[asterisk-users] OT - Gigaset and PoE

2010-12-03 Thread Olivier
Hi, Gigaset products like C470IP are using 6.5V/600mA power supply and a power jack I would qualify as a 5,5-2,5mm one. I can see 5V or 7.5V PoE here and there. 1. Is there a norm attaching power jacks electric and mechanical specifications in telecom or networking devices ? From

Re: [asterisk-users] MP3s not decoding properly for MusicOnHold.

2010-12-03 Thread salaheddine elharit
Hello, in order to play the music in asterisk like a MusicOnHold you can convert it from MP3 to GSM Regards, 2010/12/2 Steve Edwards asterisk@sedwards.com On Thu, 2 Dec 2010, Ernie Dunbar wrote: I have some MP3 files that play well in any MP3 player I throw at them, but when I try to

[asterisk-users] Caller id is not proper when I do call forward

2010-12-03 Thread Nikhil
Hi Caller id is not show showing proper when I do call forward from asterisk,bellow is the example. 1001 called 1002 and 1002 forwarded call to 1003 then callerid in 1003 phone is showing 1002,this is wrong it shound be 1001(he is actual caller). If u do blind transfer instead

Re: [asterisk-users] Push central phone book to phones

2010-12-03 Thread Jonas Kellens
On 12/02/2010 04:31 PM, Ishfaq Malik wrote: On Thu, 2010-12-02 at 16:01 +0100, Jonas Kellens wrote: On 12/02/2010 03:47 PM, Ishfaq Malik wrote: On Thu, 2010-12-02 at 15:19 +0100, Jonas Kellens wrote: Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a

Re: [asterisk-users] + on Caller-ID

2010-12-03 Thread John Novack
And yet SOME providers SEND the 1 Abiding by some standard would be nice! John Novack C F wrote: When sending CLID in the US it should never contain more than 10 digits (don't include the 1). In fact some providers will BLOCK your call if you do. On Thu, Dec 2, 2010 at 2:24 PM, John

Re: [asterisk-users] Quintum AFT800 on Asterisk 1.4.29

2010-12-03 Thread Zoel Hairi - Yahoo
All, This case solved. Thanks . J Regards, Zoel Hairi From: Zoel Hairi - Yahoo [mailto:zoelha...@yahoo.co.id] Sent: Monday, November 22, 2010 12:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Quintum AFT800 on Asterisk 1.4.29 Hi All, Is it

Re: [asterisk-users] Abandon events in cdr

2010-12-03 Thread Rodrigo Lang
Sorry, of course cdr.conf not queues.conf. marcus Am 01.12.2010 19:16 schrieb marcus rothe sync...@googlemail.com: Hi Rodrigo, have you got enabled the appropriate line in queues. Conf? Regards Marcus Thanks very much, I include the line unansweredy=yes in the cdr.conf and solve the

Re: [asterisk-users] Abandon events in cdr

2010-12-03 Thread Steve Howes
On 3 Dec 2010, at 13:47, Rodrigo Lang wrote: unansweredy = yes Remove the extra y. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] DAHDI on VMWARE

2010-12-03 Thread Danny Nicholas
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Thursday, December 02, 2010 7:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI on VMWARE

Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files corrupted

2010-12-03 Thread Mike
Thanks Jonathan, I did that, it worked. I thought it had something to do with 1.6.2 SVN, since I`ve been using Asterisk for 5 years now and the first time it happened was the day I used SVN. Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files corrupted

2010-12-03 Thread Mike
Hi Tilghman, This particular customer was one of my less sophisticated customer, and I know for sure he isn`t using anything else than Voicemailmain. Not even the basic voicemail to email function. But I will keep an eye opened for any future problem. Mike -Original Message- From:

Re: [asterisk-users] DAHDI on VMWARE

2010-12-03 Thread Daniel Tryba
On Thu, Dec 02, 2010 at 02:03:19PM -0600, Danny Nicholas wrote: We are moving our computers from a cluster of physical machines to a VMWARE server with virtual machines. We investigated and are looking to replace our TDM400P/TDM410P with AEX410P cards. Can we run asterisk with

Re: [asterisk-users] DAHDI on VMWARE

2010-12-03 Thread Danny Nicholas
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba Sent: Friday, December 03, 2010 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI on VMWARE On

Re: [asterisk-users] Sharing Fail2ban data

2010-12-03 Thread marek cervenka
I've been doing a little work that I wanted to share. We've had a number of Asterisk systems that have been under heavier than normal attack. We use fail2ban but we either have to let each system be exposed or keep all the data synchronized which is a bit of a chore. I wrote a little

