[asterisk-users] ldap realtime function do not work in asterisk 1.8.11
[code]
res_ldap.conf
[_general]
;
; Specify one of either host and port OR url. URL is preferred, as you can
; use more options.
;host=192.168.1.1 ; LDAP host
host=lync-demo.local ; LDAP host
port=389
url=ldap://ad.lync-demo.local:389
protocol=3 ; Version of the LDAP protocol to use;
default is 3.
basedn=dc=lync-demo,dc=local ; Base DN
;user=cn=administrator,dc=lync-demo,dc=local ; Bind DN
user=cn=administrator,cn=users,dc=lync-demo,dc=local ; Bind DN
;user=dc=lync-demo,dc=local ; Bind DN
pass=Esi88
[extensions]
;context = AstExtensionContext
;context = givenname
;exten = AstExtensionExten
attribute=exten=givenname
;priority = AstExtensionPriority
;attribute=priority=givenname
;app = AstExtensionApplication
;appdata = AstExtensionApplicationData
additionalFilter=(objectClass=user)
[/code]
[code]
extconfig.conf
extensions = ldap,dc=lync-demo,dc=local,extensions
[/code]
[code]
[from-internal]
include = from-internal-xfer
include = bad-number
switch = Realtime/@extensions
exten= William,1,Set(CHANNEL(secure_bridge_media)=1)
exten= William,2,Set(_SIP_SRTP_SDES=1)
exten= William,3,Set(_SIPSRTP=optional)
exten= William,4,Set(_SIPSRTP_CRYPTO=enable)
exten = William,5,Set(b=${REALTIME(extensions,givenname,William)})
exten = William,6,NoOp(${b})
exten = William,7,Set(pair=${CUT(b,|,1)});
exten = William,8,Set(col_name=${CUT(pair,=,2)});
exten= William,n,Hangup()
[/code]
I use realtime to connect ldap server at lync
But When I query the ldap , I get below error in full log .
I expect the ldap query will get back something according input
givename=William .
The REALTiME function cannot retrevie the givename from lync and output null.
There is this key/attribute in lync server .
There is openration error .
The lync ldap server is working and I can use the filter
((objectClass=user)(givenname=William)) to get the result by
php ldap_Search .. it is work .
Please advice what is wrong in asterisk I use asterisk 1.8.11 ...
[Oct 30 00:42:48] DEBUG[9260] app_queue.c: Device 'SIP/3200' changed to state
'2' (In use) but we don't care because they're not a member of any queue.
[Oct 30 00:42:48] WARNING[9264] res_config_ldap.c: Failed to query directory.
Error: Operations error.
[Oct 30 00:42:48] WARNING[9264] res_config_ldap.c: Query:
((objectClass=user)(givenname=William))
[Oct 30 00:42:48] DEBUG[9264] pbx.c: Function result is '(null)'
[Oct 30 00:42:48] DEBUG[9264] pbx.c: Launching 'Set'
[Oct 30 00:42:48] VERBOSE[9264] pbx.c: -- Executing
[William@from-internal:5] Set(SIP/3200-, b=) in new stack
Log for asterisk full
https://dl.dropbox.com/u/68357652/full.rar--
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[asterisk-users] asterisk crashed on segmentation fault
Hello, I have a problem. One every couple of months my asterisk system crashes with a segmentation fault. kernel: asterisk[20527]: segfault at 0808 rip 2aaac952d8f2 rsp 40edb910 error 4 (This is in /var/log/messages) If I look at the same timestamp in the warning log file of asterisk (/var/log/asterisk/warning), I see that the are warning about fix up channel: WARNING[24000] chan_dahdi.c: Can't fix up channel from 1 to 2 because 2 is already in use WARNING[24000] chan_dahdi.c: Ringing requested on channel 0/2 not in use on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 5 to 6 because 6 is already in use WARNING[24000] chan_dahdi.c: Ringing requested on channel 0/6 not in use on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 6 to 8 because 8 is already in use WARNING[24000] chan_dahdi.c: Ringing requested on channel 0/8 not in use on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 3 to 4 because 4 is already in use WARNING[24000] chan_dahdi.c: Hangup REQ on bad channel 0/4 on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 2 to 3 because 3 is already in use WARNING[24000] chan_dahdi.c: Hangup REQ on bad channel 0/3 on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 4 to 5 because 5 is already in use WARNING[24000] chan_dahdi.c: Hangup REQ on bad channel 0/5 on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 1 to 2 because 2 is already in use WARNING[24000] chan_dahdi.c: Answer requested on channel 0/2 not in use on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 5 to 6 because 6 is already in use WARNING[24000] chan_dahdi.c: Answer requested on channel 0/6 not in use on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 1 to 2 because 2 is already in use WARNING[24000] chan_dahdi.c: Hangup on bad channel 0/2 on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 2 to 3 because 3 is already in use WARNING[24000] chan_dahdi.c: Can't fix up channel from 3 to 4 because 4 is already in use WARNING[24000] chan_dahdi.c: Can't fix up channel from 4 to 5 because 5 is already in use WARNING[24000] chan_dahdi.c: Can't fix up channel from 5 to 6 because 6 is already in use WARNING[24000] chan_dahdi.c: Hangup