) problem > From: Mark Deneen > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Date: Friday, 19 November, 2010 17:19:48 > >> On Fri, Nov 19, 2010 at 9:42 AM, Michael wrote: >>> Hello, >>> >>> We succeed to send faxes using FFA, when the files are converted to tif >>> from PDF using gs, but it doesn't work with tif files we copy/upload >>> directly from our PCs. >>> >>> We saw in the manual that the size is important, since we got the error >>> "FAX handle 0: failed to queue document 'filename.tif'", so we set it to >>> 1680x2285, but it's still rejected. >>> >>> Is there a way to debug this further and fix it? We often have tif >>> source files that we prefer to send, without converting to pdf and back >>> to tif. >>> >>> Thank you in advance, >>> >>> Michael >>> >> I don't know if this is the case or not, but check for differences >> between the two tiff files. I wonder if one is compressed and the >> other is not? >> >> -M >> > -- _ -- 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] Installing Asterisk to it's own directory
What you did is what I would have done. That way the executables have their conf file location adjusted and everything will be inside the specified --prefix location. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 20, 2010, at 5:48 AM, Stephen Brown wrote: > Thanks... I actually did a ./configure --prefix=/root/asterisk18 and > ended up with this: -- _ -- 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] Installing Asterisk to it's own directory
Thanks... I actually did a ./configure --prefix=/root/asterisk18 and ended up with this: r...@debian-squeeze:~/asterisk18# pwd /root/asterisk18 r...@debian-squeeze:~/asterisk18# ls -al total 32 drwxr-xr-x 8 root root 4096 Nov 19 18:09 . drwx-- 5 root root 4096 Nov 19 18:37 .. drwxr-xr-x 3 root root 4096 Nov 19 18:09 etc drwxr-xr-x 3 root root 4096 Nov 19 18:09 include drwxr-xr-x 3 root root 4096 Nov 19 18:09 lib drwxr-xr-x 2 root root 4096 Nov 19 18:09 sbin drwxr-xr-x 3 root root 4096 Nov 19 18:09 share drwxr-xr-x 6 root root 4096 Nov 19 18:09 var Have I essentially accomplished the same thing by doing it this way? This is in a virtual machine alongside an Asterisk 1.6 install (for testing), I'm still a little gunshy to touch my production box as of yet. but the 1.8 install did work, I was able to make a call to the demo context :) Thanks, Stephen On 11/19/10 10:13 PM, Jose P. Espinal wrote: > Hi Stephen, > > That's what people do when building precompiled packages for certain > distros (along with a few more things). > > I use to do the following when building packages (with a few more options): > > ./configure --prefix=/usr --sysconfdir=/etc > make > make install DESTDIR=/my/destination/directory > > That would create the complete installation structure under > '/my/destination/directory' > > > Regards, > > > > Stephen Brown wrote: >> I'd like to start playing with 1.8, however I don't want to potentially >> damage anything on my existing 1.6.2 install on my production server. >> >> I'd like to test 1.8 against my existing configs leaving my 1.6.2 >> install untouched. Looking at the output of ./configure --help suggests >> that it's possible to install Asterisk into another prefix of my >> choosing, but as this is unfamiliar territory to me I'm not exactly sure >> how to accomplish this? >> >> Ideally, I'd like to just dump the newly compiled 1.8 and all it's >> dependencies into a standalone directory (say /testing/asterisk or >> something) and update my init script to point to the new binaries. I >> also run a Sangoma USB FXO card and DAHDI for a POTS line that I would >> like to test as well, should it work with the pre-compiled binaries that >> are already there? (DAHDI, etc) >> >> I've never tried this before, and before I potentially break something >> I'd like to know if it's possible and how to implement it? >> >> Thanks, >> Stephen >> >> >> -- _ -- 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] Ekiga can register but not my IP phone
On Sat, Nov 20, 2010 at 5:31 AM, Benoit Chabrier wrote: > Thanks for your help. > > you were right it also work without a stun server adding to sip.conf: > externip=78.47.x.x ; in [general] the IP of the dedicated server > nat=yes ; in the description of my peer > > Exactly. BTW, IAX doesn't have such problems ;-) but sadly not too many devices support it. Especially if you plan to connect your Asterisk boxes, even though you can use SIP, it's _strongly_ advisable you use IAX instead. -- _ -- 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] Ekiga can register but not my IP phone