Re: [asterisk-users] + on Caller-ID

2010-12-03 Thread C F
I believe there has been some talk on this list about this in the past. The providers do NOT send 1 on the PSTN only to the customer. Meaning they add it when they THINK its the final destination. You can try this by forwarding your number with that provider to another phone that the provider

Re: [asterisk-users] DAHDI on VMWARE

2010-12-03 Thread Shaun Ruffell
On 12/03/2010 08:46 AM, Danny Nicholas wrote: Actually I think I wasn't clear. Instead of the latest version of DAHDI-linux I think I should have said to the current trunk of dahdi-linux i.e. ]# svn co http://svn.asterisk.org/svn/dahdi/linux/trunk dahdi-linux-trunk ]# cd dahdi-linux-trunk

[asterisk-users] Asterisk 1.6 (Web-meetme)

2010-12-03 Thread Manmohan Singh Jandu
Hi, I am really not sure if this is related to the meetme in asterisk OR this is something to do in web-meetme. I tried to google but didnt get any proper results. I am facing one issue in Web-meetme on the expiry of any conference that we create. I am having Web-meetme 4.0.2 over Asterisk 1.6

Re: [asterisk-users] DAHDI on VMWARE

2010-12-03 Thread Danny Nicholas
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Friday, December 03, 2010 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI on VMWARE On

Re: [asterisk-users] + on Caller-ID

2010-12-03 Thread Sherwood McGowan
On Fri, Dec 3, 2010 at 9:18 AM, C F shma...@gmail.com wrote: I believe there has been some talk on this list about this in the past. The providers do NOT send 1 on the PSTN only to the customer. Meaning they add it when they THINK its the final destination. You can try this by forwarding your

Re: [asterisk-users] Correct operation of timout parameter for dial application

2010-12-03 Thread Bruce McAlister
Hi All, Just another follow-up, does anyone have any thoughts on the query below? Thanks Bruce From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce McAlister Sent: 01 December 2010 18:12 To: Asterisk Users Mailing List -

[asterisk-users] What are the minimal permissions required to read the PeerStatus and Registry events?

2010-12-03 Thread Frank Church
I am logging events from the AMI and the PeerStatus and Registry events show that the privilege for them is System,All. Can a lower set of privileges be used? All looks pretty high to me. /Frank -- _ -- Bandwidth and

Re: [asterisk-users] codec_g729a implicated in file descriptor buildup

2010-12-03 Thread Steve Murphy
On Wed, Dec 1, 2010 at 12:15 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 12/01/2010 01:05 PM, Steve Murphy wrote: Hello, I wonder if anyone else has noticed this. I see a pair of calls to pipe() within the codec_g729a, and suddenly, I have a leaked file descriptor that remains

Re: [asterisk-users] Abandon events in cdr

2010-12-03 Thread Rodrigo Lang
No, i am standing absolutely at the beginning. I think the table structure should be googleable. But i haven 't found an automatism to dump the queuelog flatfile into a database table. Found a perl script but it doesn' t work for me. Am 03.12.2010 19:29 schrieb Rodrigo Lang

Re: [asterisk-users] codec_g729a implicated in file descriptor buildup

2010-12-03 Thread Kevin P. Fleming
On 12/03/2010 01:17 PM, Steve Murphy wrote: On Wed, Dec 1, 2010 at 12:15 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 12/01/2010 01:05 PM, Steve Murphy wrote: Hello, I wonder if anyone else has noticed this. I see a

Re: [asterisk-users] Quintum AFT800 on Asterisk 1.4.29

2010-12-03 Thread alireza sadeh seighalan
hi Zoel would you tell us how do that? good luck seighalani On Fri, Dec 3, 2010 at 4:57 PM, Zoel Hairi - Yahoo zoelha...@yahoo.co.idwrote: All, This case solved. Thanks … J *Regards,* *Zoel Hairi* *From:* Zoel Hairi - Yahoo [mailto:zoelha...@yahoo.co.id] *Sent:* Monday,

Re: [asterisk-users] Asterisk 1.6 (Web-meetme)

2010-12-03 Thread Dan Austin
Manmohan wrote: I am really not sure if this is related to the meetme in asterisk OR this is something to do in web-meetme. I tried to google but didnt get any proper results. I am facing one issue in Web-meetme on the expiry of any conference that we create. I am having Web-meetme 4.0.2

Re: [asterisk-users] What are the minimal permissions required to read the PeerStatus and Registry events?