on bad channel 0/6 on span 1 WARNING[24000] chan_dahdi.c: Whoa, there's no owner, and we're having to fix up channel 6 to channel 8 WARNING[24000] chan_dahdi.c: Whoa, there's no owner, and we're having to fix up channel 1 to channel 2 WARNING[24000] chan_dahdi.c: Can't fix up channel from 5 to 6 because 6 is already in use [/size] Does anybody know what these messages mean? I use the following drives and asterisk: Asterisk 1.6.2.12 libpri 1.4.11.4-1_centos5 dahdi linux-2.4.0-1_centos5 We are using two Digium, Inc. Wildcard TE420P quad-span T1/E1/J1 card 3.3V (PCI-Express) (rev 02) Kind regards, Arjan Kroon -- _ -- 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] Add a variable to the destination channel without adding it to the source channel?
Hello all, I have the following challenge: I have to add a variable to the destination channel with the following conditions: 1) It has to be set in the dialplan, in runtime. 2) The source channel can't have the same variable has the destination. I had two ideas so far, but they seem complicated: 1) Add a local channel in the middle of the source and destination channel. With that I can use the dialplan application SET(__VAR='XXX') in the local and it will be inherited by the destination channel. 2) Call a macro or a gosub in the DIAL. With that macro or gosub I can add a variable in the destination channel. A more easy suggestion, please? :) Thanks in advance, Alex -- _ -- 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 crashed on segmentation fault
Check us out at: http://digium.com http://asterisk.org Arjan Kroon wrote: Hello, I have a problem. One every couple of months my asterisk system crashes with a segmentation fault. Normally if you are getting a segfault, that's a good reason to file a bug report. However... I use the following drives and asterisk: Asterisk 1.6.2.12 Using an old, unsupported version of Asterisk means that approach would probably be fruitless. I'd recommend you consider upgrading to either 1.8 or 11 at this point. If you absolutely must stick with 1.6.2 for whatever reason, then you can use 1.8 on a test system to see if you can reproduce your problem and if you can, file a bug report against that and hope the patch either translates well to 1.6.2 without much intervention or you could attempt to backport it yourself if it doesn't. Good luck with your problem. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 -- _ -- 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 and sip web client
On Sun, Oct 28, 2012 at 2:48 PM, giuseppe...@gmail.com wrote: Hello guys, I would like to use asterisk with a html sip web client. What asterisk version or particular question are required? If you're starting without any pre-existing configuration, it would be smart to use the current stable release of Asterisk. Ultimately, however, nearly all stable releases of Asterisk are appropriate for SIP, like nearly all models of cars are appropriate for driving on roads. This HTML SIP web client, is it something you're developing? Or something you have found existing? What are you trying to accomplish? Thanks, Regards -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] DTMF inband with telephone-event in SDP
Jakob Hirsch wrote: Hello everyone! Hola, We use Asterisk for various services like voicemail. Our SIP clients usually use rtp events (rfc2833) for DTMF, which works just fine and independent from the codec (g711 vs. g726 etc.). Now we noticed there are some SIP clients that announce telephone-event in their SDP, but send their DTMF inband. The problem with that is, that Asterisk obviously does not try to detect inband DTMF after seeing the telephone-event payload type in the SDP. Generally DTMF is something that has to be configured on both sides, you can't just configure it on one and have the negotiation force it to be that. So we are in a kind of dilemma: - dtmfmode=auto (and dtmfmode=rfc2833) will work for most, but not for the described ones. - dtmfmode=inband would also work for most, but of course not for the ones using g726 et al. Is there any Asterisk setting to force inband DTMF detection (with non-compressing codecs only, of course)? I browsed the code without result. Unfortunately there isn't a way to force this as you describe out of the box, you would have to make changes to chan_sip or explicitly have the clients configured properly. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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] motif and psi - no sound
