[asterisk-users] caller id not shown
hello all
i have asterisk 1.8.22 and have problem with caller id. this is my
scenario:
PSTN -- FXO --- FXS --- phone(223)
when i call from a 223 to another phone, every thing is ok and caller id
(223) is shown in called phone. but when i call from another phone to 223,
no caller id is shown and just zero is shown.
if i set callerid=12345 in chan_dahdi.conf file, when another phone call
223, this number (12345) is shown as caller id instead of zero. but i want
to show incoming number as caller id.
this is my chan_dahdi.conf file:
[channels]
;cidsignalling=dtmf
cidstart=polarity;; in gozine takhir dar tamas (aghab boodan yek zang) ra
az beyn mibarad.
callprogress=yes
usecallerid=yes
hidecallerid=no
callwaiting=no
transfer=yes
echocancel=yes
echotraining=yes
callerid=asreceived
group=0
callgroup=1
pickupgroup=1
usecallerid=yes
context=pstn-channels
channel=5-8
group=1
callgroup=1
pickupgroup=1
usecallerid=yes
context=phone-channels
channel=1-4
and this is my extensions.conf file:
[phone-channels]
exten=_.,1,Dial(DAHDI/8/${EXTEN})
[pstn-channels]
exten=_.,1,Dial(DAHDI/2/${EXTEN})
i searched a lot but found nothing useful:( please help me to solve it.
thanks in advance,
SAM
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Re: [asterisk-users] Meetme and maxusers option
Hi Johan. But the option maxusers should work too, right? On Fri, Jul 19, 2013 at 2:52 PM, Johan Wilfer li...@jttech.se wrote: 2013-07-19 15:35, Thiago Coutinho skrev: Hi all. I'm trying to limit the number of participants in a conference room with the realtime option maxusers, but it doesn't work. Someone have this option working properly? Try these: https://wiki.asterisk.org/wiki/display/AST/Function_GROUP https://wiki.asterisk.org/wiki/display/AST/Function_GROUP_COUNT This is how I do it. This way you can do it more flexible in the dialplan. -- Johan Wilfer -- _ -- 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 -- thiagoc O povo não deveria temer o governo. O governo é quem deveria temer o povo. V de Vingança -- _ -- 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] Fwd: Re: Asterisk T.38 Pass-Through doesn't work
On 22/07/2013 5:40 AM, Zoltán Fekete wrote: Hi! I have exactly the same problem on asterisk 1.8.22.0 and also on separate 11.2.1 when sending fax to PSTN. Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone. SpanDsp also works without any problem on my box. As I remember it was a bug in 1.8.1.x that the a=T38MaxBitRate paramater was sent as maxBitRate. Without capital M. Are you closer to the solution? I have tryed almost anything and I don't understand why sends the T38MaxBitRate:2400 parameter. Could it be because the remote endpoint does not supply the T38MaxBitRate attribute in its reply which then leads to Asterisk applying the Minimum Rate to your ATA!? I am referring to the information around lines 403 404 of https://gist.github.com/anonymous/5701207. Do you know what the Fax Rate was for the connection in https://gist.github.com/anonymous/5701150. What happens if you insert in your dialplan something like Set(FAXOPT(minrate)=4800) Cheers, Larry. -- _ -- 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] Set ringtone by dialed number
On Monday 22 July 2013, Josh Hopkins wrote: Would it be possible to set the ringtone based on the number that was dialed? If the phones you are using allow the ringing tone to be changed by sending a SIP header, yes. Example of what the goal is: Dial Denver number Incoming calls ring with ringtone 1 Dial main number Incoming calls ring with ringtone 2 But why would anyone want this? What is the point of changing the sound that your phone makes when someone calls you, depending on who you called last? We are currently using Digium D40, D50, D70 phones. -- AJS Answers come *after* questions. -- _ -- 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] Set ringtone by dialed number
Would it be possible to set the ringtone based on the number that was dialed? Example of what the goal is: Dial Denver number Incoming calls ring with ringtone 1 Dial main number Incoming calls ring with ringtone 2 We are currently using Digium D40, D50, D70 phones. -- _ -- 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 1.8 Service: -r does not give CLI
We have Asterisk1.8.11 and can not move to a newer version right now. But when we run Asterisk as a service, the -r option does not result in giving the CLI prompt? Did the option to get the CLI change? -- _ -- 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] Google Voice Calls Fail
A quick update. The nick: theory was proven to be wrong. The incoming calls consistently fail with or without nick: tag. I am concentrating on the incoming calls for now. -Vladimir On 7/21/2013 3:34 PM, Vladimir Mikhelson wrote: Hi All: Has anybody tackled the latest Google Voice issue where incoming and outgoing calls for certain Google Voice accounts fail? I have filed the bug report with details https://issues.asterisk.org/jira/browse/ASTERISK-22176 For incoming calls Asterisk does not reply to the initial XML request coming from Google Voice. Detailed comparison to a successful call initiation shows the lack of the nick: structure in the failed request. Outgoing calls connect intermittently, but no sound path gets established. Any ideas? Thank you, Vladimir -- _ -- 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] Turning off CFWD on an SPA112?
