Re: [asterisk-users] is roundrobin and rrmemory the same meaning?

2010-01-12 Thread Tony Mountifield
In article a160a7d6100926h6d2e6f88m64175b92cfcc2...@mail.gmail.com, Zhang Shukun bit...@gmail.com wrote: Dear all, I can't understand the diff between roundrobin and rrmemory strategy. Could you explain for me ? and is roundrobin means each available interface ring once or several

Re: [asterisk-users] is roundrobin and rrmemory the same meaning?

2010-01-12 Thread Zhang Shukun
Thank you! it's very helpful. now i have another question: in asterisk, each agent should login first and then can response to the caller. but i don't want to the login action. i need agent shold response directly without login first. how should i do ? can users in sip.conf to be agents? so it

Re: [asterisk-users] is roundrobin and rrmemory the same meaning?

2010-01-12 Thread Leif Neland
Zhang Shukun wrote: Thank you! it's very helpful. now i have another question: in asterisk, each agent should login first and then can response to the caller. but i don't want to the login action. i need agent shold response directly without login first. how should i do ? can users in

Re: [asterisk-users] Using HASH() and REALTIME_HASH()

2010-01-12 Thread Benoit
Le 10/01/2010 07:53, Tilghman Lesher a écrit : On Saturday 09 January 2010 15:22:29 Benoit wrote: I'm playing around with asterisk 1.6.2.0 and the first try was to replace my now non-functionning 'app-realtime' macro which emulated RealTime with REALTIME_HASH() There is very few

[asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Peter Childs
This is currently still at a proof of concept stage. After being mis-sold a Alcatel phone system, that does None of the things we asked for (Ok it takes calls but that's about it) We are looking at alternatives to try and bring some of the features we previously had on our old Analogue STS

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread James Mutuku
http://www.google.co.ke/search?q=asterisk+for+call+centersie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship

[asterisk-users] Virtual ISDN device /dev/XYZ

2010-01-12 Thread Roger Schreiter
Hello, I do remember having read some weeks ago something about a virtual device provided by asterisk, behaving like an ISDN device, i.e. like /dev/isdn0. I know iaxmodem, but iaxmodem imho unfortunately does not transport raw ISDN data (HDLC frames), but only voice. Do I remember right, and

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Peter Childs
2010/1/12 James Mutuku listmut...@gmail.com: http://www.google.co.ke/search?q=asterisk+for+call+centersie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a I can use Google just as well as the next guy, and if you'd bothered to look at the results you could see they were extremely

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Danny Nicholas
Since you are small, trixbox would probably be the ideal flavor of Asterisk for you. It is a downloadable ISO that installs Scientific Linux and Asterisk and sets you up to manage everything with a GUI interface from a browser. Once you outgrow that, you can either expand it, go for Commercial

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Doug Lytle
Peter Childs wrote: I'm not an expert on phones, I'm just an IT guy who thinks he might have a solution to a problem, that is not really his problem but is Then you'll need to be prepared to do a LOT of reading. You'll want to start off on: http://voip-info.org Then there is the

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Jeff LaCoursiere
On Tue, 12 Jan 2010, Danny Nicholas wrote: Since you are small, trixbox would probably be the ideal flavor of Asterisk for you. It is a downloadable ISO that installs Scientific Linux and Asterisk and sets you up to manage everything with a GUI interface from a browser. Once you outgrow

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Danny Nicholas
I actually meant switchvox (just to make the content of my comment be kosher), but in general, the OP should probably go with a canned solution unless he wishes to get his hands dirty. -- Danny Nicholas -- -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Doug Lytle
Jeff LaCoursiere wrote: I agree that FreePBX would be the ideal flavor for him, but I am a recent convert to Elastix. Much tighter GUI, more included stuff (like And, I'd be in the camp that would advocate getting your hands dirty and learn to program without the GUI. You'll learn a

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Richard Kenner
And, I'd be in the camp that would advocate getting your hands dirty and learn to program without the GUI. You'll learn a lot and then if you'd want to move to a GUI and something breaks, you'll have some idea on what and how to fix it. Knowing now what I do, I find a GUI to