Dmitry Melekhov wrote: 19.10.2012 08:40, Dmitry Melekhov пишет: Hello! Hola, I'm trying to use psi+ to conect to asterisk using chan_motif and vise versa. Connection looks good, but no sound. As I see there is some traffic (22.229 is my desktop with psi) 08:38:37.463506 IP 192.168.22.229.8010 192.168.22.19.17012: UDP, length 82 08:38:37.481325 IP 192.168.22.229.8010 192.168.22.19.17012: UDP, length 82 08:38:37.481885 IP 192.168.22.19.17012 192.168.22.229.8010: UDP, length 65 08:38:37.501745 IP 192.168.22.19.17012 192.168.22.229.8010: UDP, length 65 I see you've created an issue for this, thanks! I took a quick gander at the packet capture and it appears that the Jingle and RTP portion of things is perfectly fine. This may actually end up being a transcoding issue, but we'll see once the issue is assigned and researched. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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] Add a variable to the destination channel without adding it to the source channel?
Alexandre Rodrigues wrote: Hello all, Hola, I have the following challenge: I have to add a variable to the destination channel with the following conditions: 1) It has to be set in the dialplan, in runtime. 2) The source channel can't have the same variable has the destination. I had two ideas so far, but they seem complicated: 1) Add a local channel in the middle of the source and destination channel. With that I can use the dialplan application SET(__VAR='XXX') in the local and it will be inherited by the destination channel. 2) Call a macro or a gosub in the DIAL. With that macro or gosub I can add a variable in the destination channel. This is a perfect application for pre-dial handlers available in Asterisk 11. You can specify a GoSub routine to execute before dialing and in that routine you can set the variable you need. More information is available at https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers If you are going to use an earlier version you can use the 'U' option to execute a GoSub routine before the calling channel and answered channel are connected. The caveat with this is that your routine won't be executed before dialing actually occurs. Both of these may seem complicated to you because they don't expose a single option that just does exactly what you want. This is on purpose, we provide many tools to accomplish many different things as what people want to do varies greatly. I hope this helps! Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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
[asterisk-users] Bypass queue wrapup time
In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? -- Mitch -- _ -- 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] Bypass queue wrapup time
As I read the queues.conf.sample file I would say no since you would have to set the value to 0 and reload the queue. If you state your asterisk version and whether you're using realtime, someone might offer a solution. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn Sent: Monday, October 29, 2012 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bypass queue wrapup time In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? -- Mitch -- _ -- 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] Bypass queue wrapup time
I don't think you can. But you could set it to a lower value like 3 seconds and give your operators a feature key to pause themselves in the queue if they need extra work time. - Logs On Oct 29, 2012 12:15 PM, Mitch Claborn mitch...@claborn.net wrote: In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? -- Mitch -- __**__**_ -- 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-**usershttp://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] Bypass queue wrapup time
Asterisk 1.8 Not currently using realtime. Mitch On 10/29/2012 12:19 PM, Danny Nicholas wrote: As I read the queues.conf.sample file I would say no since you would have to set the value to 0 and reload the queue. If you state your asterisk version and whether you're using realtime, someone might offer a solution. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn Sent: Monday, October 29, 2012 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bypass queue wrapup time In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? -- _ -- 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] Bypass queue wrapup time
Since you're not using realtime, your best bet is probably going to be to give your operators a menu to increase/decrease their queue penalties and set your wrapup time low like the other poster suggested. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn Sent: Monday, October 29, 2012 1:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Bypass queue wrapup time Asterisk 1.8 Not currently using realtime. Mitch On 10/29/2012 12:19 PM, Danny Nicholas wrote: As I read the queues.conf.sample file I would say no since you would have to set the value to 0 and reload the queue. If you state your asterisk version and whether you're using realtime, someone might offer a solution. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn Sent: Monday, October 29, 2012 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bypass queue wrapup time In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.0 disable cdr account logs?
