RE: [Asterisk-Users] Astlinux AMP
I've looked into this. The important reasons as to 'why this shouldn't happen' are: Requires a Database - (bad for flash, also very large) Needs apache + php (+30 odd mb) A fair whack of perl modules (+10mb) == Too large, too cumbersome. --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Thursday, May 12, 2005 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Astlinux AMP Callum McGillivray wrote: Hi all, Has anyone had experience with installing AMP on a soekris box running Astlinux? Is it possible ? Cheers, Callum Callum, While technically being possible, it is not easy, at least not at this point. The laundry list of required software for AMP makes it very difficult to run in a trimmed down environment like AstLinux. AMP and it's required software is probably bigger than all of AstLinux... Someone, someday will probably build a package for it, but it doesn't seem like a viable solution right now. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] is it allowed to install 2 TE405P cards at same P.C.?
Hi, the limits from te zaptel.h: #define ZT_MAX_SPANS128 /* Max, 128 spans */ #define ZT_MAX_CHANNELS 1024/* Max, 1024 channels */ On Wed, 2005-05-11 at 10:45 -0500, Carlos Chavez wrote: On Wed, 11 May 2005 07:31:17 +0300, Yousri Farouk wrote Hello Does Asterisk allow to install two pci TE405P Cards at the same P.C.? You should be able to do it as the limit for Zap channels is 255. I do not know if the computer can handle the interrupt load generated by two cards. -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Icecast
Hi, does anyone know of * being used with icecast in any way. Does * support vorbis at all? can anyone who is working on this give me a shout. Shidan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] octtel SP 4220 gateway and Asterisk
Hi Peoples I would be interested to hear from anyone who has managed to get the Octtel SP4220 and asterisk talking together. I am using the Octtel as a gateway for a PSTN line. It passes the call on to Asterisk and then Asterisk moves the call to a particular extension. Whilst I can get it to do this, there is no sound. It appears that Asterisk is trying to create a native bridge between the gateway and the sip extension. But this fails, or connects but again with no sound. I would appreciate any help at all. You are welcome to email msn/yahoo me Regards Scott k [EMAIL PROTECTED] [EMAIL PROTECTED] (MSN) [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
mine, on the stars of saturn options: Dione, Rhea, Titan, Mimas, Enceladus, Tethys, Hyperion, Iapetus, and Phoebe Abhishek -- Drishti-Soft Solutions Pvt Ltd http://www.drishti-soft.com On 5/12/05, Christopher Stephens [EMAIL PROTECTED] wrote: Mine is called 'blacksun', as that's where it's colo'd. (idiocy in a naming convention, I know.) On Wed, 11 May 2005 19:55:36 -0700 (PDT), Matt Klein [EMAIL PROTECTED] said: Mine is named spike... On Thu, 12 May 2005, Paul Hales wrote: We bought one of those books on the worst cars ever made...every page has great names... PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Thursday, 12 May 2005 1:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Andrew Latham Subject: Re: [Asterisk-Users] What do you name yours On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: Naming Conventions for Asterisk Hostnames, . For an internal historical reason all ours come from the legends of Robin Hood. I used to work with a bunch of Lord of the Rings readers and all the machine names came from there. It always makes a good light discussion point. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Astlinux AMP
[EMAIL PROTECTED] wrote: I've looked into this. The important reasons as to 'why this shouldn't happen' are: Requires a Database - (bad for flash, also very large) Needs apache + php (+30 odd mb) A fair whack of perl modules (+10mb) == Too large, too cumbersome. You can have PBXware on it in native mode with no installation curve of any kind. Just plug compact flash, fire up the browser and you are off :) Please contact Kris for details. Senad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astlinux AMP
Hi Guys, I actually had the idea in mind that the database would be located off site... (not on the actual machine). Still, with a larger flash card, would this not be possible ? (lol - getting a larger flash card is not going to be an issue) Just an thought. Callum Rob Thomas wrote: I've looked into this. The important reasons as to 'why this shouldn't happen' are: Requires a Database - (bad for flash, also very large) Needs apache + php (+30 odd mb) A fair whack of perl modules (+10mb) == Too large, too cumbersome. --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Kristian Kielhofner Sent: Thursday, May 12, 2005 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Astlinux AMP Callum McGillivray wrote: Hi all, Has anyone had experience with installing AMP on a soekris box running Astlinux? Is it possible ? Cheers, Callum Callum, While technically being possible, it is not easy, at least not at this point. The laundry list of required software for AMP makes it very difficult to run in a trimmed down environment like AstLinux. AMP and it's required software is probably bigger than all of AstLinux... Someone, someday will probably build a package for it, but it doesn't seem like a viable solution right now. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astlinux AMP
Senad, the specs on the site for the minimum version seem to indicate a HDD of at least 2Gb. Am I wrong here... is there something that I am not seeing ? Also... PBXware costs money and I don't want my cheap $1,000 unit to become a $2,000 unit. Any info would be appreciated. Callum Senad J wrote: [EMAIL PROTECTED] wrote: I've looked into this. The important reasons as to 'why this shouldn't happen' are: Requires a Database - (bad for flash, also very large) Needs apache + php (+30 odd mb) A fair whack of perl modules (+10mb) == Too large, too cumbersome. You can have PBXware on it in native mode with no installation curve of any kind. Just plug compact flash, fire up the browser and you are off :) Please contact Kris for details. Senad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP 2000 Conference Button and ILBC
Guys. I just downloaded the recent firmware for GS GXP 2000 and I must say the phone works great but... How do you make the conf button work?? Anybody done that? Also, with great dissapointment I must ask, where is ILBC support? GS web page mentions it and the manual says it supports it almost using bolds :) soo where is it Any light on this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Problem with MeetMe
In article [EMAIL PROTECTED], Daniel Salama [EMAIL PROTECTED] wrote: I would really hate to having to install a digium card just for the timer source. [...] I'd rather stay away from building custom kernels. Any other suggestions? No. If you don't have UHCI USB, those are the only two options, unless you change to a 2.6 kernel. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] beginner in Asterisk configuration
hello, i am french student and i want configure a Asterisk server. when I want launch the server with the command safe_asterisk -vcf the server answer : Asterisk ended with exit status 1 Asterisk died with code 1 what is the signification of it please ? thank you lucas _ MSN Hotmail : antivirus et antispam gratuits http://www.imagine-msn.com/hotmail/default.aspx?locale=fr-FR ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] beginner in Asterisk configuration
Are there any errors from /var/log/asterisk/messages? Or /var/log/messages? Can you give some output from the startup when you execute asterisk? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tutu Lord Sent: Thursday, May 12, 2005 12:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] beginner in Asterisk configuration hello, i am french student and i want configure a Asterisk server. when I want launch the server with the command safe_asterisk -vcf the server answer : Asterisk ended with exit status 1 Asterisk died with code 1 what is the signification of it please ? thank you lucas _ MSN Hotmail : antivirus et antispam gratuits http://www.imagine-msn.com/hotmail/default.aspx?locale=fr-FR ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.8 - Release Date: 5/10/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snap, Crackle and Pop with Dell 1850 and TE410P
Hi, In regards to the previous thread about static and snapping on incoming calls to the TE410P card when using a Dell 1850 server I now seem to be getting significantly better call quality with two E100P cards. So far I haven't been able to make any calls with detectable static on the line. Regards, Aaron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Predictive Dialers
If I can help in beta testing or anything, please let me know Matt. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of mattf |Sent: Miércoles, 11 de Mayo de 2005 10:15 p.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Predictive Dialers | |Thanks for the suggestion, we are planning a complete overhaul |of the user interface later this summer. We definitely want to |add more color and buttons and make it look less like a |depressing grey utility box. | |MATT--- | |-Original Message- |From: Anton Krall [mailto:[EMAIL PROTECTED] |Sent: Wednesday, May 11, 2005 10:27 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Predictive Dialers | | |For example, vicidial has great features but the screenshoots |for windows show too much info on the screen for end users to |learn to like.. The screen is more admin oriented. End users |would want more buttons, etc.. For example, take a look at |Altigen.com interfaces.. Very end user oriented.. |But Im looking for something open source or more accesible. | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of mattf ||Sent: Miércoles, 11 de Mayo de 2005 03:53 p.m. ||To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ||Subject: RE: [Asterisk-Users] Predictive Dialers || ||Could you let us know what you would consider a 'friendlier user ||interface'? || ||MATT--- || ||-Original Message- ||From: Anton Krall [mailto:[EMAIL PROTECTED] ||Sent: Wednesday, May 11, 2005 4:26 PM ||To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ||Subject: RE: [Asterisk-Users] Predictive Dialers || || ||I like vicidial's features but Im looking for a friendlier user ||interface.. || |||-Original Message- |||From: [EMAIL PROTECTED] |||[mailto:[EMAIL PROTECTED] On Behalf Of mattf |||Sent: Miércoles, 11 de Mayo de 2005 01:22 p.m. |||To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |||Subject: RE: [Asterisk-Users] Predictive Dialers ||| |||What exactly are you looking for? ||| |||There are basically 3 commercial solutions: Aheeva, DACX and ||Sinedialer |||and there are 2 open-source solutions: ShadyDial and VICIDIAL ||| |||What features do you need that are not addressed by one of these? ||| |||MATT--- ||| ||| |||-Original Message- |||From: Anton Krall [mailto:[EMAIL PROTECTED] |||Sent: Wednesday, May 11, 2005 1:09 PM |||To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |||Subject: RE: [Asterisk-Users] Predictive Dialers ||| ||| |||I took a look but was wondering if there are any other options out |||there? ||| -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nuno Viegas Sent: Miércoles, 11 de Mayo de 2005 04:27 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Predictive Dialers Hi Anton, Start by having a look at this: http://www.voip-info.org/wiki-Predictive+dialer N -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: 11 May 2005 10:19 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Predictive Dialers Guys. Anybody know of any predictive dialers for Asterisk and Windows? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ||| |||___ |||Asterisk-Users mailing list |||Asterisk-Users@lists.digium.com |||http://lists.digium.com/mailman/listinfo/asterisk-users |||To UNSUBSCRIBE or update options visit: ||| http://lists.digium.com/mailman/listinfo/asterisk-users |||___ |||Asterisk-Users mailing list |||Asterisk-Users@lists.digium.com |||http://lists.digium.com/mailman/listinfo/asterisk-users |||To UNSUBSCRIBE or update options visit: ||| http://lists.digium.com/mailman/listinfo/asterisk-users ||| ||| || ||___ ||Asterisk-Users mailing list ||Asterisk-Users@lists.digium.com ||http://lists.digium.com/mailman/listinfo/asterisk-users ||To UNSUBSCRIBE or update options visit: || http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco AS5300 or 3600
Hi,I used C3640, but It was changed,
because of few DSP in it. However, configuration is same. It also depends on
used IOS version. Here are fragments from
configurations:AS5300:!clock timezone GMT
0
; in some Docs = necessary!isdn switch-type
primary-net5 ; I`m in Europe :-)isdn voice-call-failure
0!!voice call send-alertvoice rtp send-recv!voice
service voip!voice class codec 3codec preference 1
g711alawcodec preference 2 g711ulaw!controller E1
0clock source line primarypri-group timeslots
1-31description to-PSTN!translation-rule
2
; type of number (subs/national/international) depend on your telco
providerRule 0 02 ANY subscriberRule 10 any
02 ANY subscriber!!translation-rule
10
; type of number (subs/national/international) depend on your telco
providerRule 0 ^42120... 0 ANY subscriberRule 1
^42121... 1 ANY subscriberRule 2 ^42122... 2 ANY
subscriberRule 3 ^42123... 3 ANY subscriberRule 4
^42124... 4 ANY subscriberRule 5 ^42125... 5 ANY
subscriberRule 6 ^42126... 6 ANY subscriberRule 7
^42127... 7 ANY subscriberRule 8 ^42128... 8 ANY
subscriberRule 9 ^42129... 9 ANY subscriberRule 10 any
1234 ANY subscriber!interface Serial0:15description
PRI-D-CHANNEL-to-PSTNno ip addressno logging event
link-statusisdn switch-type primary-net5isdn guard-timer
3000isdn map address 0.* plan isdn type subscriberisdn
send-alertingisdn sending-completeno cdp
enable!voice-port 0:Dinput gain -6output attenuation
14echo-cancel coverage 32echo-cancel
suppressorcptone SKdescription E1bearer-cap
Speech!dial-peer voice 8 potstone ringback
alert-no-PIdestination-pattern 00Tport 0:Dprefix
00!dial-peer voice 10 potstone ringback
alert-no-PIdestination-pattern 0[1-9]port
0:Dprefix 00421!dial-peer voice 20 potstone ringback
alert-no-PIdestination-pattern 00421[1-9]port
0:Dprefix 00421!dial-peer voice 999
voipnumbering-type internationalincoming called-number
.voice-class codec 3session protocol
sipv2dtmf-relay cisco-rtp h245-signal h245-alphanumericfax
rate 7200ip qos dscp cs5 mediano
vadsupplementary-service pass-through!dial-peer voice 1
potsincoming called-number .direct-inward-dialport
0:D!dial-peer voice 4212 voipdestination-pattern
4212translate-outgoing called 10voice-class codec
3session protocol sipv2session target
ipv4:1.2.3.4:5060 ; IP address of Asteriskip qos
dscp cs5 mediano vad!sip-uaretry invite
3retry response 3retry bye 3retry cancel
3timers trying 1000sip-server
ipv4:1.2.3.4:5060 ; IP address of Asterisk!ntp server
1.2.3.5!I`m not sure, if all things are necessary and correct,
but... it`s working :-). I can place calls from asterisk to PSTN via AS5300,
and also receive calls from pstn. In this configuration, i have DDI
prefix from my telco as 4212. 421 = international prefix 2
(02) = national prefix, is my DDI prefix in which i can use 10 000
numbers.I`m using 4 digit extensions in my numbering plan at Asterisk,
so I could have DID in 1:1 mapping.Fragments of very simple asterisk
configurations:Extensions.conf[globals]CISCOSIPGW=2.2.2.2
;(IP address of AS5300)[outgoing-cisco-pstn]exten =
_90N,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],180) ; local
callsSip.conf[2.2.2.2]type=friendhost=2.2.2.2nat=nocanreinvite=yesdtmfmode=rfc2833disallow=allallow=alawallow=ulawIn
this cas, only 10 digit numbers are allowed (only national calls) to dial
via Cisco, through number 9 as an prefix for outbound calls.Hope,
that this samples will be usefull for you.PS: sorry for english, i
hope, you could understand it :-)-b- Original
Message - From: "Anton Krall" [EMAIL PROTECTED]To:
[EMAIL PROTECTED]Sent: Wednesday, May 11, 2005 7:08 PMSubject: RE:
[Asterisk-Users] Asterisk and Cisco AS5300 or 3600 Hey
Barney What are the steps necessary to make that work on the
cisco AS5300? Any configs I need to check to make it work? And what do I
need on asterisks side? Ever used cisco
3600? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of
barney |Sent: Miércoles, 11 de Mayo de 2005 05:22 a.m. |To:
Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re:
[Asterisk-Users] Asterisk and Cisco AS5300 or 3600 | | Just
in case you don't know, AS5350 supports SIP *and* H323 |after
IOS | version | 12.3 (maybe a little earlier).
| It allows you to use both at the same time, without needing |to
set it | up for one system specifically. | Haven't tried
it with Asterisk yet though. | | |I have tried it. I
have SIP trunk between Asterisk and AS5300 |(C3640 before), and it`s
working good. |It`s quite good solution, but its much more expensive as
some |PCI card direct in Asterisk (i`m using PRI interconnect to
PSTN). | |-b | |PS: sorry for poor
english | | | | On Wednesday 11 May 2005
11:23, Anton Krall wrote: | I need some advice on some h323
issues. I need to test connectivity | from Asterisk to a Cisco
RE: [Asterisk-Users] beginner in Asterisk configuration
I have download Astwind 0.1.1, my config is without Zaptel card and is mde up of one computer without client which is connected on my extensions.conf is : [general] static=yes writeprotect=no [globals] [echotest] exten = 600,1,Wait(3) exten = 600,2,Echo [local] ignorepat = 9 include = echotest thank you _ MSN Messenger : vidéoconférence gratuite http://g.msn.fr/FR1001/866 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Astlinux AMP
Callum McGillivray wrote: Senad, the specs on the site for the minimum version seem to indicate a HDD of at least 2Gb. Am I wrong here... is there something that I am not seeing ? That was a type error... corrected. thanks :) Also... PBXware costs money and I don't want my cheap $1,000 unit to become a $2,000 unit. Contact Kris for more details please... :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] beginner in Asterisk configuration
Run asterisk -vcf for test purpose. Safe_asterisk is script which runs asterisk - so you wont get any messages on screen Br, dmitry -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Tutu Lord Sendt: 12 May 2005 09:58 Til: asterisk-users@lists.digium.com Emne: [Asterisk-Users] beginner in Asterisk configuration hello, i am french student and i want configure a Asterisk server. when I want launch the server with the command safe_asterisk -vcf the server answer : Asterisk ended with exit status 1 Asterisk died with code 1 what is the signification of it please ? thank you lucas _ MSN Hotmail : antivirus et antispam gratuits http://www.imagine-msn.com/hotmail/default.aspx?locale=fr-FR ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What do you name yours
Since we have SIP and ZAP servers, we name them something completely original: SIP01 ZAP01 ZAP02 TEST01 Whoa! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Abhishek Tiwari Sent: Thursday, May 12, 2005 12:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] What do you name yours mine, on the stars of saturn options: Dione, Rhea, Titan, Mimas, Enceladus, Tethys, Hyperion, Iapetus, and Phoebe Abhishek -- Drishti-Soft Solutions Pvt Ltd http://www.drishti-soft.com On 5/12/05, Christopher Stephens [EMAIL PROTECTED] wrote: Mine is called 'blacksun', as that's where it's colo'd. (idiocy in a naming convention, I know.) On Wed, 11 May 2005 19:55:36 -0700 (PDT), Matt Klein [EMAIL PROTECTED] said: Mine is named spike... On Thu, 12 May 2005, Paul Hales wrote: We bought one of those books on the worst cars ever made...every page has great names... PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Thursday, 12 May 2005 1:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Andrew Latham Subject: Re: [Asterisk-Users] What do you name yours On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: Naming Conventions for Asterisk Hostnames, . For an internal historical reason all ours come from the legends of Robin Hood. I used to work with a bunch of Lord of the Rings readers and all the machine names came from there. It always makes a good light discussion point. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.8 - Release Date: 5/10/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime voicemail login incorrect
Adam Goryachev wrote:
On Thu, 2005-05-12 at 07:35 +0800, Ronald Wiplinger wrote:
I have two ways to go to the voicemail box, either by dialing 8500
from the phone which received the voicemail (without a password) or
from another phone by dialing 8501 and key in the mailbox and the
password. However, with Realtime the password will be rejected as
login incorrect. What do I miss?
exten = 8500,1,VoicemailMain(s${CALLERIDNUM})
exten = 8500,2,hangup
exten = 8501,1,VoicemailMain
exten = 8501,2,hangup
AFAIK, you *must* specify the context when using realtime, even if the
context is default. Change the above to:
exten = 8500,1,VoicemailMain([EMAIL PROTECTED])
exten = 8500,2,hangup
exten = 8501,1,VoicemailMain(@default)
exten = 8501,2,hangup
Or something like that I assume.