Hi all, I'm not sure how this happened, but one of my customers managed to turn call forwarding on on his spa112. I thought I had that turned off in the provisioning file. I have this in the provisioning file: Cfwd_All_Serv_1_No/Cfwd_All_Serv_1_ Cfwd_Busy_Serv_1_No/Cfwd_Busy_Serv_1_ And I have a similar entry for line 2. When I dial the device, I use this line to dial it: exten = 888,n,dial(sip/000E084B8xxx-1,60,Trit) Here is what I get on the server console: [Jul 19 15:49:42] -- Called sip/000E084B8xxx-1 [Jul 19 15:49:42] -- Got SIP response 302 Moved Temporarily back from 99.114.250.142:5060 [Jul 19 15:49:42] -- Now forwarding SIP/3CCE73D31786-1-00017bdb to 'Local/2146647512@magic' (thanks to SIP/000E084B8xxx-1-00017bdc) [Jul 19 15:49:42] -- Forwarding SIP/3CCE73D31xxx-1-00017bdb to 'Local/214xx@magic' prevented. [Jul 19 15:49:42] == Everyone is busy/congested at this time (1:1/0/0) [Jul 19 15:49:42] -- Auto fallthrough, channel 'SIP/3CCE73D31xxx-1-00017bdb' status is 'BUSY' So, my question is, how do I turn this feature off on this device? My customer really isn't in a position to be able to plug a handset into the device and dial *73. Can this be disabled from the provisioning file? Is there anything else I can do to prevent this? TIA, Mike Diehl. -- _ -- 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] Which is the stable version to use?
Hello I need to deploy asterisk on production and same thing for DAHDI, which version is recommended for this? Regards Bilal-- _ -- 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] Fwd: Re: Asterisk T.38 Pass-Through doesn't work
The a=T38MaxBitRate issue you refer to was one that was actually discovered at my company and submitted by a colleague. It was fixed in 11.3.0 and 1.8.21.0. However, I think that it wouldn't help based on the description below being that the parameter was missing altogether. I think if that parameter is missing, then the code would in fact default to 2400 as a safe value. Kevin Larsen - Systems Analyst From: Zoltán Fekete bl...@gyoz.info To: asterisk-users@lists.digium.com, Date: 07/21/2013 04:40 PM Subject:[asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work Sent by:asterisk-users-boun...@lists.digium.com Hi! I have exactly the same problem on asterisk 1.8.22.0 and also on separate 11.2.1 when sending fax to PSTN. Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone. SpanDsp also works without any problem on my box. As I remember it was a bug in 1.8.1.x that the a=T38MaxBitRate paramater was sent as maxBitRate. Without capital M. Are you closer to the solution? I have tryed almost anything and I don't understand why sends the T38MaxBitRate:2400 parameter. regards, Blaxy On 06/03/2013 05:03 PM, Larry Moore wrote: Have you checked the installed version of firmware against the latest available from Cisco? Oh! I didn't guess to check. The firmware was not fresh, but upgrading doesn't help. Looking at your SIP information when your ITSP initiated a T.38 session it did not indicate a maxmimum bitrate, it would appear your spa112 attempted to negotiate a connection at 2400bps. Whether there is a way to force my provider to indicate maximum bitrate? Do you have a sip debug session when you sent a fax from your Asterisk box to the PSTN, it would be interesting to see if it sends it as a t.38 or reverts to G711 audio. I have collect a set of debugs (with fresh SPA112 firmware) and actual config files: == spa112 — cmd ReceiveFax https://gist.github.com/anonymous/5701032 == cmd SendFax — PSTN https://gist.github.com/anonymous/5701150 == spa112 — PSTN https://gist.github.com/anonymous/5701207 == sip.conf https://gist.github.com/anonymous/5701231 == udptl.conf https://gist.github.com/anonymous/5701247 __ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com __-- _ -- 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] FTP server down?
