Re: [asterisk-users] Looking for good IAX ATA
On Fri, Apr 10, 2009 at 09:50:34PM -0400, John Rogers wrote: I *was* using the X100P FXS ATA but they discontinued it last year in late October. Several inquiries into when they will re-release/replace it with another IAX ATA have gone unanswered. Atcom.cn is a MFG in china with no USA point of presence - I looked into them, but they don't have any resellers here in the US. I'd like to try to source something in the US if I can, that way my chances of support may be better should something go wrong. Also it doesn't appear that there is an option to buy on the atcom.cn site. It seems there are very few (if any) options for IAX ATAs and that's a sad thing, since IAX is far more scalable then SIP and much easier to get through firewalls (just forward port 4569) than SIP is. I've been very frustrated trying to make SIP work through our Monowall firewall without any success and with MUCH frustration and aggrevation! IAX is so much nicer - wish OEMs would start offering this. What would it take to make one? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there documentation explaining res_config_curl?
Eric Chamberlain wrote: Is there any documentation that explains res_config_curl? We use the 1.4 backported version - it works so well I just can't sing it's praises enough. We use it for realtime voicemail and realtime queues / queue members. Have a look at bug #11747 for some documentation. Julian Specifically, the format of realtime calls made to the web server and what the return string for each call should look like? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT XEN asterisk and a digium board
On Fri, Apr 10, 2009 at 11:55:37PM -0300, David fire wrote: hi how i can give the control of a digium card to the virtual machine? i am using XEN do you recomendo other virtual machine? VMWare openVZ etc...? Thanks David I figure OpenVZ should work well, as it is not a VM. However you will need some cooperation from the host to generate the device files. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [astersik-users] ss7 consultancy $1000 USD
I am looking for someone who will implement an ss7 to sip media server. email me personally for more details. I expect a professional attitude and preferably someone I can communicate in English. There will be phone conversations and IM communications. If you do not have experience implementing this, please do not reply. Bounty is $1000 USD, will be paid cash with signed contract. Thanks. Apu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Greetings Will Not Save
Hi All, -My asterisk will not save voicemail greetings when you call in and record them. -It also will not save voicemail messages after emailing them,even though delete=no. -Folder permissions are fine, no errors in asterisk cli. -If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, and then call in and record new unavail message, unavail.wav disappears? -Asterisk and FreePBX source installs on CentOS 5.4 Can anyone help point me towards any possible info to fix this, i'm stumped and losing hair! Respectfully, Dr. Kenneth Noisewater, Phd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Greetings Will Not Save
Dr. Kenneth Noisewater wrote: -Asterisk and FreePBX source installs on CentOS 5.4 Without version numbers and console output and samples of your dialplan, it'g going to make it very difficult to help. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] Grandstream blind transfer issue
Hi All,
Thanks for your suggestions.
I am using DeadAgi application for origination of calls, i have set same
context to Transfer context.
I have also added Tt options in dial options.
When I am receiving calls to grandstream phone,
I am using transfer button to transfer the call, but it is not transfering
with AGI application,
Can anyone provides me suggestions for blind transfer with AGI application.
My Dialplan is given Below. I have used PHPAGI for the origination of calls.
[bt200]
exten = _X.,1,Set(__TRANSFER_CONTEXT=bt200)
exten = _X.,n,DeadAGI(testing_agi/testing.php)
exten= h,1,NoOp(${DIALSTATUS})
Thanks,
Max Alex
Voip Developer
On Wed, Apr 8, 2009 at 9:47 PM, Klaus Darilion klaus.mailingli...@pernau.at
wrote:
Haven't you read my email?
1. Wrong list
2. Missing log entries (set debug 4, set verbose 4)
klaus
Max Alex schrieb:
Hi All,
Thanks for your reply.
I got this refer message in asterisk.
but there is not any active channel of blind transfer.