Regards,
Adam
Adam,
it seems you are right, you need the context there, but than I cannot
use it in Realtime anymore, since I have more than one context, ..
I would than need for each context an extra extension number.
It makes no sense either, since one phone number should have anyway only
ONE context, or could be there a case that one could have more than one
context?
bye
Ronald
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice mail - Extension at vs Phone Number OGM
Is there a way to make an outside call hear The person at phone number is unavail, but when an internal extension calls another extension, they hear The person at extension number is unavail? I swear Ive read this somewhere before but Im not typing in the right search. I probably found it before by complete accident Of course, we want the outside caller to hear a phone number seven digits long, while an extension hears just that, an extension. Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice mail - Extension at vs Phone Number OGM
Chris Coulthurst wrote: Is there a way to make an outside call hear The person at phone number is unavail, but when an internal extension calls another extension, they hear The person at extension number is unavail? I swear Ive read this somewhere before but Im not typing in the right search. I probably found it before by complete accident Of course, we want the outside caller to hear a phone number seven digits long, while an extension hears just that, an extension. You can choose whatever you want, by using different context. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wrong password on Auth for Notify
I have this warning popping up on one particular server. chan_sip.c handle_response: Forbidden - Wrong password on authentication for Notify. I have looked around but cannot find what would be the cause of the warning? Can anyone throw some light on this warning, why it is caused? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Icecast
I know you can use slimserver as a music source, and slimserver supports tons of formats, so maybe that's your answer. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shidan Sent: Thursday, May 12, 2005 2:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Icecast Hi, does anyone know of * being used with icecast in any way. Does * support vorbis at all? can anyone who is working on this give me a shout. Shidan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco contract for 7940/7960 firmware access
Maybe not the place for this but thought I'd post the info for others. I purchased a cisco 7960 off ebay and needed to convert to SIP for *. I know * supports SCCP but I wont go into that here. I'd read on voip-info.org that a contract could be purchased for approx $8 to allow me to download the firmware. I though, being in the UK, i'd get one through a reseller in the UK. What a shock! Apparently, the $8 ( £5 ) contract is no longer available and the cheapest contract they provide with firmware download access is about £56 ( $104 ). Just like to say thanks to the kind soul who helped me out with the firmware ( you know who you are! ). Just waiting A WEEK for the email from cisco with my contract number and then I'll have to wait ANOTHER WEEK while they verify the number that THEY are sending me! Rant over! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wrong password on authentication for Notify
I have this warning popping up on one particular server. chan_sip.c handle_response: Forbidden - Wrong password on authentication for Notify. I have looked around but cannot find what would be the cause of the warning= ? Can anyone throw some light on this warning, why it is caused? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AreskiCC Install Problems
Those files I indicated to check : /var/lib/pgsql/data (on a redhat flavor) pg_hba.conf - This one needs lines similar to local all all password host all all0.0.0.0 0.0.0.0 password (not you probably want a more restrictive ip range / net mask here!!) postgresql.conf make sure it has a line tcpip_sockets=true Make sure you have the following packages rh-postgres-server php-pgsql or the files containted within Finally, if you haven't, make sure you restart both postgres and apache to ensure they have seen the changes to the config (apache needs to see the updates containted within php-pgsql as an after thought, it is required that php-globals=on, I have never had to set that and am not sure which file its in (I do belive however that it refers to an apache config file not an areski one) As a hope thought - I have sucsessfuly got both versions of areskicc working at some point, so its not flawed code. On 5/11/05, Julius Igugu [EMAIL PROTECTED] wrote: Make sure postgresql is running and the database username/passwords are correct. --- Kanuri, Seshu (Company IT) [EMAIL PROTECTED] wrote: I have followed the Idiots' guide for installation, but still could not make it work. When I try to login at the web page coming from /var/www/html/areski , I get the following errors: Can some body give me some hints where and what to check for this error?. I am looking for info on the changes we have to make for 1) the database name 2) user name 3) password 4)connection name (server running postgresql) in all the files involved in the application, so that it works. Seshu --- Warning: pg_pconnect(): Unable to connect to PostgreSQL server: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? . in /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 68 Database error: Link-ID == false, pconnect failed PostgreSQL Error: 0 () Warning: pg_pconnect(): Unable to connect to PostgreSQL server: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? . in /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 68 Database error: Link-ID == false, pconnect failed PostgreSQL Error: 0 () Warning: pg_errormessage(): supplied argument is not a valid PostgreSQL link resource in /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 101 Warning: Cannot modify header information - headers already sent by (output started at /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php:68) in /var/www/html/areskicc/lib/module.access.php on line 66 Warning: Cannot modify header information - headers already sent by (output started at /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php:68) in /var/www/html/areskicc/lib/module.access.php on line 67 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Julius Igugu SouthWork Co. Ltd. __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail Passwords
yep, I think you're right that the voicemail.conf file is being
dynamically rebuilt. The reason that was not being reflected before is
that I had the voicemail.conf file open and therefore asterisk could not
write to it. However, I noticed that when I closed it and re-opened it,
that the changes to the password were reflected just as you surmised.
So that solves my question below.
Thanks!
Jeff Heath
On Wed, 2005-05-11 at 19:33, BJ Weschke wrote:
Looking at app_voicemail.c with the copy I have here, it looks like
vm_change_password is trying to dynamically rebuild the voicemail.conf
file. It writes a voicemail.conf.new file, and then replaces one with
the other once it's done.
What version of asterisk are you running?
Do you get an WARNINGs or any other kind of logging info when you
reset the password? Looking at the code, it's supposed to issue
warnings if it cannot open the old file for read and/or open the new
file for write.
On 5/11/05, BJ Weschke [EMAIL PROTECTED] wrote:
I see what you're saying. Unless someone else responds with the
issue, I'll look at the code a little later this evening. It sounds
like the changed password via IVR is going into the ast-db, and then
that new value is ignoring what's in voicemail.conf.
On 5/11/05, Jeff Heath [EMAIL PROTECTED] wrote:
On Tue, 2005-05-10 at 21:25, BJ Weschke wrote:
voicemail.conf
edit that file and issue a reload to change them.
I tried this, but I still can't get access to voicemail from one of the
phones.
This is a test system that I setup about a month ago. Got busy and am
just now getting back to it. I have 2 SIP phones and the Asterisk
server. The default voicemail password is 1234 for both extensions. I
changed the password for one of them and (doh!) forgot/lost it.
Since this is a test system, I tried an experiment. I went into the
phone where I can get access to voicemail, and I manually changed the
password from 1234 to 4567. Then I issued a reload (the default
passwords in voicemail.conf are 1234). Then I accessed voicemail again,
and the password is 4567 not 1234.
This makes sense to me. Otherwise, every time asterisk was restarted or
reloaded all the user's personal voicemail passwords would be reset.
Surely, I'm not the first dope that's changed a password and forgot it
:-)
I can't believe there's not a file somewhere that the administrator can
directly edit to change user voicemail passwords, but I've been
searching the Wiki and googling on lists.digium.com and searched all the
Asterisk documentation I can find and I can't find it.
So, how does the administrator reset a user's password?
fyi, here are my extensions.conf and voicemail.conf
extensions.conf
[general]
static = yes
writeprotect = yes
[from-sip]
exten = 4035,1,Dial(SIP/4035,20)
exten = 4035,2,Voicemail(u4035)
exten = 4035,102,Voicemail(b4035)
exten = 4035,103,Hangup
exten = 4009,1,Dial(SIP/4009,20)
exten = 4009,2,Voicemail(u4009)
exten = 4009,102,Voicemail(b4009)
exten = 4009,103,Hangup
; This defines the number to access VM.
; The caller's extension number is passed as a variable, so
; all the user needs to do is type in the password.
exten = 4040,1,VoicemailMain(${CALLERIDNUM})
[local]
include = from-sip
voicemail.conf
[general]
format = wav49|gsm|wav
serveremail = asterisk
attach = yes
maxmessage = 180
maxgreet = 60
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 3
[default]
4009 = 1234,Jeff
4035 = 1234,Pam
On 5/10/05, Jeff Heath [EMAIL PROTECTED] wrote:
Where are user's voicemail passwords stored and how does the asterisk
administrator change them?