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Robert Lister
Do you have any idea of numbers of users, and number/type of external lines as this can be quite important when deciding what type of asterisk setup and hardware to go with. (For example, if your lines are already presented over ISDN PRI or BRI, or if they are provided over IP, by an IP telephony

Re: [asterisk-users] Hardware issue, was Using HASH() and REALTIME_HASH()

2010-01-12 Thread Tilghman Lesher
On Tuesday 12 January 2010 04:44:36 Benoit wrote: I just experienced another problem however i have two rnis cards, one b410p and one te220, while the later works prefectly i can't really make the first one work, using DAHDI or mISDN under asterisk 1.6. If you're having trouble with any

Re: [asterisk-users] Sipgate DTMF not detected

2010-01-12 Thread Kristian Kielhofner
On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes steve-li...@geekinter.net wrote: Codec? I've had 2833 do funny things with anything other than a/ulaw (might just be me though..) S -- Codecs other than G711u/a don't support inband DTMF. Seeing as INFO is rarely used that pretty much leaves

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Lenz Emilitri
Yes it is - we have thousands of happy clients worldwide. :) My suggestion is to go for somebody who has relevant experience and is going to do the install for you. Unless your CC is very small, you don't want to be looking up the manuals when you went live and start having quality issues If

Re: [asterisk-users] Realtime LDAP Queues crashes

2010-01-12 Thread Jorge Salamero Sanz
On Friday 08 January 2010 01:38:42 Gavin Henry wrote: What are the LDAP searches like? after updating and applying this patch: http://issues.asterisk.org/view.php?id=13573 doesn't crash and the queries i get are ok: conn=0 op=67 SRCH base=dc=nodomain scope=2 deref=0

Re: [asterisk-users] is roundrobin and rrmemory the same meaning?

2010-01-12 Thread Lenz Emilitri
You can list phones directly as static members of the queue. this is generally sub.optimal because if. e.g. an agent of yours is home sick, her phone will be ringing and you'll be wasting caller time. Also by tracking logins and logoffs you can measure agent productivity, and this is pretty useful

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Jeff LaCoursiere
On Tue, 12 Jan 2010, Richard Kenner wrote: And, I'd be in the camp that would advocate getting your hands dirty and learn to program without the GUI. You'll learn a lot and then if you'd want to move to a GUI and something breaks, you'll have some idea on what and how to fix it. Knowing

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Richard Kenner
Your comments both come from having taken a short look at FreePBX and dismissed it without investigating how powerful it can be. Yes, but the discussion is about COMPLEXITY, not power! Sure, there are hooks where you can do anything you want, but if you were to set up identical configurations

[asterisk-users] Send 503 or 603 error after answer()

2010-01-12 Thread jonas kellens
Hello list. Is it possible in the Asterisk dialplan to send a 503 Service Unavailable of 603 Decline after having answered the call with Answer() in the dialplan ?? Suppose that I first want to check the call in a MySQL-database, while I put some MoH, and then let the call go through or send

Re: [asterisk-users] Sipgate DTMF not detected

2010-01-12 Thread listu...@spamomania.co.uk
On Tue, 2010-01-12 at 10:37 -0500, Kristian Kielhofner wrote: On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes steve-li...@geekinter.net wrote: Codec? I've had 2833 do funny things with anything other than a/ulaw (might just be me though..) S -- Codecs other than G711u/a don't

[asterisk-users] Question about SIP registration

2010-01-12 Thread Aggio Alberto
Hi guys, I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC, with eth0 set to address 192.168.1.1 (NATted over public network, with address 89.X.Y.Z) and eth1 set to address 1.1.1.1. In [sip.conf] I set general option bindaddr=0.0.0.0; IP address to

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Jeff LaCoursiere
On Tue, 12 Jan 2010, Richard Kenner wrote: Your comments both come from having taken a short look at FreePBX and dismissed it without investigating how powerful it can be. Yes, but the discussion is about COMPLEXITY, not power! I thought the discussion was about how an IT guy with no

[asterisk-users] Inserting a wait in a sip dial

2010-01-12 Thread evert
Hi All, After searching and didnt found it, im just sending my situation here, maybe someone knows where i should look. Im using Asterisk 1.6.1.10 Internally the user with a sip phone dials a number for instance 0623456789 It goes fine to the specific dial rule: which is: exten =

