[asterisk-users] Chan_mobile instability
Hi all, I'm testing chan_mobile for a couple of months and I'm facing some instability problems. I would appreciate if somebody could help me with these issues: - after a call the bluetooth connection disconnects; - when I make a outgoing call and the other side answer, sometimes asterisk is not informed, so it continues to ring my side but the other side can already hear my voice. asterisk-1.6.0.15 asterisk-addons-1.6.0.3 tks -- Rafael S. Seste ___ -- 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] Prevent cell phone voice mail capturing call
Hi, I've a DID number that gets passed to three internal phones and a cell phone via my outbound IAX trunk. If the cell phone is off or out of coverage, its voice mail captures the call. What's the best way to avoid this? Is there a recommended way to force the cell phone user to press 1 before the call is passed there ala google voice? Or is there another way to detect the presence of the answering machine rather than a human? Thanks, Russell. ___ -- 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] Prevent cell phone voice mail capturing call
You can dial the cell like this Dial(DAHDI/1c/w5551212) instead of Dial(DAHDI/1/w5551212) The 'c' makes the other end press 1 to start the call. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Horn Sent: Thursday, November 05, 2009 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Prevent cell phone voice mail capturing call Hi, I've a DID number that gets passed to three internal phones and a cell phone via my outbound IAX trunk. If the cell phone is off or out of coverage, its voice mail captures the call. What's the best way to avoid this? Is there a recommended way to force the cell phone user to press 1 before the call is passed there ala google voice? Or is there another way to detect the presence of the answering machine rather than a human? Thanks, Russell. ___ -- 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] RTP Proxy
Hi all, I would like to ask please how to configure asterisk in order to unforce rtp traffic to pass through it and send them to a separate RTp proxy? Regards ___ -- 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] Prevent cell phone voice mail capturing call
On Thu, Nov 5, 2009 at 3:51 PM, Danny Nicholas da...@debsinc.com wrote: You can dial the cell like this Dial(DAHDI/1c/w5551212) instead of Dial(DAHDI/1/w5551212) Danny - thanks, however I think that's a feature of DAHDI. My outbound trunk is IAX. I don't think that's a standard feature of the dial command. Has anyone else re-implemented it for other channels? Russell. ___ -- 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] SIP 503 instead of SIP 480 in asterisk debug mode
Hi All,
I was actually trying to use the dialplan application that uses 'Dial' and
when the: Dial(SIP/xxx...@|20|) command is executed and the
destination number rings for 20 sec after which I receive as 503 Service
Unavailable, but not 480 Temporarily unavailable.
Dial(SIP/xxx...@|20|)
exten = XX,n,NoOp(Dialstatus:${DIALSTATUS})
exten = XX,n,Congestion
I can see that the DialStatus is NoAnswer but sends the 503 Service
unavailable message instead of 480 Temporarily Unavailable. Is there any
way of trying to get as 480 Temporarily available as this is the industry
standard for 'NoANSWER' ?
Thank you very much for your help.