JR Richardson wrote:
My bad. I sent Igor to the boneyard to fetch 1.6.0.28 and it appears to me
that by commenting out lines 309-312 and doing a fresh make you eliminate
the extra files (or make them empty).
Appriciate the suggestion but commenting out 309-312 refused to compile:
cdr_csv.c
/* if (!ast_strlen_zero(cdr-accountcode)) {
if (writefile(buf, cdr-accountcode))
ast_log(LOG_WARNING, Unable to write CSV
record to account file '%s' : %s\n, cdr-a$
*/ }
You need to place the */ after the } or else they are mismatched and
like you have seen, the universe will explode.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
--
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Re: [asterisk-users] Add a variable to the destination channel without adding it to the source channel?
Thanks Joshua for the quick reply. Both of these may seem complicated to you because they don't expose a single option that just does exactly what you want. That's exactly what I was thinking. I know that asterisk provides tools to implement a lot of different things and because of this I was checking with the asterisk community to see if there where more easy ways of achieving it. From your reply I can see that, in asterisk 1.8, I wasn't complicating it too much. :) By the way, in Portuguese is Olá not Hola. ;) Cheers, On Mon, Oct 29, 2012 at 3:21 PM, Joshua Colp jc...@digium.com wrote: Alexandre Rodrigues wrote: Hello all, Hola, I have the following challenge: I have to add a variable to the destination channel with the following conditions: 1) It has to be set in the dialplan, in runtime. 2) The source channel can't have the same variable has the destination. I had two ideas so far, but they seem complicated: 1) Add a local channel in the middle of the source and destination channel. With that I can use the dialplan application SET(__VAR='XXX') in the local and it will be inherited by the destination channel. 2) Call a macro or a gosub in the DIAL. With that macro or gosub I can add a variable in the destination channel. This is a perfect application for pre-dial handlers available in Asterisk 11. You can specify a GoSub routine to execute before dialing and in that routine you can set the variable you need. More information is available at https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers If you are going to use an earlier version you can use the 'U' option to execute a GoSub routine before the calling channel and answered channel are connected. The caveat with this is that your routine won't be executed before dialing actually occurs. Both of these may seem complicated to you because they don't expose a single option that just does exactly what you want. This is on purpose, we provide many tools to accomplish many different things as what people want to do varies greatly. I hope this helps! Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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 -- _ -- 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] Read sometimes disconnects user
Hello, I am asking the user to enter his mobile phone followed by # using Read(). From time to time the Read() application disconnects the user while he is typing his number, though there is a 15 seconds timeout, and even if I type the number very fast it still may happen to me. *same = n,Read(mobileNumber,app/input-mobile,10,,2,15)* In the logs: When it fails: - - SIP/ipbx-iwred-02e Playing 'app/input-mobile.slin' (language 'fr') - - User disconnected When it succeeds: - - SIP/ipbx-iwred-02e Playing 'app/input-mobile.slin' (language 'fr') - - User entered '0476123456' The strange thing is that I cannot understand when it happens, it seems completely random... sometimes I type the mobile number very slowly, sometimes very fast, and it doesn't seem to matter. Anyone encountered such an error? Thank you for any ideas! -- _ -- 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] Read sometimes disconnects user
On Tue, 30 Oct 2012, Thomas Thomas wrote: I am asking the user to enter his mobile phone followed by # using Read(). From time to time the Read() application disconnects the user while he is typing his number, though there is a 15 seconds timeout, and even if I type the number very fast it still may happen to me. It has been my casual observation that the speed at which I enter digits on my phone is unrelated to the speed at which my cell provider delivers the digits to my Asterisk box. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