[asterisk-users] SIP Security

2010-01-12 Thread Juan C. Villa
Hey guys, I've been running asterisk on my server for some time now (currently running Asterisk 1.6.2.0). I am having security issues with my SIP accounts. Unauthorized people have been able to access the server (bots) and they have been able to make calls (in today's case to Cuba). Here's a

Re: [asterisk-users] Question about SIP registration

2010-01-12 Thread Warren Selby
Instead of host=dynamic, use host=1.1.1.1, or host=1.1.1.0/255.255.255.0. Thanks, --Warren Selby On Jan 12, 2010, at 11:16 AM, Aggio Alberto alberto.ag...@loquendo.com wrote: Hi guys, I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC, with eth0 set to

[asterisk-users] Minimal Asterisk Web Interface?

2010-01-12 Thread Tim Nelson
I'm looking for a web GUI to Asterisk that I can run on some small embedded hardware. I've used FreePBX in the past but the overhead is not to my liking and it is entirely too complicated. I do not wish to change my entire OS just for the GUI either (aka AstLinux). Is there anything out there?

[asterisk-users] VMs IMAP Storage

2010-01-12 Thread --[ UxBoD ]--
Hi, is it possible to store a VM in multiple mailboxes ? if not; would it be right to file a RFE so that you could specify on imapuser something like: imapuser=us...@domain.comus...@domain.com like you can with SIP, sounds etc ? Would make it very nice indeed for shared mailboxes. Thoughts ?

Re: [asterisk-users] Question about SIP registration

2010-01-12 Thread Robert Lister
On Tue, 2010-01-12 at 18:16 +0100, Aggio Alberto wrote: Then I have configured an account as following: [999] type=friend username=999 You don't appear to have a secret= line in there with a password option... or did you snip it? Can someone explain me this kind of behaviour? Is it

Re: [asterisk-users] SIP Security

2010-01-12 Thread --[ UxBoD ]--
- Juan C. Villa juan...@villafam.com wrote: Hey guys, I've been running asterisk on my server for some time now (currently running Asterisk 1.6.2.0). I am having security issues with my SIP accounts. Unauthorized people have been able to access the server (bots) and they have been

Re: [asterisk-users] Send 503 or 603 error after answer()

2010-01-12 Thread Kevin P. Fleming
jonas kellens wrote: Is it possible in the Asterisk dialplan to send a 503 Service Unavailable of 603 Decline after having answered the call with Answer() in the dialplan ?? No. Answer generates (for a SIP channel) a '200 OK', which is a final response. You cannot send any further final

Re: [asterisk-users] Minimal Asterisk Web Interface?

2010-01-12 Thread Michael Iedema
Hi Tim, On Tue, Jan 12, 2010 at 6:56 PM, Tim Nelson tnel...@fudnet.net wrote: I'm looking for a web GUI to Asterisk that I can run on some small embedded hardware. I've used FreePBX in the past but the overhead is not to my liking and it is entirely too complicated. I do not wish to change

Re: [asterisk-users] SIP Security

2010-01-12 Thread Martin
Lets just say that you turned off the security ... [general] context=default ; Default context for incoming calls so everyone that can connect to your IP port 5060 UDP can access default context... why would you allow this context to place outgoing calls then ? secret=blah also

Re: [asterisk-users] SIP Security

2010-01-12 Thread Juan C. Villa
Martin, I changed all the passwords to blah so I would not reveal them on this email. The password if much more complex than that. It appears that my problem was that I was allowing guest calls. I have beefed up the security, activated fail2ban, along with other things. But thanks anyways!

Re: [asterisk-users] Minimal Asterisk Web Interface?

2010-01-12 Thread Tzafrir Cohen
On Tue, Jan 12, 2010 at 08:56:05PM +0300, Tim Nelson wrote: I'm looking for a web GUI to Asterisk that I can run on some small embedded hardware. I've used FreePBX in the past but the overhead is not to my liking and it is entirely too complicated. I do not wish to change my entire OS just

Re: [asterisk-users] is roundrobin and rrmemory the same meaning?

