Re: [asterisk-users] Mp3 for IVR prompts
On Saturday 10 October 2009 21:45:37 RSCL Mumbai wrote: Can I convert my .WAV IVR greetings, MOH and other recordings into G729 format to prevent transcoding and hence CPU usage ? You can do it with a one-time conversion. This will require a single g729 license: *CLI file convert foo.wav foo.g729 -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image
On 3 Oct 2009, at 20:38, James Stocks wrote: On 3 Oct 2009, at 16:37, Jonathan Thurman wrote: On Sat, Oct 3, 2009 at 6:17 AM, James Stocks stoc...@stocksy.co.uk wrote: Hi everyone, I hope someone can help me with a problem I'm having with Cisco 7940 phones on the SIP 8.12 image. When I place a call from one of the handsets, the call proceeds as normal for 20 seconds and is then terminated by Asterisk (1.4.26.2): We are runing 08-12-00 on 7940/60s just fine (Asterisk 1.6.1.1), and have been for a while. As far as I can tell, the 'a=silenceSupp:off - - - -' header is not accepted by the 7940, which seems like a bug in the SIP image to me. However, I can't find a way to report this problem to Cisco without a support contract (which I do not have). Reverting to version 7.5 fixes the problem, but it is still present in 8.11. The problem is not present if the PSTN initiates the call, nor is it present if I allow the handsets to reinvite each other. Here's the sip.conf snippet if it helps: That all looks fine to me. What do your SIPDefault.cnf and SIPMAC.cnf files look like? -Jonathan Hi Jonathan, Thanks for your reply. Here's the two files, SIPDefault.cnf: # Image Version image_version: P0S3-8-12-00 # Proxy Server proxy1_address: pabx.spruce # IP address here alternatively # Proxy Registration (0-disable (default), 1-enable) proxy_register: 1 # Setting for Message messages_uri: 222 # Time Server sntp_mode: unicast sntp_server: snakebite.spruce # IP address here alternatively time_zone: GMT dst_offset: 1 dst_start_month: March dst_start_day: dst_start_day_of_week: Sun dst_start_week_of_month: 4 dst_start_time: 02 dst_stop_month: Oct dst_stop_day: dst_stop_day_of_week: Sunday dst_stop_week_of_month: 4 dst_stop_time: 2 dst_auto_adjust: 1 date_format: D/M/Y # XML file that specifies the dialplan desired dial_template: dialplan #Time Format (0-12hr, 1-24hr [default]) time_format_24hr: 1 # URL for external Phone Services services_url: http://pabx.spruce/openxmldir/PhoneUI/index.php; # IP address here alternatively # URL for external Directory location directory_url: http://pabx.spruce/openxmldir/PhoneUI/index.php; # IP address here alternatively # URL for branding logo logo_url: http://pabx.spruce/cisco/asterisk.bmp; # IP address here alternatively and SIPmac.cnf: # Image Version image_version: P0S3-8-12-00 phone_label: # Line 1 appearance line1_displayname: James line1_shortname:200 James line1_name: 200 line1_authname: 200 line1_password: *removed* # Line 2 appearance line2_displayname: Work line2_shortname: 206 Work line2_name: 206 line2_authname: 206 line2_password: *removed* # Line 3 appearance line3_displayname: line3_shortname: line3_name: UNPROVISIONED line3_authname: UNPROVISIONED line3_password: UNPROVISIONED # Line 4 appearance line4_displayname: line4_shortname: line4_name: UNPROVISIONED line4_authname: UNPROVISIONED line4_password: UNPROVISIONED # Line 5 appearance line5_displayname: line5_shortname: line5_name: UNPROVISIONED line5_authname: UNPROVISIONED line5_password: UNPROVISIONED # Line 6 appearance line6_displayname: line6_shortname: line6_name: UNPROVISIONED line6_authname: UNPROVISIONED line6_password: UNPROVISIONED # Phone Prompt (The prompt that will be displayed on console and telnet) phone_prompt: SIP Phone ; Limited to 15 characters (Default - SIP Phone) # Phone Password (Password to be used for console or telnet login) phone_password: *removed* ; Limited to 31 characters (Default - cisco) # User classifcation used when Registering [ none(default), phone, ip ] user_info: none OK. For anyone finding this thread, the problem exists in Asterisk 1.4, but upgrading to Asterisk 1.6.1.6 appears to eliminate the problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mp3 for IVR prompts
On Sun, 11 Oct 2009 09:10:27 -0500, Tilghman Lesher wrote: On Saturday 10 October 2009 21:45:37 RSCL Mumbai wrote: Can I convert my .WAV IVR greetings, MOH and other recordings into G729 format to prevent transcoding and hence CPU usage ? You can do it with a one-time conversion. This will require a single g729 license: *CLI file convert foo.wav foo.g729 There are embedded Astrisk distros like Astlinux that target small form factor hardware. They have distributed the standard prompts in various encodings for quite some time. I think that they may have been amonst the first to do so. Platforms like the Soekris Net4801 could only sustain a couple of G.729 transcodes, making native prompts a practical necessity. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image
On Sun, Oct 11, 2009 at 8:03 AM, James Stocks stoc...@stocksy.co.uk wrote: OK. For anyone finding this thread, the problem exists in Asterisk 1.4, but upgrading to Asterisk 1.6.1.6 appears to eliminate the problem. Sorry, I lost your last response in my inbox... Your phone configs look fine. The only thing that we do differently is disable VAD on the phones. Never used 1.4, only the 1.6 branch. Glad to see that you got it working. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel problems on SuSE 9.3
- Original Message - From: Philipp Kempgen philipp.kemp...@amooma.de To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 05, 2009 10:39 PM Subject: Re: [asterisk-users] Zaptel problems on SuSE 9.3 Angus Asterisk schrieb: It seems that the zaptel startup script does not work. I noticed at startup the line: /etc/init.d/zaptel: line 40: /etc/init.d/functions: No such file or directory Just some feedback which might be helpful. The VIA box I am running on has an internal modem and I think that might have had a resource clash with the Digium board. So I disabled that plus other hardware devices I din't need in the bios. The zaptel startup script doesn't seem to work on suse so I added: modeprobe wctdm ztcfg -vvv asterisk in /etc/init.d/boot.local I think Suse is probably a bit over the top for asterisk. Too much stuff running. I might look at some really cut down linux distros. What do other people use? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Recording and Posting