TIA,
Jeff Heath
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360
Colin Similar to Gary's response in that I haven't seem many of these issues. One that is similar, is that of you saying you need to press voicemail key twice to get *97 (or eqivilent code) This as I understand it is not a fault of snom, but a feature of asterisk and the whole MWI protocol. When asterisk signals the phone to say it has voicemail (any phone) it sends in from an address of [EMAIL PROTECTED]. the message text is basically that which pops up on the bottom line of the display. When you press the voicemail key, or even the soft voicemail key it first tries to make contact with unknown as this helps ensure that the right line acesseses its voicemail without the user having to be aware of which line the voicemail is waiting for them on. You have two choices, a change the address of the MWI indicator to come from [EMAIL PROTECTED] on the asterisk box or add some lines in your message-centre context that is similar to exten = Unknown,1,Voicemail etc Either of these will bring asterisk up to the level of the snoms features. I have only one minor issue, and thats if I have several people ringing into the phone, when I am not already on a call (all calls are still in the setup phase) I can't choose by pressing the flashing lights, I have to dump them using the soft no thanks or the hard x key You almost sound like you have a earlier firmware issue. The latest one is 3.60f a direct link to the firmware is http://www.snom.com/download/share/ I tell a lie -the very latest firmware is 3.60h - as of the 4th May David On 5/12/05, Gary Stimson [EMAIL PROTECTED] wrote: Hi Colin I've been using a Snom 360 for 2 weeks and am generally pleased with it. On Wednesday 11 May 2005 22:12, Colin E. McDonald wrote: I am having major problems with the first run of Snom 360s that rolled out last month. Issues: Speakerphone/Hands Free volume spikes up and down during a call. Haven't seen that problem. You have to manually set the volume during every call. When you set the volume, press OK. Then it's stored for next time. This makes it totally unusable. The sound will cut out completely at the beginning of a call sporadically. Have you tried a different provider? Call comes through speaker phone after you pick up handset and then cuts to handset a couple of seconds later I don't have that issue. There is a mnaufacturing defect where the display cable is disconnected so you get what appears to be DOA desk sets. Nor that one. Maybe I was lucky! Have to press the retrieve message button twice pretty regularly to get it to dial vociemail (*97) in asterisk Haven't got the VM button configured yet, or tried to. Major problem with calls being dropped when you place callers on hold I haven't tried putting callers on hold yet. Have you updated to the latest firmware? Copy the firmware URL from snom.com into the relevant box on the phone's web interface, save and reboot the phone. Gary -- Gary Stimson Zedcore Systems ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GXP2000 firmware update
Thx for the pointer Peter. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Peter Bowyer |Sent: Jueves, 12 de Mayo de 2005 12:45 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Grandstream GXP2000 firmware update | |On 11/05/05, Anton Krall [EMAIL PROTECTED] wrote: | I just downloaded the zip file from grandstreams website to |upgrade my | gxp2000 firmware from 1.0.0.3 to the latest but seems there are some | files missing on the zip file... Anybody been able to |upgrade their firmware? | | My website shows this files as missing: | | 201.133.125.152 - - [11/May/2005:16:47:16 -0500] GET | /firmware/ring1.bin HTTP/1.0 200 12737 - Grandstream |GXP2000 1.0.0.3 | [Wed May 11 16:47:17 2005] [error] [client 201.133.125.152] |File does | not | exist: /usr/local/apache/htdocs/voip/firmware/ring2.bin | 201.133.125.152 - - [11/May/2005:16:47:17 -0500] GET | /firmware/ring2.bin HTTP/1.0 404 289 - Grandstream |GXP2000 1.0.0.3 | [Wed May 11 16:47:18 2005] [error] [client 201.133.125.152] |File does | not | exist: /usr/local/apache/htdocs/voip/firmware/ring3.bin | 201.133.125.152 - - [11/May/2005:16:47:18 -0500] GET | /firmware/ring3.bin HTTP/1.0 404 289 - Grandstream |GXP2000 1.0.0.3 | [Wed May 11 16:47:19 2005] [error] [client 201.133.125.152] |File does | not | exist: /usr/local/apache/htdocs/voip/firmware/cfg000b8200 | 201.133.125.152 - - [11/May/2005:16:47:19 -0500] GET | /firmware/cfg000b8200 HTTP/1.0 404 295 - Grandstream GXP2000 | 1.0.0.3 | [Wed May 11 16:47:21 2005] [error] [client 201.133.125.152] |File does | not | exist: /usr/local/apache/htdocs/voip/firmware/cfg.txt | 201.133.125.152 - - [11/May/2005:16:47:21 -0500] GET | /firmware/cfg.txt HTTP/1.0 404 287 - Grandstream GXP2000 1.0.0.3 | |I *think* those files are optional - custom ringtones and |MAC-specific config. | |I left my TFTP server pointed to 168.75.215.188, and the |phones upgraded themselves to v 1.0.1.6 without intervention | |Peter |-- |Peter Bowyer |Email: [EMAIL PROTECTED] |Tel: +44 1296 768003 |VoIP: sip:[EMAIL PROTECTED] |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Live Voip
I've got an 800 number through livevoip and have not noticed any failures (incoming or outgoing). There certainly could have been a failure once in a while, just have not heard or observed it. Of the several itsp's I've tried over the last six to twelve months, its been the most stable and responsive. Same for teliax.com, which I have several did numbers. Search the list... To me, they're good enough for call-termination at this point, but not reliable (or available) enough to receive my inbound traffic. -Original Message- Hi all, Before I setup an account with them, I'd like to hear other people's impression of LiveVoip. I'm considering using them for 800 numbers, and I'd like to feel comfortable that others here on the list have had good experiences with them. Thanks, sorry if this is the wrong list for this. :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Making Asterisk run on Mysql backend
Hello there, I have configured my asterisk to run on Mysql backend. But the Asterisk was unable to pick the peer details from the database. This is how I configured the Asterisk to run with mysql on the backend. Edit /usr/src/asterisk/channels/Makefile, change it to enable the MYSQL_FRIENDS USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 cd /usr/src/asteriskmake cleanmake make install Created the database with the following structure: CREATETABLE`sipfriends`( `name`varchar(40)NOTNULLdefault'', `username`varchar(40)default'', `secret`varchar(40)NOTNULLdefault'', `context`varchar(40)NOTNULLdefault'', `ipaddr`varchar(20)NOTNULLdefault'', `port`int(6)NOTNULLdefault'0', `regseconds`int(11)NOTNULLdefault'0', PRIMARYKEY(`name`) )TYPE=MyISAM; And there was no dial tone on the phone. Can you please tell me as to whats wrong in the configuration and how as to how does asterisk fetch the details of the peers from the database. Thanks in advance Regards, Bharat M. Sarvan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Status of FAX
Yep it's called Jfax but it's a commercial service that there is a charge for. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Coulthurst Sent: Wednesday, 11 May 2005 11:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Status of FAX What about some sort of 'fax proxy'? Could an internal extension, set for a fax machine, really send the fax to a middleman internally, and have that fax middleman resend the fax via a pots line, to eliminate net latency? Not sure if this is a viable option, but somehow it seems sane... Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Jay Milk |Sent: Wednesday, May 11, 2005 2:13 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Status of FAX | |http://www.voip-info.org/wiki-Asterisk+fax | |May work if you force G.711 (alaw/ulaw) codec -- that's what Vonage |does, and it works with *some* fax machines. | | -Original Message- | From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] | Sent: Wednesday, May 11, 2005 3:19 PM | To: asterisk-users@lists.digium.com | Subject: [Asterisk-Users] Status of FAX | | | Hi people, what is the current status of send/receive fax on | asterisk extensions, i dont want to receive the fax and send | an email or viceversa, i want to connect a standard fax | machine to a Linksys' ATA (FXS RJ11 port) . Webdoc?, pointers? Thanks | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/aster isk-users | To | UNSUBSCRIBE or update options visit: | |http://lists.digium.com/mailman/listinfo/asterisk-users | | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco AS5300 or 3600
This has been great !! Thx Barney
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
barneySent: Jueves, 12 de Mayo de 2005 03:30 a.m.To:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Asterisk and Cisco AS5300 or 3600
Hi,I used C3640, but It was changed,
because of few DSP in it. However, configuration is same. It also depends
on used IOS version. Here are fragments from
configurations:AS5300:!clock timezone GMT
0
; in some Docs = necessary!isdn switch-type
primary-net5 ; I`m in Europe :-)isdn voice-call-failure
0!!voice call send-alertvoice rtp send-recv!voice
service voip!voice class codec 3codec preference 1
g711alawcodec preference 2 g711ulaw!controller E1
0clock source line primarypri-group timeslots
1-31description to-PSTN!translation-rule
2
; type of number (subs/national/international) depend on your telco
providerRule 0 02 ANY subscriberRule 10 any
02 ANY subscriber!!translation-rule
10
; type of number (subs/national/international) depend on your telco
providerRule 0 ^42120... 0 ANY subscriberRule 1
^42121... 1 ANY subscriberRule 2 ^42122... 2 ANY
subscriberRule 3 ^42123... 3 ANY subscriberRule 4
^42124... 4 ANY subscriberRule 5 ^42125... 5 ANY
subscriberRule 6 ^42126... 6 ANY subscriberRule 7
^42127... 7 ANY subscriberRule 8 ^42128... 8 ANY
subscriberRule 9 ^42129... 9 ANY subscriberRule 10 any
1234 ANY subscriber!interface Serial0:15description
PRI-D-CHANNEL-to-PSTNno ip addressno logging event
link-statusisdn switch-type primary-net5isdn guard-timer
3000isdn map address 0.* plan isdn type subscriberisdn
send-alertingisdn sending-completeno cdp