Thanks for your help. you were right it also work without a stun server adding to sip.conf: externip=78.47.x.x ; in [general] the IP of the dedicated server nat=yes ; in the description of my peer 2010/11/19, Alejandro Imass : > On Fri, Nov 19, 2010 at 8:23 AM, Benoit Chabrier wrote: >> Thanks Alejandro, you were right it was just a NAT problem ! i add a >> stun server in the phone configuration and it works :) >> > > Cool. Also Asterisk SIP conf file has some NAT settings as well that > you can play with and perhaps do away with the stun server config in > the phone. Here is a great article that explains in detail the issues > with SIP and NAT: http://www.voipuser.org/forum_topic_7295.html > >> 2010/11/19, Alejandro Imass : >>> On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier wrote: Hello, I have a Sip phone (Siemens C470IP) which works perfectly with > > -- > _ > -- 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] FFA (Fax For Asterisk) tif file (size) problem
Hi, We played around with the different parameters of the tif files and found that the issue was with the resolution. Most files generated on the PC have a 200x200 resolution, but it seems that FFA only accepts 204x196 resolution. Right now, we added a process to change the file resolution using ImageMagick, but it would make sense to allow also 200x200. Michael Original Message Subject: Re: [asterisk-users] FFA (Fax For Asterisk) tif file (size) problem From: Mark Deneen To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Friday, 19 November, 2010 17:19:48 > On Fri, Nov 19, 2010 at 9:42 AM, Michael wrote: >> Hello, >> >> We succeed to send faxes using FFA, when the files are converted to tif >> from PDF using gs, but it doesn't work with tif files we copy/upload >> directly from our PCs. >> >> We saw in the manual that the size is important, since we got the error >> "FAX handle 0: failed to queue document 'filename.tif'", so we set it to >> 1680x2285, but it's still rejected. >> >> Is there a way to debug this further and fix it? We often have tif >> source files that we prefer to send, without converting to pdf and back >> to tif. >> >> Thank you in advance, >> >> Michael >> > > I don't know if this is the case or not, but check for differences > between the two tiff files. I wonder if one is compressed and the > other is not? > > -M > -- _ -- 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] Make call in AMI.
2010/11/19 Rodrigo Lang > > > [Nov 19 16:49:56] NOTICE[23371]: chan_local.c:655 local_alloc: No such > extension/context 04191028...@intermovel creating local channel > can you display interMovel context ? Is there any entry matching 0419102889 in interMovel context ? -- _ -- 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] Installing Asterisk to it's own directory
On Fri, 19 Nov 2010, Jose P. Espinal wrote: > Hi Stephen, > > That's what people do when building precompiled packages for certain > distros (along with a few more things). > > I use to do the following when building packages (with a few more options): > > ./configure --prefix=/usr --sysconfdir=/etc > make > make install DESTDIR=/my/destination/directory > > That would create the complete installation structure under > '/my/destination/directory' Also make sure that /my/destination/directory/etc/asterisk/asterisk.conf has the right paths in it and you start asterisk with a -C flag to point it to the conf file. Gordon > > > Regards, > > > > Stephen Brown wrote: >> I'd like to start playing with 1.8, however I don't want to potentially >> damage anything on my existing 1.6.2 install on my production server. >> >> I'd like to test 1.8 against my existing configs leaving my 1.6.2 >> install untouched. Looking at the output of ./configure --help suggests >> that it's possible to install Asterisk into another prefix of my >> choosing, but as this is unfamiliar territory to me I'm not exactly sure >> how to accomplish this? >> >> Ideally, I'd like to just dump the newly compiled 1.8 and all it's >> dependencies into a standalone directory (say /testing/asterisk or >> something) and update my init script to point to the new binaries. I >> also run a Sangoma USB FXO card and DAHDI for a POTS line that I would >> like to test as well, should it work with the pre-compiled binaries that >> are already there? (DAHDI, etc) >> >> I've never tried this before, and before I potentially break something >> I'd like to know if it's possible and how to implement it? >> >> Thanks, >> Stephen >> >> >> > > -- > Jose P. Espinal > http://www.eslackware.com > IRC: [OFTC|FreeNode] > Khratos @ #slackware | #asterisk/-doc/-bugs > > > -- > _ > -- 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