2010-01-12 Thread Robert Lister
On Tue, 2010-01-12 at 11:26 +0800, Zhang Shukun wrote: Dear all, I can't understand the diff between roundrobin and rrmemory strategy. Could you explain for me ? and is roundrobin means each available interface ring once or several times and ring another? roundrobin is deprecated in 1.4

Re: [asterisk-users] Send 503 or 603 error after answer()

2010-01-12 Thread jonas kellens
Thank you for your answer. So if I use early media (not putting answer() at the beginning of my dialplan), how can I send a 503 or 603 from the dialplan ?? Kind regards, Jonas. On Tue, 2010-01-12 at 12:05 -0600, Kevin P. Fleming wrote: jonas kellens wrote: Is it possible in the Asterisk

Re: [asterisk-users] Inserting a wait in a sip dial

2010-01-12 Thread David Gibbons
snip But then the other peer says: -- Called *31#w06123456...@xs4all-out -- SIP/xs4all-out-0234 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/evert-0233' status is 'CONGESTION' Anyone an idea where i should look, or

Re: [asterisk-users] Inserting a wait in a sip dial

2010-01-12 Thread Danny Nicholas
Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a 1/2 second delay before dialing, ww1234 a 1 second delay, etc. Try it with 2 or 3 w's instead of 1... -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Inserting a wait in a sip dial

2010-01-12 Thread evert
The problem is only that, it first needs to dial *31#, then wait 1 sec or so, and then dial the number. So it would be needed that its Dial(SIP/*31#w061234123412) But this doesnt seem to work. Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a 1/2 second delay before

[asterisk-users] Problem logs queue_log-mysql

2010-01-12 Thread Dpto. de Sistemas
] queue_log = yes queue_log_name = queue_log Thanks, Best regards!! Cristian Arguello. __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4765 (20100112) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com

Re: [asterisk-users] Inserting a wait in a sip dial

2010-01-12 Thread Danny Nicholas
This doesn't work? Dial(SIP/*31#ww061234123412) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ev...@disruptor.nl Sent: Tuesday, January 12, 2010 12:59 PM To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Inserting a wait in a sip dial

2010-01-12 Thread evert
Ok my problem is solved now, it was easyer fixed by adding: Set(CALLERPRES()=unavailable) That did exactly the same as the *31# would have done. So for me the problem is solved. The problem is only that, it first needs to dial *31#, then wait 1 sec or so, and then dial the number. So it

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Doug Lytle
Jeff LaCoursiere wrote: I thought the discussion was about how an IT guy with no previous asterisk experience could get up and running the fastest. By FAR that answer is to No, actually he said, This is currently still at a proof of concept stage. By whose estimation? To even get

Re: [asterisk-users] Minimal Asterisk Web Interface?

2010-01-12 Thread Tim Nelson
- Michael Iedema mich...@askozia.com wrote: Hi Tim, On Tue, Jan 12, 2010 at 6:56 PM, Tim Nelson tnel...@fudnet.net wrote: I'm looking for a web GUI to Asterisk that I can run on some small embedded hardware. I've used FreePBX in the past but the overhead is not to my liking and it

Re: [asterisk-users] Minimal Asterisk Web Interface?

2010-01-12 Thread Tim Nelson
- Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Tue, Jan 12, 2010 at 08:56:05PM +0300, Tim Nelson wrote: I'm looking for a web GUI to Asterisk that I can run on some small embedded hardware. I've used FreePBX in the past but the overhead is not to my liking and it is entirely too

Re: [asterisk-users] Inserting a wait in a sip dial

2010-01-12 Thread David Gibbons
snip This doesn't work? Dial(SIP/*31#ww061234123412) /snip When I was browsing the sip debugs, it seemed that the 'w' was not being honored for one reason or another. My thought at the time was maybe it didn't work at all over SIP. Does the w *just* work with dahdi or does it work over sip as

Re: [asterisk-users] Inserting a wait in a sip dial

2010-01-12 Thread Kevin P. Fleming
David Gibbons wrote: snip This doesn't work? Dial(SIP/*31#ww061234123412) /snip When I was browsing the sip debugs, it seemed that the 'w' was not being honored for one reason or another. My thought at the time was maybe it didn't work at all over SIP. Does the w *just* work with

Re: [asterisk-users] Inserting a wait in a sip dial

2010-01-12 Thread David Gibbons
snip 'w' is really only supported on channels where digit-by-digit dialing is the norm, which generally means analog trunks (or digital trunks using CAS signaling). /snip Thanks Kevin, that's what I figured (though not quite so concisely)... Going foward, is there any way to programmatically