2010-12-03 Thread Jim Dickenson
A problem that I have always had with AMI events is that they are not controllable at a very fine level. As an example, turning on call class gives WAY more that one might always want. I had posted a patch some time ago that added a new class. The patch was rejected with a comment that some

Re: [asterisk-users] Asterisk 1.6 (Web-meetme)

2010-12-03 Thread Manmohan Singh Jandu
Hi Dan, I am currently on Asterisk 1.6.2.14. Thanks Regards Manmohan Singh On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I am really not sure if this is related to the meetme in asterisk OR this is something to do in web-meetme. I tried to google

[asterisk-users] Issue with MOH - Asterisk 1.4.17

2010-12-03 Thread Napoleón Ernesto López Espinoza
Hi, I'm currently working with Asterisk 1.4.17 under ubuntu server 8.04.2. MOH stopped working suddenly a few days ago with no apparent reason. I already checked the wiki and tried different things. I already verified the following items from the wiki: 1. Make sure your asterisk user has read

Re: [asterisk-users] Asterisk 1.6 (Web-meetme)

2010-12-03 Thread Dan Austin
Manmohan wrote: I am currently on Asterisk 1.6.2.14. Do you have schedule=yes in meetme.conf? I incorrectly remembered/thought that all of the Realtime features were controlled by that option, only a small number, such as end times and CDR logging On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin

Re: [asterisk-users] Issue with MOH - Asterisk 1.4.17

2010-12-03 Thread Doug Lytle
Napoleón Ernesto López Espinoza wrote: We're sorry, your call did not go through. Any clues about this issue? How about some output from your console when it fails? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve

Re: [asterisk-users] Asterisk 1.6 (Web-meetme)

2010-12-03 Thread Manmohan Singh Jandu
Hi Dan, In meetme.conf the schedule=yes was commented, after removing its working fine. But one strange thing had started now. I started getting segmentation fault. following are the errors which i see in it: warning: difference appears to be caused by prelink, adjusting expectations

Re: [asterisk-users] Asterisk 1.6 (Web-meetme)

2010-12-03 Thread Dan Austin
The errors you posted do not point to a the problem. Did you build from source or are you using packages? If from source, grep for useropts in app_meetme.c and The second instance should be: char useropts[OPTIONS_LEN + 1] = ; If the line does not have the = , then the issue is that the bug I

Re: [asterisk-users] Issue with MOH - Asterisk 1.4.17

2010-12-03 Thread Darrick Hartman (lists)
On 12/03/2010 03:30 PM, Doug Lytle wrote: Napoleón Ernesto López Espinoza wrote: We're sorry, your call did not go through. Any clues about this issue? How about some output from your console when it fails? It's would also be advised to use a much more recent version. Asterisk 1.4.17 has

Re: [asterisk-users] alarm POTS lines

2010-12-03 Thread Jeff LaCoursiere
On Thu, 2 Dec 2010, Kevin P. Fleming wrote: On 12/02/2010 10:58 AM, Jeff LaCoursiere wrote: I would love to see a DSP modem that could answer an asterisk channel, send the data stream over TCP to some remote asterisk, which could then relay the stream by making an outbound DSP modem call

[asterisk-users] Polycom Park by EFK

2010-12-03 Thread Andrew Joakimsen
Has anyone gotten one-touch call parking to work on Polycom phones via the Enhanced Feature Keys feature working? I've looked at various examples, they appear correct, but the phones (501, 3.1.x firmware) show the Park button while in a call but this does not actually cause the call to be parked.

Re: [asterisk-users] Zaptel / Asterisk on Solaris

2010-12-03 Thread RR
On Thu, Dec 2, 2010 at 5:05 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote: Assuming Solaris is anything like Linux, the installer will just be a shell script. Open the script in a text editor and search for the text of the error message. It will be wrapped inside an `if` statement, just

Re: [asterisk-users] Zaptel / Asterisk on Solaris

2010-12-03 Thread RR
On Wed, Dec 1, 2010 at 3:58 PM, RR ranjt...@gmail.com wrote: Zaptel package isn't installing though ...crashes midway complaining that: *Operating environment requirement not met. This package requires Solaris 7 or better. checkinstall script suspends* huh? I'm running 5.11, which

Re: [asterisk-users] Polycom Park by EFK

2010-12-03 Thread Ryan Wagoner
On Fri, Dec 3, 2010 at 8:02 PM, Andrew Joakimsen joakim...@gmail.com wrote: Has anyone gotten one-touch call parking to work on Polycom phones via the Enhanced Feature Keys feature working? I've looked at various examples, they appear correct, but the phones (501, 3.1.x firmware) show the Park