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Hi Community, I''m currently trying to create a blog software in rails, but I''ve got a problem: I generated scaffolding for my articles and only want administrators to write, edit or delete articles. So I wanted to move this parts to another, secured controller. The one controller to display articles is called articles :), the other is also called articles, but is located in a

Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* *** -------------- next

Hi All, While I'm certainly comfortable compiling from sources, I'm trying to do an rpm only asterisk install on CentOS 5.7. I'm using the asterisk repositories and I installed all the asterisk18 rpms, but find that chan_gtalk and res_jabber are missing. Is there a separate rpm that includes support for gtalk? Thanks in advance. -Gaurav -------------- next part -------------- An

Hi, I'm trying to use Asterisk to place Automated Voice Calls. A verbose log from Asterisk CLI taken when I place a call in the spool directory looks like this: -- Attempting call on SIP/MTN-NEW/my-number for application MP3Player(/myfile) (Retry 1) == Using SIP RTP CoS mark 5 > Channel SIP/MTN-NEW-00000005 was never answered. [Jul 20 10:52:11] NOTICE[14580]: pbx_spool.c:339

Hello, I see a lot of these messages in the debug log : /[May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May 3 15:47:09] DEBUG[19081] audiohook.c: Read factory 0xae173e0 and write factory 0xae17e18 both fail to provide 160 samples [May 3

...ailto:llvm-dev at lists.llvm.org>" <llvm-dev at lists.llvm.org <mailto:llvm-dev at lists.llvm.org>> > Subject: [llvm-dev] Submitting patches for LLVM -- llvm-commits vs. Phabricator? > > Hi, > > I've recently submitted a patch to llvm-commits (as requested byhttps://llvm.org/docs/DeveloperPolicy.html#making-and-submitting-a-patch <https://llvm.org/docs/DeveloperPolicy.html#making-and-submitting-a-patch>) and the mailing list answered with a notice that my message is held for moderator approval (with the reason: "Post by non-member to a members-on...

Yup, Phabricator is generally preferred for patches. Additionally, are you subscribed to the mailing list? I can't find where I read it now, but I believe your messages are held for moderation if you aren't subscribed. You can subscribe at http://lists.llvm.org/mailman/listinfo/llvm-commits if needed. From: llvm-dev <llvm-dev-bounces at lists.llvm.org> on behalf of Christoph Kindl

Hi, I've recently submitted a patch to llvm-commits (as requested by https://llvm.org/docs/DeveloperPolicy.html#making-and-submitting-a-patch) and the mailing list answered with a notice that my message is held for moderator approval (with the reason: "Post by non-member to a members-only list"). I'm therefore wondering if I should've submitted my patch via Phabricator

Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen

Hello, notice in the console output beneath that there is a resultid 6 but it can not be cleared : [May 5 11:46:27] -- Executing [s at sub:3] MYSQL("SIP/vart-00000336", "Connect connid localhost dialplan host Asterisk") in new stack [May 5 11:46:27] -- Executing [s at sub:4] MYSQL("SIP/vart-00000336", "Query resultid 4 DELETE FROM pickuptbl WHERE

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

Hello, when using BLF with Asterisk 1.6, I notice that the Caller-ID information is not displayed on the monitoring key of my Innovaphone IP200A. If the IP-phone of my colleague rings, I should see on my partner key the number of the caller. This is information that is being send in the xml-body of the NOTIFY-message. I do not see this information in the xml-body of a NOTICE-message from

...sts.llvm.org>" <llvm-dev at lists.llvm.org <mailto:llvm-dev at lists.llvm.org>> >> Subject: [llvm-dev] Submitting patches for LLVM -- llvm-commits vs. Phabricator? >> >> Hi, >> >> I've recently submitted a patch to llvm-commits (as requested byhttps://llvm.org/docs/DeveloperPolicy.html#making-and-submitting-a-patch <https://llvm.org/docs/DeveloperPolicy.html#making-and-submitting-a-patch>) and the mailing list answered with a notice that my message is held for moderator approval (with the reason: "Post by non-member to a members-on...

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Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from

List users, Did anyone else recently receive spam from DIDForSale with the subject "DIDForSale 2012 achievements"? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the

Hi List, I have another issue on allowing outgoing calls to PSTN on Asterisk via Avaya Phones, I hope that anyone could help me fix this issue: *When I dial through Avaya phone i just here a "good bye message" reply from asterisk server. And here is the log:* == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back to exten 's' == Starting

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B.