Hello, I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. Many thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working
I just double checked the setting of the remote asterisk and it has the same setting as mine. Sip.conf has in Global: dtmfmode = rfc2833 individual extension has no dtmf setting at all, so global setting take precedence. All units Linksys, Sipura have DTMF Tx Method: Auto Linksys has an additional setting: DTMF Tx Mode: Strict My asterisk is using old Sipura units and dtmf tones to access voicemail are recognized. The remote asterisk is using newer Linksys units and dtmf to voicemail does not work, the phone hangs up. The strange part is: PSTN -- Asterisk (voicemail access) works OK on both sytemes. Asterisk (w/Linksys) -- Asterisk (w/Sipura) to Voicemail works OK Asterisk (w/Linksys) -- Asterisk (w/Linksys) to Voicemail DOES NOT work Asterisk (w/Sipura) -- Asterisk (w/Linksys) to Voicemail DOES NOT work So it seems to me the Linksys units don't work as they suppose to. -- Joseph On 10/11/09 01:27, Ivan Stepaniuk wrote: Joseph wrote: I've tried setting my asterisk dtmf to rfc2833, inband it is not working. The other Asterisk Linksys is set dtmf = auto If understand correctly, you have two asterisk servers and when you dial from one the other, DTMF is not recognized. I also asume you are using SIP to connect them as you mentioned dtmfmode. In any case, this should be set to the same value on both sides, both rfc2833, or both info. You don't wand inband and auto is just rfc2833 with automatic inband fallback. -- Iv?n Stepaniuk Alba Fot?nica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream 2010
The Grandstream 286s automatically re register when a connection is restored. Our Grandstream 2010s don't. Does anyone know of a setting that makes them reregister? I has tweaked Watchdog timer and anything that looked promising. Cary Fitch Affordable Telecom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Dan, You can do this directly in the dialplan. See the System command. It allows you to call any program on the system (ftp, scp, mv, etc). Keep in mind that depending on the volume of calls you're handling, you might run into I/O issues on the disk side. If you're talking about a machine under enough load, you might need another alternative. -Elliot From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Sunday, October 11, 2009 7:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call Recording and Posting Hello, I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. Many thanks Dan Journo This message is intended only for the use of the individual (s) or entity to which it is addressed and may contain information that is privileged, confidential, and/or proprietary to Calling Circles LLC and its affiliates. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution, forwarding or copying of this communication is prohibited without the express permission of the sender. If you have received this communication in error, please notify the sender immediately and delete the original message. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
On Mon, 12 Oct 2009, Dan Journo wrote: I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. There is no built in facility -- but there are all the parts. There is the curl() application, but I don't know if it exposes enough curl to upload files. There is the system() application which will let you execute any command line you can construct. There is the agi() application which lets an external program interact with the dialplan. Monitoring a temp directory with an external program would be the worst way. Personally, I would wrap up the entire call recording solution in an AGI so you have a full featured language (my preference is C) and can hide all the ugly details and keep your dialplan simple and maintainable. I've done these kind of applications where either a control file needed to be written and uploaded with the recording or a database needed updating. Both of these can get ugly in a dialplan. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4
We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on attended transfers. What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed on hold and extension B calls external third party C. After explaining caller A issue to Party C, Ext B brings Caller A onto the call and introduces A to C. After the into, ext B then drops off the call while A C continue the call. Any help would be appreciated. Thanks Much, Jeff Johnson This email and any attached files are confidential and intended solely for the intended recipient(s). If you are not the named recipient you should not read, distribute, copy or alter this email. Any views or opinions expressed in this email are those of the author and do not represent those of the company. Warning: Although precautions have been taken to make sure no viruses are present in this email, the company cannot accept responsibility for any loss or damage that arise from the use of this email or attachments.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4
I don't think this can be done. In your scenario, B is effectively the host and if B drops the line, both A and C will be dropped off as well. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff Johnson Sent: Monday, 12 October 2009 2:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4 We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on attended transfers. What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed on hold and extension B calls external third party C. After explaining caller A issue to Party C, Ext B brings Caller A onto the call and introduces A to C. After the into, ext B then drops off the call while A C continue the call. Any help would be appreciated. Thanks Much, Jeff Johnson This email and any attached files are confidential and intended solely for the intended recipient(s). If you are not the named recipient you should not read, distribute, copy or alter this email. Any views or opinions expressed in this email are those of the author and do not represent those of NeturallySpeaking, LLC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