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread Peter Childs
2010/1/12 Robert Lister r...@lentil.org: Do you have any idea of numbers of users, and number/type of external lines as this can be quite important when deciding what type of asterisk setup and hardware to go with. (For example, if your lines are already presented over ISDN PRI or BRI, or if

Re: [asterisk-users] Multi-Tenant Parking

2010-01-12 Thread Michael Wyres
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UxBoD Sent: Tuesday, 12 January 2010 17:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multi-Tenant Parking Should that not say parkinglot

Re: [asterisk-users] Multi-Tenant Parking

2010-01-12 Thread Michael Wyres
Have you looked at this? http://www.google.com/#q=app_valetparking I have - but would rather use the inbuilt functionality if possible before resorting to third-party code... IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any

Re: [asterisk-users] Inserting a wait in a sip dial SOLVED (kind of)

2010-01-12 Thread David Gibbons
snip Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? /snip Well, due to the lack of responses, either I missed something obvious or nobody cares. I'm really hoping I didn't miss something obvious... :). In any event, I got curious of my own

Re: [asterisk-users] Sipgate DTMF not detected

2010-01-12 Thread Kristian Kielhofner
On Tue, Jan 12, 2010 at 12:09 PM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: Assuming that I enable debugging using: asterisk -rvv CLI sip set debug on Then with this: dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw I see nothing nothing showing keypresses

[asterisk-users] AudioCodes MP-114 - SAS (Stand Alone Survivability) configuration

2010-01-12 Thread Joseph
I have AudioCodes MP-114 and I'm trying to configure SAS (Stand Alone Survivability); when Asterisk is down the MediaPack gateway should forward the call IN/OUT through the gateway (without asterisk in the middle), but it is not working. I'm working with tech. support from the source I

[asterisk-users] Xorcom 32 channel FXS gateway

2010-01-12 Thread C F
Anyone on the list ever used it? I'm trying to quote a system with 192 analog ports, one of the options are the Xorcom 32 channel FXS USB Channel Banks. Any input would be appreciated. TIA -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Send 503 or 603 error after answer()

2010-01-12 Thread Kevin P. Fleming
jonas kellens wrote: So if I use early media (not putting answer() at the beginning of my dialplan), how can I send a 503 or 603 from the dialplan ?? By using the proper method of canceling the call... Busy, Congestion, or an explicit cause code passed to Hangup. -- Kevin P. Fleming Digium,

Re: [asterisk-users] Xorcom 32 channel FXS gateway

2010-01-12 Thread Carlos Chavez
On Tue, 2010-01-12 at 18:05 -0500, C F wrote: Anyone on the list ever used it? I'm trying to quote a system with 192 analog ports, one of the options are the Xorcom 32 channel FXS USB Channel Banks. Any input would be appreciated. I have used Astribanks for a while now and they are

Re: [asterisk-users] Xorcom 32 channel FXS gateway

2010-01-12 Thread Michelle Dupuis
You can address the order of detection problem using udev rules... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Tuesday, January 12, 2010 6:53 PM To: Asterisk Users List Subject: Re:

Re: [asterisk-users] Inserting a wait in a sip dial SOLVED (kind of)

2010-01-12 Thread Steve Murphy
Dave-- I remember adding a feature a long time ago for snoms, to the source code, to send dtmf out for some button press on a snom phone, in the 'outward' direction, I think to activate a feature or somesuch. (Boy, is my memory hazy!) At any rate, I was able to inject dtmf, but I had to do it in

Re: [asterisk-users] Inserting a wait in a sip dial SOLVED (kind of)

2010-01-12 Thread Jim Dickenson
If you need to inject dtmf tones or sound into an existing channel you can use chanspy with option w. I play sound files using the AMI to originate a call to an extension that does chanspy on one leg and a playback on the other. I use channel variables to say which channel to play to and which

Re: [asterisk-users] Xorcom 32 channel FXS gateway

2010-01-12 Thread C F
Any echo issues using FXS ports? On Tue, Jan 12, 2010 at 6:53 PM, Carlos Chavez cur...@telecomabmex.com wrote: On Tue, 2010-01-12 at 18:05 -0500, C F wrote: Anyone on the list ever used it? I'm trying to quote a system with 192 analog ports, one of the options are the Xorcom 32 channel FXS