Best Regards
Sandesh
___
-- 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] MeetMe thinks DAHDI is missing 1.6.0.10
Hi, I've noticed that my MeetMe install seems to think chan_dahdi is missing: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) However, it definitely is since I have 3 PRIs functioning normally :) Is there anything I should check before I restart asterisk this evening to see if that fixes it? Thanks. -- James ** Please CC me on all responses. Thanks! ___ -- 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] faxes received on mISDN
On Thu, Nov 5, 2009 at 6:27 AM, Vieri rentor...@yahoo.com wrote: Despite the simpler setup, the faxes don't come in. From the logs I can see that Asterisk receives fax calls and dials the iaxmodem (on localhost). However, no data is transmitted according to Hylafax. Modify your dialplan to record the calls. Listen to the recording. Does the call ever connect? Does it sound like garbage? When you can hear what's going wrong you should be able to make better guesses. ___ -- 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] 7777 *65
can you define are not working? I just tried it on my cell phone and doesn't work either. Probably because ATT didn't define them. 2009/11/5 Torintino T torinti...@hotmail.com: I found and *65 are not working. Please how can i re-enable them again. Thanks Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. ___ -- 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] Asterisk 1.4 DISA is jumoing after one digit in the DISA context
Dear list, I have problems with DISA on an specific server with Asterisk 1.4.26.2. After starting DISA I can only press one key and DISA is jumping direct into the context without waiting for further digits. In dtmf.log I found this: [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin '7' received on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin passthrough '7' on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF end '7' received on SIP/214-00d92db0, duration 60 ms [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF end accepted with begin '7' on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF end '7' has duration 60 but want minimum 80, emulating on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF end emulation of '7' queued on SIP/214-00d92db0 If Iam using the dialplan on another server there is no problem. If Iam using READ I do not have problems to enter digits by DTMF so I assume its related to DISA. best regards Marc ___ -- 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] programming phones
Ott Rose wrote: I have question thats not really about astrisk but I figure you guys are doing this sort of thing. We use Aastra 6757i phones. there is some support for XML. the question is how would i go about learning to customize these phones? Read the manual on the Aastra website. It's actually quite comprehensive and not really directly related to Asterisk. It would however be interesting to see examples of what other people are doing with the XML on Aastra's or other applications on Polycoms. There was a guy at the Polycom booth during Astricon that had a very cool medical application using the Polycom VVX1500 phones. Darrick ___ -- 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] How to resell my trunk/provider to others?
Thank you Tarek!
I now know is possible.. i was a little confuse about it. Your
response gave some guidance.. all i have to do now is get the details.
I will be using A2Billing for this.
Regards,
Carlos C
On Nov 5, 2009, at 6:48 AM, Tarek Sawah wrote:
If you are a FreePBX user then i would suggest that you have a look
at A2billing. it's a great tool for RETAIL if you are selling to
individuals.
BUT if you are doing a termination business.. which i suspect is the
situation here .. then first of all and before offering any
assistant .. i would advise you to secure your money .. many of
companies that buy termination services from people do not pay on
time .. and sometimes they don't pay at all.
if you have secured your money the second step is easy.. get the IP
address of the person sending you calls.. and create a peer in your
sip.conf permitting their IP to send calls through your system.. and
use a different context than the one you are using for your local
users which is from-internal
and then identify that new context in your extensions.conf or in
your case i would suggest using the extensions_custom.conf so if
your context for example was [from-client] your dialplan is simple
###
[from-client]
exten = _.,1,Dial(SIP/GATEWAYIP/${EXTEN})
exten = _.,n,Hangup
###
that is all about it.. that's how i solve similar situation.. but my
advice is never use Asterisk in Termination.. it doesn't offer you
the best solution.. as Asterisk breaks the call into two combined
streams or Legs as some asterisk Gurus love to call it.
if you can't afford a Quimtum Gateway or Cisco .. you can still work
with Asterisk for a while. Please remember that asterisk requires a
lot of resources for the Encoding when using G729 and G723 codecs
that are widely used in Termination.
hope this put some light on your request..
have a nice day.
-- AHD Tarek Sawah Integrated Digital Systems CCNP, MCSE, RHCE, VoIP
Syria: +963 944 618286 USA: +1 347 562 2308
From: li...@latinbits.com
Date: Wed, 4 Nov 2009 14:43:55 -0500
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to resell my trunk/provider to others?
Hello,
I've been tasked to look for ways to resell to others the service
that
one of a trunk provides.. In other words, i want to configure my
current Asterisk (Ver. 1.4.26.1) with Freepbx 2.6.0 so i can act
as a
trunk to others.. I would provide an IP to them from one of my
servers
and they will use that IP to connect to me and i will connect them
to
my trunk/provider.
If possible, please provide some guidance as to where to start or a
link since i searched in google with no valuable results.. Maybe am
looking incorrectly.
Regards,
Carlos
___
-- 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
Windows 7: Unclutter your desktop. Learn more.
___
-- 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] How to resell my trunk/provider to others?
Hello,
Yeah i will be using asterisk and will be getting a Core 2 Due for the
production server.
Thanks,
Carlos C.
On Nov 5, 2009, at 8:21 AM, B.Masoud @ SH wrote:
Hello,
I am doing termination for about a year, I have used quintum 24
ports for termination, compared to asterisk with digum 24 ports too,
it’s SHIT, just use powerful PC like dual core cpu /2gb ram, u will
never notice any latency or echo.