enable!voice-port 0:Dinput gain -6output
attenuation 14echo-cancel coverage 32echo-cancel
suppressorcptone SKdescription E1bearer-cap
Speech!dial-peer voice 8 potstone ringback
alert-no-PIdestination-pattern 00Tport 0:Dprefix
00!dial-peer voice 10 potstone ringback
alert-no-PIdestination-pattern 0[1-9]port
0:Dprefix 00421!dial-peer voice 20 potstone
ringback alert-no-PIdestination-pattern
00421[1-9]port 0:Dprefix 00421!dial-peer
voice 999 voipnumbering-type internationalincoming
called-number .voice-class codec 3session protocol
sipv2dtmf-relay cisco-rtp h245-signal h245-alphanumericfax
rate 7200ip qos dscp cs5 mediano
vadsupplementary-service pass-through!dial-peer voice 1
potsincoming called-number
.direct-inward-dialport 0:D!dial-peer voice
4212 voipdestination-pattern
4212translate-outgoing called 10voice-class codec
3session protocol sipv2session target
ipv4:1.2.3.4:5060 ; IP address of Asteriskip qos
dscp cs5 mediano vad!sip-uaretry invite
3retry response 3retry bye 3retry cancel
3timers trying 1000sip-server
ipv4:1.2.3.4:5060 ; IP address of Asterisk!ntp
server 1.2.3.5!I`m not sure, if all things are necessary and
correct, but... it`s working :-). I can place calls from asterisk to PSTN
via AS5300, and also receive calls from pstn. In this configuration, i
have DDI prefix from my telco as 4212. 421 = international
prefix 2 (02) = national prefix, is my DDI prefix in which i can
use 10 000 numbers.I`m using 4 digit extensions in my numbering plan
at Asterisk, so I could have DID in 1:1 mapping.Fragments of very
simple asterisk
configurations:Extensions.conf[globals]CISCOSIPGW=2.2.2.2
;(IP address of AS5300)[outgoing-cisco-pstn]exten =
_90N,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],180) ; local
callsSip.conf[2.2.2.2]type=friendhost=2.2.2.2nat=nocanreinvite=yesdtmfmode=rfc2833disallow=allallow=alawallow=ulawIn
this cas, only 10 digit numbers are allowed (only national calls) to dial
via Cisco, through number 9 as an prefix for outbound calls.Hope,
that this samples will be usefull for you.PS: sorry for english, i
hope, you could understand it :-)-b- Original
Message - From: "Anton Krall" [EMAIL PROTECTED]To:
[EMAIL PROTECTED]Sent: Wednesday, May 11, 2005 7:08 PMSubject: RE:
[Asterisk-Users] Asterisk and Cisco AS5300 or 3600 Hey
Barney What are the steps necessary to make that work on the
cisco AS5300? Any configs I need to check to make it work? And what do
I need on asterisks side? Ever used cisco
3600? |-Original Message- |From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of barney
|Sent: Miércoles, 11 de Mayo de 2005 05:22 a.m. |To: Asterisk Users
Mailing List - Non-Commercial Discussion |Subject: Re:
[Asterisk-Users] Asterisk and Cisco AS5300 or 3600 | |
Just in case you don't know, AS5350 supports SIP *and* H323 |after
IOS | version | 12.3 (maybe a little earlier).
| It allows you to use both at the same time, without needing |to
set it | up for one system specifically. | Haven't
Re: [Asterisk-Users] What do you name yours
On Wed, May 11, 2005 at 05:40:57PM +0200, Dave Cotton wrote: On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: For an internal historical reason all ours come from the legends of Robin Hood. I used to work with a bunch of Lord of the Rings readers and all the machine names came from there. It always makes a good light discussion point. So far we have only installed singular machines for clients. So I name them palantir. I wanted a good name that I could reuse and it would make sense. So we have [EMAIL PROTECTED] and [EMAIL PROTECTED] and [EMAIL PROTECTED], etc... Seemed like a cool thought at the time... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting * to a PBX throught a PRI.
Hi everybody, We are thinking in connect out PBX (with a new PRI card) to * (with card TE110P) thought an E1. We will have to configure the framing, coding, channels, etc...our doubt is: How must we select the signalling in * 'pri_cpe' or 'pri_net'? It's depend if our PBX card emulate to be the network side or the customer side? Thanks in advance. Regards. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] forum www.asterisk-italia.it
Matt Riddell wrote: For all italian speaking users please visit and contribute to www.asterisk-italia.it! I don't seem to be able to resolve that link. Sorry! Not a very good start :-) we had some dns propagation issues that are now solved. The website is online now... Thanks! Paolo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceBlue GSM
Hello All * users. I have been looking for a way to allow GSM termination through Asterisk to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on asterisk with the ZAP channels via the Digium TDM 400P. I am unable to find any place that can tell me the cost of the VoiceBlue with a currency to I can calculate the cost of buying one. Alternativly - or just out of interist - I only really need to handle one GSM call @ a time and have a SMS capability... is there anyone that can suggest the best way to do so without doing a hack/patch to make a device to interact with asterisk? -- Kind Regards Etienne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kphone--asterisk--Kphone
On Tue, May 10, 2005 at 12:01:17PM +0530, Sudhananda wrote: I am running asterisk on one linux PC and want to talk through this server using Kphone installed on 2 different PC's. These are the extra lines added to sip.conf and extensions.conf respectively. sip.conf [jitha] type=friend host=dynamic secret=jitha context=sip dtmfmode=inband [sudhananda] type=friend host=dynamic secret=sudhananda context=sip This is what I use for kphone and it works fine: [kphone] type=friend ; either friend (peer+user), peer or user host=dynamic ; we have a static but private IP address callerid=kphone 25 dtmfmode=inband ; either RFC2833 or INFO for the BudgeTone context=internal disallow=all ; need to disallow=all before we can use allow= allow=ulaw; Note: In user sections the order of codecs extensions.conf [sip] exten=1,1,Dial(SIP/jitha,20,tr) exten=2,1,Dial(SIP/sudhananda,20,tr) Both the Kphones got registered to the asterisk but when i dial the number it gives me the following log on asterisk Asterisk Ready. *CLI -- Registered SIP 'sudhananda' at 172.16.2.35 port 5060 expires 900 -- Executing Dial(SIP/sudhananda-aa77, SIP/jitha|20|tr) in new stack -- Called jitha -- SIP/jitha-f4bc is ringing -- SIP/jitha-f4bc answered SIP/sudhananda-aa77 -- Attempting native bridge of SIP/sudhananda-aa77 and SIP/jitha-f4bc I see no problems here yet. and one Kphone status is ringing and on other it is connected. how to solve this problem. You might want to check the codecs in use. Are they both on the local network? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] delay before execution of call file
hello i am using a call file. i want to insert delay before execution of this call file. any idea how to do this Channel: SIP/2000 MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: default Extension: 6000 Priority: 1 i am making a callback system. when person rings to callback number this call file is created now it is trying to call back when the call is already connected Kamran Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Status of FAX
I'm not finding that on the Jfax website. Can you point me to more info on how the act as a VOIP Fax Proxy? Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, May 12, 2005 7:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Status of FAX Yep it's called Jfax but it's a commercial service that there is a charge for. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Coulthurst Sent: Wednesday, 11 May 2005 11:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Status of FAX What about some sort of 'fax proxy'? Could an internal extension, set for a fax machine, really send the fax to a middleman internally, and have that fax middleman resend the fax via a pots line, to eliminate net latency? Not sure if this is a viable option, but somehow it seems sane... Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Jay Milk |Sent: Wednesday, May 11, 2005 2:13 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Status of FAX | |http://www.voip-info.org/wiki-Asterisk+fax | |May work if you force G.711 (alaw/ulaw) codec -- that's what Vonage |does, and it works with *some* fax machines. | | -Original Message- | From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] | Sent: Wednesday, May 11, 2005 3:19 PM | To: asterisk-users@lists.digium.com | Subject: [Asterisk-Users] Status of FAX | | | Hi people, what is the current status of send/receive fax on | asterisk extensions, i dont want to receive the fax and send an | email or viceversa, i want to connect a standard fax machine to a | Linksys' ATA (FXS RJ11 port) . Webdoc?, pointers? Thanks | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/aster isk-users To | UNSUBSCRIBE or update options visit: | |http://lists.digium.com/mailman/listinfo/asterisk-users | | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceBlue GSM
Etienne, I am not sure I understand all what you require. Do you need to know the cost of the voiceblue of 2N or you need to find solution that can allow you send GSM calls ? There are several alternatives: 1-) Voiceblue as you mentioned; 2-) You can buy a voip2GSM Gateway. To which you no longer need hardware, you just register the voip2GSM devise to Asterisk and then it is ready to receive and send calls just like any other sip phone. Cost of this is around 400 USD / UNIT When you talk about sms capability, dyou want to originate or receive SMSs through the devise? Selon Etienne Pretorius [EMAIL PROTECTED]: Hello All * users. I have been looking for a way to allow GSM termination through Asterisk to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on asterisk with the ZAP channels via the Digium TDM 400P. I am unable to find any place that can tell me the cost of the VoiceBlue with a currency to I can calculate the cost of buying one. Alternativly - or just out of interist - I only really need to handle one GSM call @ a time and have a SMS capability... is there anyone that can suggest the best way to do so without doing a hack/patch to make a device to interact with asterisk? -- Kind Regards Etienne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Show useragents?