Re: [asterisk-users] Multi-Tenant Parking (HALF SOLVED)

2010-01-12 Thread Michael Wyres
I have found that this seems to be a functional difference between the Park() and the ParkAndAnnounce() functions. Park() respects the parking lot specification, yet ParkAndAnnounce() does not respect the fact that you’ve tried to arbitrarily set the parking lot. The code below “works” as

Re: [asterisk-users] is roundrobin and rrmemory the same meaning?

2010-01-12 Thread Zhang Shukun
2010/1/12 Lenz Emilitri lenz.lo...@gmail.com: You can list phones directly as static members of the queue. i know i can configure the queue.conf and agents.conf to add queue name and queue members by hand. Could i use functions to create queue name and add queue members dynamiclly. because i

Re: [asterisk-users] is roundrobin and rrmemory the same meaning?

2010-01-12 Thread Zhang Shukun
2010/1/13 Robert Lister r...@lentil.org: On Tue, 2010-01-12 at 11:26 +0800, Zhang Shukun wrote: Dear all, I can't understand the diff between roundrobin and rrmemory strategy. Could you explain for me ? and is roundrobin means each available interface ring once or several times and ring

[asterisk-users] Odd Voicemail Issue

2010-01-12 Thread William Stillwell ( Lists )
I have several extensions in the Central Timezone, the Server is in the Eastern Timezone. all the voicemail files have a datetimestamp of EST not of the tz= option under the usermail ... voicemail.conf under [general] tz=EST under [default] mailbox_a,password,,,tz=CST6CDT

Re: [asterisk-users] Odd Voicemail Issue

2010-01-12 Thread William Stillwell ( Lists )
ok, I figured it out.. tz=zonename from zonemessages all fixed. - Original Message - From: William Stillwell ( Lists ) To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, January 12, 2010 9:59 PM Subject: [asterisk-users] Odd Voicemail Issue I

[asterisk-users] Polycom Mute Problem

2010-01-12 Thread Michael
Hey Yall I have an interesting situation. When a person is on the phone (Polycom 501) and another call hits the phone the phone mutes on the users side not the person outside the asterisk system. It stays mute until the call goes to voicemail. Its like the beep we should get from the call waiting

Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-12 Thread Arun Sasidhar
Hi, I got a solution for this problem from Freepbx forumhttp://www.freepbx.org/forum/freepbx/users/caller-id-not-working#comment-23520. Is anybody know about this DTMF to FSK converter? Is this solution solve my problem? Any way I will try it and get back. -- Thanks, Arun S System

[asterisk-users] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?

2010-01-12 Thread hadi motamedi
Dear All I have Asterisk 1.4 installed on my Debian server . I am considering upgrading my Asterisk to the latest version (1.6) . Can you please let me know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk 1.6 ? Thank you --

Re: [asterisk-users] Faxing: Anyone have a compiled executable?

2010-01-12 Thread Doug
At 23:52 1/10/2010, Doug wrote: At 15:33 1/7/2010, Tzafrir Cohen wrote: On Thu, Jan 07, 2010 at 12:50:03AM -0600, Doug wrote: At 00:22 1/7/2010, Tzafrir Cohen wrote: On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote: At 16:49 1/5/2010, Tzafrir Cohen wrote: On Tue, Jan 05,

Re: [asterisk-users] Inserting a wait in a sip dial

2010-01-12 Thread Johann Steinwendtner
Kevin P. Fleming wrote: David Gibbons wrote: snip This doesn't work? Dial(SIP/*31#ww061234123412) /snip When I was browsing the sip debugs, it seemed that the 'w' was not being honored for one reason or another. My thought at the time was maybe it didn't work at all over SIP. Does the

Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]

2010-01-12 Thread listu...@spamomania.co.uk
On Tue, 2010-01-12 at 16:52 -0500, Kristian Kielhofner wrote: On Tue, Jan 12, 2010 at 12:09 PM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: Assuming that I enable debugging using: asterisk -rvv CLI sip set debug on Then with this: dtmfmode=rfc2833