Regards,
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Tarek Sawah
Sent: Thursday, November 05, 2009 2:49 PM
To: Asterisk Users
Subject: Re: [asterisk-users] How to resell my trunk/provider to
others?
If you are a FreePBX user then i would suggest that you have a look
at A2billing. it's a great tool for RETAIL if you are selling to
individuals.
BUT if you are doing a termination business.. which i suspect is the
situation here .. then first of all and before offering any
assistant .. i would advise you to secure your money .. many of
companies that buy termination services from people do not pay on
time .. and sometimes they don't pay at all.
if you have secured your money the second step is easy.. get the IP
address of the person sending you calls.. and create a peer in your
sip.conf permitting their IP to send calls through your system.. and
use a different context than the one you are using for your local
users which is from-internal
and then identify that new context in your extensions.conf or in
your case i would suggest using the extensions_custom.conf so if
your context for example was [from-client] your dialplan is simple
###
[from-client]
exten = _.,1,Dial(SIP/GATEWAYIP/${EXTEN})
exten = _.,n,Hangup
###
that is all about it.. that's how i solve similar situation.. but my
advice is never use Asterisk in Termination.. it doesn't offer you
the best solution.. as Asterisk breaks the call into two combined
streams or Legs as some asterisk Gurus love to call it.
if you can't afford a Quimtum Gateway or Cisco .. you can still work
with Asterisk for a while. Please remember that asterisk requires a
lot of resources for the Encoding when using G729 and G723 codecs
that are widely used in Termination.
hope this put some light on your request..
have a nice day.
-- AHD Tarek Sawah Integrated Digital Systems CCNP, MCSE, RHCE, VoIP
Syria: +963 944 618286 USA: +1 347 562 2308
From: li...@latinbits.com
Date: Wed, 4 Nov 2009 14:43:55 -0500
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to resell my trunk/provider to others?
Hello,
I've been tasked to look for ways to resell to others the service
that
one of a trunk provides.. In other words, i want to configure my
current Asterisk (Ver. 1.4.26.1) with Freepbx 2.6.0 so i can act
as a
trunk to others.. I would provide an IP to them from one of my
servers
and they will use that IP to connect to me and i will connect them
to
my trunk/provider.
If possible, please provide some guidance as to where to start or a
link since i searched in google with no valuable results.. Maybe am
looking incorrectly.
Regards,
Carlos
___
-- 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
Windows 7: Unclutter your desktop. Learn more.
___
-- 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] Prevent cell phone voice mail capturing call
Russell Horn wrote: Hi, I've a DID number that gets passed to three internal phones and a cell phone via my outbound IAX trunk. If the cell phone is off or out of coverage, its voice mail captures the call. What's the best way to avoid this? Is there a recommended way to force the cell phone user to press 1 before the call is passed there ala google voice? Or is there another way to detect the presence of the answering machine rather than a human? Thanks, Russell. Require the cell phone user to press a button to accept the call (much the same way that the followme app does). ___ -- 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] asterisk,libpri,zaptel
hi all, i have started installing asterisk on CENTOS 5.3. i have to install these three under /usr/src asterisk-current.tar.gz libpri-current.tar.gz zaptel-current.tar.gz now i had installed ASTERISK-1.6.1.9.TAR.GZ LIBPRI-1.4.10.2.TAR.GZ ZAPTEL IS PENDING guys plz suggest me which version is stable for all these three and i am not able to install zaptel from asterisk.org___ -- 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] Question about callerid?
Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). Anyhow, I can see that the gateway is passing caller id info to asterisk because the console will display something like: [Nov 4 13:01:19] NOTICE[32]: chan_sip.c:13362 handle_request_invite: Failed to authenticate user SEATTLE SCHOOLS sip:2062524...@89.89.89.253 ;tag=1c492497235 So the caller ID info is right there. However on my extensions (or softphones) the id shows as the extension # (ie 2003). Is there something I need to do to set the callerid? I can't seem to find this in the examples? Thanks in advance for helping with my (I am sure) stupid question... Marty ___ -- 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] Prevent cell phone voice mail capturing call
On 6/11/09 3:25 PM, Darrick Hartman wrote: Russell Horn wrote: Hi, I've a DID number that gets passed to three internal phones and a cell phone via my outbound IAX trunk. If the cell phone is off or out of coverage, its voice mail captures the call. What's the best way to avoid this? Is there a recommended way to force the cell phone user to press 1 before the call is passed there ala google voice? Or is there another way to detect the presence of the answering machine rather than a human? Thanks, Russell. Require the cell phone user to press a button to accept the call (much the same way that the followme app does). In fact it sounds like what he's actually wanting is the followme app: http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Asterisk 1.4 remote pickup
On 6/11/09 3:37 AM, Antony Stone wrote: On Thursday 05 November 2009 14:28, Danny Nicholas wrote: Hi. I have several Asterisk 1.4.21 machines, each with ISDN cards in them, and Polycom SIP phones on people's desks. I'm trying to work out how to provide a remote pickup facility along the following lines: The normal (as defined in features.conf) way to pick the call would be *82233. Features.conf defines *8 as a global pickup to be followed by an extension. Thanks, I'll investigate this and see if that works instead. What we do is create an Asterisk database entry: Pickup/NUMBER/GROUP Where NUMBER is the extension, and Group is the Pickup Group. We then set pickup mark variable in the macro that dials the extension. Then if someone dials *79 (or whatever) it picks up the group that the person dialling *79 is in. I.E. * Call goes to Jon (who is in group 3) * He is away from his desk * Jane dials *79 (also in group 3) and picks up the call If Fred (in group 5) were to dial *79 he would not pick up the call. Names have been changed to protect the innocent :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Asterisk-stat! - help needed (once again due to mailserver problem)
On 26/10/09 3:47 AM, Lukasz Pakula wrote:
Dear all,
I'm trying to install Asterisk-stat (ASTERISK CDR ANALYSER) following:
http://www.voip-info.org/wiki/index.php?page=Asterisk+CDR+Areski+GUI
however it fails to run properly - lots of lines like:
*Notice*: Undefined variable: s in
*/home/lukasz/DATA/www/asterisk-stat/cdr.php* on line *26*
*Notice*: Undefined variable: t in
*/home/lukasz/DATA/www/asterisk-stat/cdr.php* on line *27*
That's not an error - it's a notice - it means you have error_reporting
set to E_ALL in php.ini.
Depending on which version of Linux you use the file could be in a few
places.
If you are using Debian it would be in:
/etc/php5/apache2/php.ini
You'll need to restart Apache after changing the setting.
If you're brave you could surround the lines creating the problem with:
if (isset($s)) {
// Do something with $s
}
(replacing the commented line // with the line in question)
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
___
-- 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] Concurrent calls including mysql taking lot of time for execution
On 23/10/09 6:11 AM, jonas kellens wrote: On Thu, 2009-10-22 at 13:45 +1300, Matt Riddell wrote: It's really simple you just read from standard input and write to standard output. If you tell us a programming language you'd like to use (i.e. php/c/perl/bash etc) we can give you a link to some docs and examples. Might I highjack this thread to ask for this documentation ? I want to use php. :) Sorry been moving house for the week - easiest one to use for PHP is PHPAGI: http://phpagi.sourceforge.net/ -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] IAX jitterbufer oddity
On 27/10/09 2:07 AM, Steve Davies wrote: Hi, First a confession - The box in question is a 1.2.35 box, so this may be solved in a newer version as I know the JB code is all hugely changed, but... It may be worth checking into. Scenario: - IAX outbound call from Asterisk, which rings okay. - Remote end sends ANSWER, which we immediately ACK. - The ANSWER control packet gets put into the JB (that's how I read the code) - The remote end is clustered, and we receive a TXREQ within 1ms of our ACK - chan_iax2 starts to process the TXREQ correctly. What I think happens at this point is that the ANSWER control frame now leaves the JB in order, but is not processed because the channel state has moved into the new transferring state, so ANSWER has no meaning, app_dial never forwards the ANSWER control event to the calling channel, and the bridge is never fully completed, so it all eventually times out. Disabling the JB in IAX does resolve the issue, but is not ideal. I have tried to follow the code in the various versions 1.2, 1.4 and 1.6, but it is just too complicated. Does anyone know if this was addressed since 1.2, or can it still happen in 1.4 or 1.6? Just a shot - all boxes using NTP? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] OT - mISDN and B410P questions