When the phones register with asterisk Saved useragent Sipura/SPA841-3.1.2(d) for peer I can see the firmware, which is handy for ensuring they are all up to date. How can I list all the useragents? Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ITSPs with good phone support
With the recent service outage at Broadvoice, there has been a lot of discussion here, on broadband reports, Voxilla, etc., regarding whether VOIP is mature, or ready for the masses, etc. One particular point I've seen repeated, and with which I agree: we're willing to deal with less than five 9s, even one or 2 9s, as long as we have good communication regarding the issue and its resolution. In other words, good customer relations are as important or even more important than the highest quality of service. snip Who's the leading contender for customer service of the year award among the dozens of providers that show up on the Wiki? Seems the majority have been or were an isp at one time, and have implemented a defacto isp model for customer service. I'd have to guess that a fair number really don't have a clue what a good/reasonable target happens to be. The flip side of that is that good customer service staffing is expensive and is often times treated as an unwanted / under-planned / under-budgeted operating overhead that is viewed by many as the target for cost control, etc. For the few times that I've had to interact with livevoip.com and teliax.com support, both have been very responsive. Both still seem to emphasize the use of email for interaction, but at least my issues were resolved very quickly with that approach. (For both companies, as soon as they realized that I knew what the hell I was talking about, diagnosing the issue and resolving the problem occurred very quickly. I can just imagine how many calls/emails they get where their customer is reporting a problem that involves a total lack of skills, understanding, mis- configured BOYD equipment, etc. We certainly see it on this list!) If you read between the lines, its not difficult to see that many of the itsp's were started with a primary objective of being purchased by some other larger company. We've already seen some results of that in several recent forms. Those companies only do whatever they think is necessary to 'appear' solid, which also includes managing their customer service overhead to some reasonable level (defined in their minds, not ours). As the itsp space shakes out over time, the winners are likely to be those that do offer some reasonable customer service in combination with acceptable marketing plans, etc. Until that happens, customer service is likely to vary rather dramatically even within the same itsp. So, what you see for custoemr service today from an itsp may be rather different from what you see tomorrow or next week, and those changes can certainly be positive or negative. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
David John Walsh [EMAIL PROTECTED] wrote: I quite like the idea that came about earlier with regards to Romand and Greek gods, I am thinking (if I ever get off the phone to google today) of findind the roman and greek gods of communication.. You are thinking of Mercury and Hermes, the Roman and Greek names respectively for the same god. You may have heard of Mercury Communications Ltd., so your idea isn't entirely original ;) -- Room Service? Send up a larger room. - Groucho Marx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
it was a wheel. still went on it again an hour later once they put it back on!!! /never/ trust french theme parks :) As a consultant focusing primarily on network performance and security for the past twelve years, and working with clients in 40+ US States, we've seen - systems named after continents - systems named after species of fish - Nascar race tracks - Encoded name (State, City, Dept, Function, Number) - characters from Starwars - bodies of water - etc, etc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast_yyerror - 'space' in Caller-ID - string comparison
I've some code to manipulate incoming Caller-ID - so its suitable for
replying to...
[sipdef]
exten = s,1,NoOp(FWD SIP: ${CALLERIDNAME} ${CALLERIDNUM})
; Alter incoming calles from pulver - add a '87'
exten = s,2,Gotoif($[${CALLERIDNAME} = ${CALLERIDNUM}]?3:4)
exten = s,3,SetCIDName(87${CALLERIDNUM})
exten = s,4,SetCIDNum(87${CALLERIDNUM})
exten = s,5,Goto(default,s,1)
When Executing the above - and I presume incoming Caller Info looks like
the name is Mark Elkins and the Number is 638936...
The purpose is to prefix the number (only the number) with 87.
Sometimes, incoming CallerID data looks like -- 638936 638936
therefore the checking of both Name and number.
-- Executing NoOp(SIP/292951-b11f, FWD SIP: Mark Elkins
638936) in new stack
May 12 14:36:59 WARNING[28824]: ast_expr.y:486 ast_yyerror:
ast_yyerror(): syntax error: parse error; Input:
Mark Elkins = 638936
^
-- Executing GotoIf(SIP/292951-b11f, Mark?3:4) in new stack
-- Goto (sipdef,s,4)
-- Executing SetCIDNum(SIP/292951-b11f, 87638936) in new stack
-- Executing Goto(SIP/292951-b11f, default|s|1) in new stack
-- Goto (default,s,1)
What solutions are there to getting rid of the yyerror??
--
. . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kphone--asterisk--Kphone
- Original Message -
From:
Michael
George
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, May 12, 2005 5:22
PM
Subject: Re: [Asterisk-Users]
Kphone--asterisk--Kphone
On Tue, May 10, 2005 at 12:01:17PM +0530, Sudhananda
wrote: I am running asterisk on one linux PC and want to talk
through this server using Kphone installed on 2 different PC's. These
are the extra lines added to sip.conf and extensions.conf
respectively. sip.conf [jitha]
type=friend host=dynamic secret=jitha
context=sip dtmfmode=inband [sudhananda]
type=friend host=dynamic secret=sudhananda
context=sipThis is what I use for kphone and it works
fine:[kphone]type=friend
; either "friend" (peer+user), "peer" or
"user"host=dynamic
; we have a static but private IP addresscallerid="kphone"
25dtmfmode=inband
; either RFC2833 or INFO for the
BudgeTonecontext=internaldisallow=all
; need to disallow=all before we can use
allow=allow=ulaw
; Note: In user sections the order of codecs
extensions.conf [sip]
exten=1,1,Dial(SIP/jitha,20,tr)
exten=2,1,Dial(SIP/sudhananda,20,tr) Both the Kphones got
registered to the asterisk but when i dial the number it gives me the
following log on asterisk Asterisk Ready.
*CLI
-- Registered SIP 'sudhananda' at 172.16.2.35 port 5060 expires
900 -- Executing
Dial("SIP/sudhananda-aa77", "SIP/jitha|20|tr") in new
stack -- Called
jitha -- SIP/jitha-f4bc is
ringing -- SIP/jitha-f4bc answered
SIP/sudhananda-aa77 -- Attempting
native bridge of SIP/sudhananda-aa77 and SIP/jitha-f4bcI see no
problems here yet. and one Kphone status is ringing and on other
it is connected. how to solve this problem.You might want to
check the codecs in use. Are they both on the
localnetwork?
I am using G.711 ulaw codec. yeah both are in the
same network.-- -MThere are 10 kinds of people in this
world:Those who can count in binary and those who
cannot.___Asterisk-Users
mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo
UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: failure notice
Can we get this guy booted off the list somehow? -Original Message- From: [EMAIL PROTECTED] [mailto:MAILER- [EMAIL PROTECTED] Sent: Thursday, 12 May 2005 8:36 AM To: Dean Collins Subject: failure notice Hi. This is the qmail-send program at smtp.register.it. I'm afraid I wasn't able to deliver your message to the following addresses. This is a permanent error; I've given up. Sorry it didn't work out. [EMAIL PROTECTED]: This message is looping: it already has my Delivered-To line. (#5.4.6) --- Below this line is a copy of the message. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX.CC/SixTel
Same thing happend to me. I order a 954-XXX-XXX DID on 04-06-2005 and I'm still waiting. My order status also says pending. On 5/11/05, BJ Weschke [EMAIL PROTECTED] wrote: I ordered a 973-XXX- and 585-XXX- DID from them on 2/3 and 2/7 of this year respectively. Their customer service portal still lists these orders as pending though they told me back when I ordered them that provisioning would happen within 1 business day. On 5/11/05, Wiley Siler [EMAIL PROTECTED] wrote: Anyone have an opinion about these guys and their recent performance? I need some local DIDs and they provide for my area code. Thanks, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi and chan_misdn
Could someone please comment on the current state of chan_capi, chan_misdn and chan_modem channel drivers in terms of functionality and stability. Specifically, which channel driver would be best for a passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that chan_capi distinguishes between junghanns and non-junghans cards, and that chan_misdn is better suited for general misdn compatibility. A second point I'd like some clarification on is the purpose of Junghann's BRIStuff patch. Is this patch only necessary for chan_capi or also for chan_misdn? Does this patch add functionality to asterisk or is it only intended to smooth chan_capi integration into asterisk? Thanks in advance! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP : chan_mgcp.c:1509 find_subchannel
I solved the problem by rechecking my configuration files, namely mgcp.conf and
extensions.conf.
I changed the EPIDx strings in the ATA188 to a001 and a002 (and changed
accordingly in other config
files), the context from default to ext_mgcp in mgcp.conf and set all the ports
to 2427 and now it
works.
Hope this helps.