Not sure if this is the right place to mention it, but . The server downloads.asterisk.org was refusing FTP connections last night, and still does not seem to be accepting them this morning. FTP may not be modern or trendy, but the ability to navigate around folders textually is nonetheless extremely handy when using a machine with no X server, from its own console. -- AJS Thanks for your concern and suggestions. Unfortunately, the decision was made to only offer our public downloads (e.g. downloads.digium.com, downloads.asterisk.org) over the HTTP protocol, not the FTP protocol, quite some time ago. As stated in Kevin Fleming's announcement on July 26th, 2007 to the Asterisk Announce mailing list, this was done primarily for reasons related to our marketing department ( http://lists.digium.com/pipermail/asterisk-announce/2007-July/85.html). If your response was misunderstood, please let us know and provide clarification. Thanks again! -- Chris Hozian Digium, Inc. | Network and Computer Systems Administrator 445 Jan Davis Drive NW - Huntsville, AL 35806 - US main: +1 256 428 6000 | fax: +1 256 864 0464 -- _ -- 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] Fwd: Re: Asterisk T.38 Pass-Through doesn't work
On 23/07/2013 6:18 AM, Kevin Larsen wrote: The a=T38MaxBitRate issue you refer to was one that was actually discovered at my company and submitted by a colleague. It was fixed in 11.3.0 and 1.8.21.0. However, I think that it wouldn't help based on the description below being that the parameter was missing altogether. I think if that parameter is missing, then the code would in fact default to 2400 as a safe value. The Case Insensitive checking of the T.38 attributes was introduced in these versions. Looking at the ITU-T T.38 Implementors' Guide (11 May 2012)in Table H.2, if the T38MaxBitRate attribute is omitted they suggest using the default value, they indicate a default value of 14400. Larry. -- _ -- 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] Fwd: Re: Asterisk T.38 Pass-Through doesn't work
On 22/07/2013 10:19 PM, Larry Moore wrote: On 22/07/2013 5:40 AM, Zoltán Fekete wrote: Hi! I have exactly the same problem on asterisk 1.8.22.0 and also on separate 11.2.1 when sending fax to PSTN. Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone. SpanDsp also works without any problem on my box. As I remember it was a bug in 1.8.1.x that the a=T38MaxBitRate paramater was sent as maxBitRate. Without capital M. Are you closer to the solution? I have tryed almost anything and I don't understand why sends the T38MaxBitRate:2400 parameter. Could it be because the remote endpoint does not supply the T38MaxBitRate attribute in its reply which then leads to Asterisk applying the Minimum Rate to your ATA!? I am referring to the information around lines 403 404 of https://gist.github.com/anonymous/5701207. Do you know what the Fax Rate was for the connection in https://gist.github.com/anonymous/5701150. What happens if you insert in your dialplan something like Set(FAXOPT(minrate)=4800) Another suggestion might be to set the variable in the peer configuration like; setvar=FAXOPT(minrate)=4800 Larry. -- _ -- 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] Google Voice Calls Fail
If anybody reads this thread here is the solution. It appeared to be some strange corruption of my Asterisk. As I started debugging and recompiled everything returned back to normal. What still puzzles me how some Google Voice accounts continued working all the time. -Vladimir On 7/22/2013 12:02 PM, Vladimir Mikhelson wrote: A quick update. The nick: theory was proven to be wrong. The incoming calls consistently fail with or without nick: tag. I am concentrating on the incoming calls for now. -Vladimir On 7/21/2013 3:34 PM, Vladimir Mikhelson wrote: Hi All: Has anybody tackled the latest Google Voice issue where incoming and outgoing calls for certain Google Voice accounts fail? I have filed the bug report with details https://issues.asterisk.org/jira/browse/ASTERISK-22176 For incoming calls Asterisk does not reply to the initial XML request coming from Google Voice. Detailed comparison to a successful call initiation shows the lack of the nick: structure in the failed request. Outgoing calls connect intermittently, but no sound path gets established. Any ideas? Thank you, Vladimir -- _ -- 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 -- _ -- 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