On 25/10/09 11:52 AM, Paul Hales wrote: I have used both misdn and dahdi_bri over the last year, and would happy take dahdi if for no other reason that it's much easier to install. A patch is available to allow dahdi_bri to work with Asterisk 1.4, and I have used that successfully. Ooh really? Where would I find that? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] OT - mISDN and B410P questions
On 25/10/09 11:52 AM, Paul Hales wrote: I have used both misdn and dahdi_bri over the last year, and would happy take dahdi if for no other reason that it's much easier to install. A patch is available to allow dahdi_bri to work with Asterisk 1.4, and I have used that successfully. Which brings me to another question - what does Digium recommend people use on a 1.4 system with their b410p card these days? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] SendJabber question sending Links
On 5/11/09 9:14 PM, Stefan Schmidt wrote: Hello, i use sendjabber notifications when a call is answered to send the answering user information about the caller also with links to our CRM or ticket system. My problem is that i dont know how i can make a link like CRM and not have to use http://crm.x.y/fubar?user=1234. i´ve allready googled for this question, but i´ve only found how to xml format an url, but not how i can send it with sendjabber application. Does anybody have an idea how i can do this? It might pay to rephrase your question. You're trying to send a link, and what's going wrong? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] odbc to ms-sql server
Hi all, I'm trying to set up an odbc connection to a ms-sql server from an asterisk 1.6.1 install My problem is that I cannot get asterisk to build func_odbc res_odbc.so I installed yum -y install unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel And then went on to reconfigure / recompile asterisk after a ./configure --with-odbc=/usr/lib/ I get ### checking for mandatory modules: UNIXODBC... ok configure: creating ./config.status And then when I go to make menuselect; [XXX]Res_odbc [XXX] func_odbc [XXX] cdr_odbc Can anyone help out with what I am missing? [I've gotten to a stage where tsql and isql connections to my sql db work, however, getting odbc right is making me pull my hair out a bit] ___ -- 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] odbc to ms-sql server
2009/11/6 Neeraj Chand neeraj.ch...@ocis.com.au Hi all, I'm trying to set up an odbc connection to a ms-sql server from an asterisk 1.6.1 install My problem is that I cannot get asterisk to build func_odbc res_odbc.so I installed yum -y install unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel And then went on to reconfigure / recompile asterisk after a ./configure --with-odbc=/usr/lib/ I get ### checking for mandatory modules: UNIXODBC... ok configure: creating ./config.status And then when I go to make menuselect; [XXX]Res_odbc [XXX] func_odbc [XXX] cdr_odbc Can anyone help out with what I am missing? [I've gotten to a stage where tsql and isql connections to my sql db work, however, getting odbc right is making me pull my hair out a bit] ___ -- 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 Hello, Try use only ./configure . -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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] app read accept # sign
hello, I'm using Asterisk 1.6.0.5 . And I'm creating IVR, and I need that Read application accepts # sign, So is it possible? And maybe there is a workaround? Thanks -- Pagarbiai / Best Regards, Giedrius ___ -- 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] odbc to ms-sql server
Gotcha! Missed libtool! :) -Original Message- From: Neeraj Chand Sent: Friday, 6 November 2009 6:43 PM To: 'asterisk-users@lists.digium.com' Subject: RE: odbc to ms-sql server Hi all, I'm trying to set up an odbc connection to a ms-sql server from an asterisk 1.6.1 install My problem is that I cannot get asterisk to build func_odbc res_odbc.so I installed yum -y install unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel And then went on to reconfigure / recompile asterisk after a ./configure --with-odbc=/usr/lib/ I get ### checking for mandatory modules: UNIXODBC... ok configure: creating ./config.status And then when I go to make menuselect; [XXX]Res_odbc [XXX] func_odbc [XXX] cdr_odbc Can anyone help out with what I am missing? [I've gotten to a stage where tsql and isql connections to my sql db work, however, getting odbc right is making me pull my hair out a bit] ___ -- 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