JFD
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
Behalf Of [EMAIL PROTECTED]
Sent: 10 mai 2005 12:07
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] MGCP : chan_mgcp.c:1509 find_subchannel
Nevermind, I have solved the problem.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
Behalf Of [EMAIL PROTECTED]
Sent: 10 mai 2005 10:33
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] MGCP : chan_mgcp.c:1509 find_subchannel
Importance: High
When I try to connect to * using a Cisco ATA 188 configured with a MGPC
firmware (v3.1.1), I just
keep getting this message every 30 seconds or so :
May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway
'192.168.1.27' (and thus its
endpoint '*') does not exist
Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port
2427) sends UDP packets
to * (IP 192.168.1.59, port 2727). On the other hand, after sending 2 packets
at startup, * does not
respond to the ATA188.
This certainly looks like a configuration problem, but I just can't seem to
find exactly what is
wrong. If someone has experienced the same problem or know what is wrong then I
would really
appreciate your help.
Thanks,
JF
Modules.conf :
[modules]
autoload=yes
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
noload = chan_modem.so
noload = chan_modem_aopen.so
noload = chan_modem_bestdata.so
noload = chan_modem_i4l.so
;noload = res_musiconhold.so
noload = app_festival.so
noload = app_url.so
noload = app_image.so
noload = app_disa.so
noload = app_qcall.so
noload = app_adsiprog.so
noload = app_ices.so
;noload = codec_g726.so
;noload = codec_alaw.so
;noload = format_vox.so
;noload = format_h263.so
noload = format_jpeg.so
;noload = cdr_csv.so
;noload = cdr_manager.so
;noload = app_zapras.so
;noload = app_flash.so
;noload = app_zapbarge.so
;noload = app_zapscan.so
;noload = app_talkdetect.so
;noload = app_alarmreceiver.so
;noload = chan_skinny.so
;noload = chan_sip.so
noload = chan_alsa.so
;noload = chan_oss.so
[global]
chan_modem.so=yes
Mgcp.conf :
[general]
port = 2727
bindaddr = 192.168.1.59
[MGCP1]
context=default
host=192.168.1.27
line=aaln/1
line=aaln/2
[MGCP2]
context=default
host=192.168.1.28
line=aaln/1
line=aaln/2
Extensions.conf :
[general]
static = yes
writeprotect = no
[globals]
TMGCP1=MGCP/aaln/[EMAIL PROTECTED]
TMGCP2=MGCP/aaln/[EMAIL PROTECTED]
TMGCP3=MGCP/aaln/[EMAIL PROTECTED]
TMGCP4=MGCP/aaln/[EMAIL PROTECTED]
TSIP1=SIP/SIP1
TSIP2=SIP/SIP2
[default]
exten = 70,1,Dial(${TMGCP1},20,tr)
exten = 71,1,Dial(${TMGCP2},20,tr)
exten = 72,1,Dial(${TMGCP3},20,tr)
exten = 73,1,Dial(${TMGCP4},20,tr)
exten = 74,1,Dial(${TSIP1},20,tr)
exten = 75,1,Dial(${TSIP2},20,tr)
ATA188 config :
Cisco ATA 188 (MGCP) Configuration :
UIPassword: *
UseTftp:1
TftpURL:0
CfgInterval: 3600
EncryptKey: *
EncryptKeyEx:
Dhcp: 0
StaticIP: 192.168.1.27
StaticRoute: 192.168.1.1
StaticNetMask: 255.255.255.0
EPID0orSID0: .
EPID1orSID1: .
CA0orCM0: 192.168.1.59:2727
CA1orCM1: 0
CA0UID: 0
CA1UID: 0
MGCPVer: NCS1.0
RetxIntvl: 500
RetxLim: 10
MGCPPort: 2427
CodecName: PCMU,PCMA,G723,G729
LBRCodec: 3
PrfCodec: 1
AudioMode: 0x00350035
ConnectMode: 0x9400
CallerIdMethod: 0xc0019e60
DNS1IP: 0.0.0.0
DNS2IP: 0.0.0.0
Domain: .
NumTxFrames: 2
TOS: 0x68b8
OpFlags: 0x0002
VLANSetting: 0x002b
Polarity: 0x
FXSInputLevel: 0
FXSOutputLevel: -4
SigTimer: 0x0064
RingCadence: 2,4,25
DialTone: 2,31538,30831,1380,1740,1,0,0,1000,0,0
BusyTone: 2,30467,28959,1191,1513,0,4000,4000,0,0,0
ReorderTone: 2,30467,28959,1191,1513,0,2000,2000,0,0,0,0,0,0,0,0,0
RingBackTone: 2,30831,30467,1943,2111,0,16000,32000,0,0,0
CallWaitTone: 1,30831,0,5493,0,0,2400,2400,4800,0,0
AlertTone: 1,30467,0,5970,0,0,480,480,1920,0,0
NPrintf: 0.0.0.0.0
TraceFlags: 0x
SyslogIP: 0.0.0.0.514
SyslogCtrl: 0x
MediaPort: 16384
CFGID: 0x
ata0013199e70f5
Version: v3.1.1 atamgcp (Build 040629A)
Features: 0x0017
HardwareVersion: 0x0010 0x
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users
Re: [Asterisk-Users] VoiceBlue GSM
2-) You can buy a voip2GSM Gateway. To which you no longer need hardware, you just register the voip2GSM devise to Asterisk and then it is ready to receive and send calls just like any other sip phone. Cost of this is around 400 USD / UNIT That is interesting. What is the make and the model that you are referring to? Is there a website with more info? I currently use Quescom IP400 GSM but they are expensive (although they support up to 12 GSM channels). -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone ever implement an *outbound* dial-by-name??
Also off the top of my head.. How about: specify a context in voicemail.conf: [outward-dial-by-name] 2125551212 = 1000,John Smith 301212 = 1000,George Lucas or if you use 9 to dial out: [outward-dial-by-name] 92125551212 = 1000,John Smith 9301212 = 1000,George Lucas Again, I have not tried this. On 5/12/05, El Flynn [EMAIL PROTECTED] wrote: Michael Jones wrote: Hi All; I'm a newbie so please be gentle. I'm a new * user and am using it to control the 3 IP phones in my house. I'm using asterisk because I enjoy the flexibility and I'm sort of a tinkerer. Here's my question: Everyone has used the dial by directory function where you dial the user's name to connect to that extension. Instead of an inward dial, I'm thinking how cool it'd be to have an outward dial-by-name, where from any extension you can spell a name and dial it outbound via a trunk line. Off the top of my head.. specify a context in voicemail.conf: [outward-dial-by-name] 1000 = 1000,John Smith 1001 = 1000,George Lucas then another context in extensions.conf [outward-dial-by-name] 1000 = Zap/g1/5551234567 ; john smith's phone number 1001 = Zap/g1/555123 ; george's mobile phone and finally in your dialplan (assuming you use a context called internal for all your internal phones..) [internal] ; some other stuff exten = 123,1,Directory(outward-dial-by-name|outward-dial-by-name) disclaimer: untested stuff. your mileage may vary. don't sue me if it don't work Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Open Source MGCP Softphone
Has anyone heard of a working Open Source Softphone compatible with the MGCP protocol ? Right now, I know of the eyeP softphone, but it is not Open Source. Thanks for any help. JFD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with VIA Chipset
On May 11, 2005 05:15 pm, Armin Lediger wrote: I am trying to install asterisk 1.0.7 on a VIA EPIA 5000 board - anyone of you already managed to do so? I got V1.0.6 running, but 1.0.7 seems not to compile. Just a correction; this isn't about a VIA chipset; this is about a VIA processor. There are many, many Asterisk installs with Intel or AMD processors and VIA chipsets that work just fine. It looks like others have given you the 'fix' -- I just wanted to make sure that the archives showed that this is a processor problem, not a chipset problem. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime voicemail login incorrect
it seems you are right, you need the context there, but than I cannot use it in Realtime anymore, since I have more than one context Why would having more than one context stop you fomr using RealTime? Doesn't stop us. I would than need for each context an extra extension number. It makes no sense either, since one phone number should have anyway only ONE context, or could be there a case that one could have more than one context? Uh..no..I've got about 5 different instances of the extension 3113. They are all in different contexts. Contexts (for us at least) are broken into customers. Each context has specific extensions for their company. All have their own VoiceMailMain entry. Pretty easy. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inbound ANI DNIS format
Hello, Being totally fed up with the lack of quality and reliability from both VoicePulse and BroadVoice, We are switching to a direct IP connection to Global Crossing. We've installed a local point-to-point T1 into their CO, and they will give/take SIP g729a directly and act as the gateway for us. In setting up the inbound SIP service, they are asking the question, In what format do I want my ANI DNIS presented? They provided examples, such as *ANI*DNIS, etc. Does anyone out there know how Asterisk expects to see this information on inbound calls? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ast_yyerror - 'space' in Caller-ID - stringcomparison
I also get this when doing a Manager Click2Dial application except the ^^ in
the error go on a few thousand times. The call still completes but you still
get the error.
-Matthew
Mark Elkins wrote:
I've some code to manipulate incoming Caller-ID - so its suitable for
replying to...
[sipdef]
exten = s,1,NoOp(FWD SIP: ${CALLERIDNAME} ${CALLERIDNUM})
; Alter incoming calles from pulver - add a '87'
exten = s,2,Gotoif($[${CALLERIDNAME} = ${CALLERIDNUM}]?3:4)
exten = s,3,SetCIDName(87${CALLERIDNUM})
exten = s,4,SetCIDNum(87${CALLERIDNUM})
exten = s,5,Goto(default,s,1)
When Executing the above - and I presume incoming Caller Info looks
like the name is Mark Elkins and the Number is 638936...