--
REFER sip:1...@192.168.1.25 sip%3a1...@192.168.1.25 mailto:
sip%3a1...@192.168.1.25 sip%253a1...@192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0
From: sip:7...@192.168.1.30:5060;transport=udp;tag=3699e1bcbed17687
To: 1101 sip:1...@192.168.1.25 sip%3a1...@192.168.1.25
mailto:sip%3a1...@192.168.1.25 sip%253a1...@192.168.1.25
;tag=as32ed6c48
Contact: sip:7...@192.168.1.30:5060;transport=udp
Supported: replaces, path
Refer-To: sip:1631...@192.168.1.25 sip%3a1631...@192.168.1.25
mailto:sip%3a1631...@192.168.1.25sip%253a1631...@192.168.1.25
Referred-By: sip:7...@192.168.1.25 sip%3a7...@192.168.1.25 mailto:
sip%3a7...@192.168.1.25 sip%253a7...@192.168.1.25
Call-ID: 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
mailto:4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
CSeq: 34526 REFER
User-Agent: Grandstream BT200 1.1.6.46
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
-
--- (14 headers 0 lines) ---
Call 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
mailto:4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25 got a SIP call
transfer from caller: (REFER)!
SIP transfer to extension 1631...@outgoing by 7...@192.168.1.25
mailto:7...@192.168.1.25
localhost*CLI
--- Transmitting (NAT) to 192.168.1.30:5060 http://192.168.1.30:5060
---
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP
192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0;received=192.168.1.30
From: sip:7...@192.168.1.30:5060;transport=udp;tag=3699e1bcbed17687
To: 1101 sip:1...@192.168.1.25 sip%3a1...@192.168.1.25
mailto:sip%3a1...@192.168.1.25 sip%253a1...@192.168.1.25
;tag=as32ed6c48
Call-ID: 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
mailto:4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
CSeq: 34526 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:1...@192.168.1.25 sip%3a1...@192.168.1.25 mailto:
sip%3a1...@192.168.1.25 sip%253a1...@192.168.1.25
Content-Length: 0
Is there any options we need to enable in asterisk or grandstream phone?
I have already user transfer option 'Tt' in dialplan of this.
Please provide me some help.
Thanks in advance!!
Thanks,
Max Alex
Voip Developer
On Wed, Apr 8, 2009 at 2:04 AM, Klaus Darilion
klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at
wrote:
Max Alex wrote:
Hi All,
I have working asterisk version 1.4.24.
I have a blind transfer issue with grandstream bt200.
Does it work with other phones? That means is it a Grandstream isue
or a
general issue?
I have updated the latest firmware to the phone.
The phone is sending the *refer* to asterisk but asterisk is not
able to
connect with the another call
Why? some log messages would help us helping you.
that i have checked in sip debug.
I am using transfer button of the grandstream phone.
Can anybody provide help for this issue?
Please ask again on the user mailing lists and provide some log
messages
Thanks in advance!!
Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Looking for good IAX ATA
Most ATAs I've seen are primarily SOC (System On Chip) implementations. I've never really taken one apart, but perhaps now is a good time. I recently also purchased a QuickPhones QA-342 wifi rechargeable handset: http://www.voipsupply.com/quickphones-qa-342-wifi-sip-phone?utm_source=quickphones-wifi-catutm_medium=banner I was curious to try this out, even though I knew it was SIP only. In the office, it works GREAT - LONG battery life, good reception, but no IAX support. A good IAX ATA and IAX protocol stack in a phone like the QA-342 would be a hands down winner all around. Immagine, being able to roam anywhere with a device like this or ATA and not having to fuss with SIP/NAT. I wish I had knowledge on building embedded devices, else I'd build my own by now... On Sat, Apr 11, 2009 at 3:51 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Fri, Apr 10, 2009 at 09:50:34PM -0400, John Rogers wrote: I *was* using the X100P FXS ATA but they discontinued it last year in late October. Several inquiries into when they will re-release/replace it with another IAX ATA have gone unanswered. Atcom.cn is a MFG in china with no USA point of presence - I looked into them, but they don't have any resellers here in the US. I'd like to try to source something in the US if I can, that way my chances of support may be better should something go wrong. Also it doesn't appear that there is an option to buy on the atcom.cn site. It seems there are very few (if any) options for IAX ATAs and that's a sad thing, since IAX is far more scalable then SIP and much easier to get through firewalls (just forward port 4569) than SIP is. I've been very frustrated trying to make SIP work through our Monowall firewall without any success and with MUCH frustration and aggrevation! IAX is so much nicer - wish OEMs would start offering this. What would it take to make one? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - John Website: http://wizworks.net Blog: http://news.wizworks.net for all the latest! Pictures: http://www.flickr.com/photos/wizworks Sign the petition to Apple to enable iPod support on Linux!: http://www.petitiononline.com/eb221998/petition.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OFF TOPIC] wich virtualization solution to use?
hi there are a lot of virtualization solution out there and every one is the best and has some pro and some cons... wich one do you recomend? the idea to isolate diferents servers asterisk apache ... it is a good idea? sorry for the off topic but here is a place where are a lot of linux gurus Thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OFF TOPIC] wich virtualization solution to use?
It's a good idea if it makes sense to you organisationally. There is no definitive answer on that; it is a methodological question. David fire wrote: hi there are a lot of virtualization solution out there and every one is the best and has some pro and some cons... wich one do you recomend? the idea to isolate diferents servers asterisk apache ... it is a good idea? sorry for the off topic but here is a place where are a lot of linux gurus Thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [astersik-users] ss7 consultancy $1000 USD
Apu Islam wrote: I am looking for someone who will implement an ss7 to sip media server. email me personally for more details. I expect a professional attitude and preferably someone I can communicate in English. There will be phone conversations and IM communications. If you do not have experience implementing this, please do not reply. Bounty is $1000 USD, will be paid cash with signed contract. Thanks. This is a pretty simple thing to do using Asterisk with its native SS7 stack (libss7). You might even be able to do it yourself. In any case, a lot of the people that have experience doing things like this are actually on the Asterisk-SS7 mailing list (you can subscribe to it at lists.digium.com). Please let me know if you have any issues with using and/or configuring libss7, since I very much want it to be easy for people to use and configure. (I wrote it) :-) -- Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OFF TOPIC] wich virtualization solution to use?
On Sat, Apr 11, 2009 at 12:04 PM, David fire ddf...@gmail.com wrote: hi there are a lot of virtualization solution out there and every one is the best and has some pro and some cons... wich one do you recomend? the idea to isolate diferents servers asterisk apache ... it is a good idea? sorry for the off topic but here is a place where are a lot of linux gurus Thanks If all your virtual machines are linux, openvz is probably the easiest and provides the best performance. But all it does is linux. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisknow 1.5 with X100P and TDM400P
Hi All, Sorry if this has been around a millions times.. I have been off this list for a few months now.. I have installed the latest asterisknow (upgraded asterisk to 1.6 as well) and I am having a hard time getting my X100P and TDM400P working.. Its all new to me with dahdi because my old server is still running Asterisk 1.0.2 so there have been lost of changes.. Can someone point me in the right direction for setting up dahdi on asterisk now? I cant seem to get anything to even show when running dahdi show channels from the CLI.. Obviously I need to make it work in a way that will be compatible with asterisknow and the web gui.. Here are the details of my cards.. [r...@pbx asterisk]# dahdi_cfg -vv DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 02) 2 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 [r...@pbx asterisk]# dahdi_scan [1] active=yes alarms=OK description=Wildcard X101P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X101P location=PCI Bus 00 Slot 15 basechan=1 totchans=1 irq=11 type=analog port=1,FXO [2] active=yes alarms=OK description=Wildcard TDM400P REV E/F Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV E/F location=PCI Bus 00 Slot 14 basechan=2 totchans=4 irq=11 type=analog port=2,FXS port=3,none port=4,none port=5,none Thanks in advance.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Frequency Cycle Timeout - E1-R2 METROTEL COLOMBIA
Glad to hear it worked for you. I'd certainly like to take a look this Monday and see why openr2 did not work for you. Moy On Fri, Apr 10, 2009 at 10:42 PM, Giovanny Magallanes gmagalla...@gmail.com wrote: Hi Moises and Steve, I tried with all protocol variants for Openr2 (AR, BR, CN, CZ, CO, EC, ITU, MX, PH, VE) and setting mfcr2_skip_category=yes, but the problem persists. I tried with Unicall and, in this way, I could make and receive calls without problems, using protocol variant BR or CO (I did not try with another variants). Moises, if you wish the next Monday (4/13) we can chat (off-list) for this issue. Thanks for your attention. Giovanny Magallanes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there documentation explaining res_config_curl?
On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote: Eric Chamberlain wrote: Is there any documentation that explains res_config_curl? We use the 1.4 backported version - it works so well I just can't sing it's praises enough. We use it for realtime voicemail and realtime queues / queue members. Have a look at bug #11747 for some documentation. Thank you, that bug does have useful information. We are working on moving from res_config_odbc to res_config_curl, so all asterisk requests go through our django backend, rather than django and asterisk sharing database tables. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT XEN asterisk and a digium board
On Fri, 2009-04-10 at 23:55 -0300, David fire wrote: hi how i can give the control of a digium card to the virtual machine? i am using XEN do you recomendo other virtual machine? VMWare openVZ etc...? with lspci you can determine the hardware adres of your boards, and pass it through from your DOM-o to your DOM-u. I'v several DOM-u's doing asterisk, but none of them use voip-hardware.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisknow 1.5 with X100P and TDM400P
WipeOut wrote: Hi All, Sorry if this has been around a millions times.. I have been off this list for a few months now.. I have installed the latest asterisknow (upgraded asterisk to 1.6 as well) and I am having a hard time getting my X100P and TDM400P working.. Its all new to me with dahdi because my old server is still running Asterisk 1.0.2 so there have been lost of changes.. Can someone point me in the right direction for setting up dahdi on asterisk now? I cant seem to get anything to even show when running dahdi show channels from the CLI.. Obviously I need to make it work in a way that will be compatible with asterisknow and the web gui.. Here are the details of my cards.. [r...@pbx asterisk]# dahdi_cfg -vv DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 02) 2 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 [r...@pbx asterisk]# dahdi_scan [1] active=yes alarms=OK description=Wildcard X101P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X101P location=PCI Bus 00 Slot 15 basechan=1 totchans=1 irq=11 type=analog port=1,FXO [2] active=yes alarms=OK description=Wildcard TDM400P REV E/F Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV E/F location=PCI Bus 00 Slot 14 basechan=2 totchans=4 irq=11 type=analog port=2,FXS port=3,none port=4,none port=5,none Thanks in advance.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I managed to get it working.. Not sure what I did exactly but its working.. :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with asterisk 1.4.24.1
when I make a call to the pstn it shows me this error: aximum retries exceeded on transmission 9d4a24f8-b6737...@192.168.10.19 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Apr 11 20:35:34] WARNING[3169]: chan_sip.c:1998 retrans_pkt: Hanging up call 9d4a24f8-b6737...@192.168.10.19 - no reply to our critical packet (see doc/sip-retransmit.txt). bug? voicemail same error: Executing [*...@netsoluciones:1] Answer(SIP/192.168.10.3-09443498, ) in new stack -- Executing [*...@netsoluciones:2] VoiceMailMain(SIP/192.168.10.3-09443498, ) in new stack -- SIP/192.168.10.3-09443498 Playing 'vm-login' (language 'es') -- SIP/192.168.10.3-09443498 Playing 'vm-password' (language 'es') -- SIP/192.168.10.3-09443498 Playing 'vm-youhave' (language 'es') [Apr 11 20:38:11] WARNING[3169]: chan_sip.c:1976 retrans_pkt: Maximum retries exceeded on transmission eb6f84bfe5bae...@192.168.10.27 for seqno 46643 (Critical Response) -- See doc/sip-retransmit.txt. [Apr 11 20:38:11] WARNING[3169]: chan_sip.c:1998 retrans_pkt: Hanging up call eb6f84bfe5bae...@192.168.10.27 - no reply to our critical packet (see doc/sip-retransmit.txt). == Spawn extension (netsoluciones, *981, 2) exited non-zero on 'SIP/192.168.10.3-09443498' regardss -- rickygm http://gnuforever.homelinux.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