The purpose is to prefix the number (only the number) with 87.
Sometimes, incoming CallerID data looks like -- 638936 638936
therefore the checking of both Name and number.
-- Executing NoOp(SIP/292951-b11f, FWD SIP: Mark Elkins
638936) in new stack
May 12 14:36:59 WARNING[28824]: ast_expr.y:486 ast_yyerror:
ast_yyerror(): syntax error: parse error; Input:
Mark Elkins = 638936
^
-- Executing GotoIf(SIP/292951-b11f, Mark?3:4) in new stack
-- Goto (sipdef,s,4)
-- Executing SetCIDNum(SIP/292951-b11f, 87638936) in new stack
-- Executing Goto(SIP/292951-b11f, default|s|1) in new stack
-- Goto (default,s,1)
What solutions are there to getting rid of the yyerror??
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming calls picked-up then simply hanged-up
I don't think my first posting went thru. I am trying to set up Asterisk for the first time. I am new to this. I am using [EMAIL PROTECTED] I have a TDM400P with one FXO and one FXS The system is working for outgoing calls and if I test incoming calls using . But when doing an actual call the system seems to answer the call and then immediately hang up. I made a small test following some instructions and made changes to the from-pstn context to look like this: [from-pstn] exten = s,1,Answer() exten = s,2,Wait(4) exten = s,3,Playback(goodbye) exten = s,4,Hangup() The incoming calls are set up to go from the PSTN to the Digital Receptionist. But I get the same behavior if I have incoming call send to the extension I have set up. Has anyone else seen this behavior? Any ideas as to what I should try? Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cellsocket problem
This is what I getafter Zap/4-1 answer I can press # and
the call go thru just fine..I just can find a way to force the # go in
automaticly @ end... :-( any ideas?
===Connected to
Asterisk 1.0.7 currently running on pbx (pid = 1089)
Verbosity is at least 3
-- Remote UNIX connection
-- Executing Macro(SIP/2007-a956,
dialout-trunk|4|2831234) in new stack
-- Executing GotoIf(SIP/2007-a956, 0?4) in new stack
-- Executing SetCallerID(SIP/2007-a956, 2007) in new
stack
-- Executing Goto(SIP/2007-a956, 6) in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing SetGroup(SIP/2007-a956, OUT_4) in new
stack
-- Executing CheckGroup(SIP/2007-a956, 1) in new
stack
-- Executing SetVar(SIP/2007-a956,
DIAL_NUMBER=2831234) in new stack
-- Executing SetVar(SIP/2007-a956, DIAL_TRUNK=4) in
new stack
-- Executing AGI(SIP/2007-a956, fixlocalprefix) in
new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse
/etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Dial(SIP/2007-a956, ZAP/4/2831234) in
new stack
-- Called 4/2831234
-- Zap/4-1 answered SIP/2007-a956
-- Hungup 'Zap/4-1'
== Spawn extension (macro-dialout-trunk, s, 11) exited
non-zero on 'SIP/2007-a956' in macro 'dialout-trunk'
== Spawn extension (from-internal, 2831234, 1) exited
non-zero on 'SIP/2007-a956'
-- Executing Macro(SIP/2007-a956, hangupcall) in new
stack
-- Executing ResetCDR(SIP/2007-a956, w) in new stack
-- Executing NoCDR(SIP/2007-a956, ) in new stack
-- Executing Wait(SIP/2007-a956, 5) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited
non-zero on 'SIP/2007-a956' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero
on 'SIP/2007-a956'
pbx*CLI
[cellsocket-custom]
exten = _NXX,1,Dial(Zap/4/${EXTEN}#)
exten = _NXX,2,Macro(outisbusy) ; No available
circuits
also tried
[cellsocket-custom]
exten = _NXX,1,Dial(Zap/4/w${EXTEN}#)
exten = _NXX,2,Macro(outisbusy) ; No available
circuits
also tried
[cellsocket-custom]
SHARP=#
exten = _NXX,1,Dial(Zap/4/w${EXTEN}${SHARP})
exten = _NXX,2,Macro(outisbusy) ; No available
circuits
attachment: winmail.dat___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and FastStart
Hi all, When I enabled faststart in oh323.conf, calls from H323 endpoint to SIP phones could not complete. The originating phone kept ringing when calls were answered by SIP phones. fastStart=yes h245Tunnelling =yes h245inSetup=yes Please can you advise. Many Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming calls picked-up then simply hanged-up
You should put your asterisk into verbose mode using asterisk -c or if you are using a server asterisk -r and you can trace out what happens and it will be in the log file called full in the /var/log/asterisk directory and then you can probably figure out what happened. Your incoming call context must eventually dial an extension, but I am sure you will see what is going on if you debug like that. on Thursday 05/12/2005 fhunter([EMAIL PROTECTED]) wrote I don't think my first posting went thru. I am trying to set up Asterisk for the first time. I am new to this. I am using [EMAIL PROTECTED] I have a TDM400P with one FXO and one FXS The system is working for outgoing calls and if I test incoming calls using . But when doing an actual call the system seems to answer the call and then immediately hang up. I made a small test following some instructions and made changes to the from-pstn context to look like this: [from-pstn] exten = s,1,Answer() exten = s,2,Wait(4) exten = s,3,Playback(goodbye) exten = s,4,Hangup() The incoming calls are set up to go from the PSTN to the Digital Receptionist. But I get the same behavior if I have incoming call send to the extension I have set up. Has anyone else seen this behavior? Any ideas as to what I should try? Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Broadvoice is finally starting to give answers
trixter http://www.0xdecafbad.com wrote: They paid 100% of the *UNDISPUTED* charges but nothing is said about the disputed ones. Typo or intentional? It also sounds to me like its an access charge issue, but I may be reading too much into this. Sounds like BroadVoice paid their bill according to their interpretation of their contract with the carrier. And that the carrier interpreted the contract differently and billed them a significantly larger amount (thus the use of the word undisputed). It also sounds like this dispute went on for quite a while. The carrier finally pulled the plug. It also sounded like the carrier is initiating a lawsuit against BroadVoice. So to sum up, it seems like a basic contract dispute. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with MeetMe
It sounds like you don't have USB support compiled in the kernel. Chris - Original Message - From: Daniel Salama [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2005 11:55 PM Subject: Re: [Asterisk-Users] Problem with MeetMe Chris/BJ, I am running REL3 with kernel 2.4.21-27.0.4.ELsmp. I enabled USB devices in the BIOS. Here are the problems I'm seeing: [EMAIL PROTECTED]: ~ modprobe zaptel [EMAIL PROTECTED]: ~ modprobe ztdummy /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb- uhci.o failed /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod ztdummy failed [EMAIL PROTECTED]: ~ lsmod Module Size Used byNot tainted zaptel183104 0 (unused) soundcore 7044 0 (autoclean) iptable_filter 2412 0 (autoclean) (unused) ip_tables 16544 1 [iptable_filter] e1000 77884 2 floppy 57552 0 (autoclean) sg 37388 0 (autoclean) usbcore81152 1 ext3 89992 2 jbd55156 2 [ext3] 3w-9xxx 570016 3 sd_mod 13936 6 scsi_mod 115240 2 [sg 3w-9xxx sd_mod] [EMAIL PROTECTED]: ~ modprobe -r zaptel [EMAIL PROTECTED]: ~ modprobe usb-uhci /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb- uhci.o failed /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod usb-uhci failed [EMAIL PROTECTED]: ~ modprobe usb-ohci [EMAIL PROTECTED]: ~ modprobe zaptel [EMAIL PROTECTED]: ~ modprobe ztdummy /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb- uhci.o failed /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod ztdummy failed [EMAIL PROTECTED]: ~ lsmod Module Size Used byNot tainted zaptel183104 0 (unused) usb-ohci 23176 0 (unused) soundcore 7044 0 (autoclean) iptable_filter 2412 0 (autoclean) (unused) ip_tables 16544 1 [iptable_filter] e1000 77884 2 floppy 57552 0 (autoclean) sg 37388 0 (autoclean) usbcore81152 1 [usb-ohci] ext3 89992 2 jbd55156 2 [ext3] 3w-9xxx 570016 3 sd_mod 13936 6 scsi_mod 115240 2 [sg 3w-9xxx sd_mod] It still won't load ztdummy. I can't get usb-uhci to work. I read on the wiki that ztdummy requires uhci. What's the difference between ohci and uhci? Thanks, Daniel On May 11, 2005, at 9:16 PM, Chris wrote: I forgot because I haven't moved to a 2.6 kernel. Chris - Original Message - From: BJ Weschke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2005 6:56 PM Subject: Re: [Asterisk-Users] Problem with MeetMe Only if you're on a 2.4 kernel. A 2.6 kernel doesn't require USB for it's timing source. On 5/11/05, Chris [EMAIL PROTECTED] wrote: Edit the Makefile for the zaptel drivers. You will see two commented lines that say ztdummy. Uncomment them and rebuild. Once you install the rebuild, do a modprobe ztdummy and you should be good to go. BTW, you do need an active USB for ztdummy to load. Chris - Original Message - From: Daniel Salama [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2005 3:28 PM Subject: Re: [Asterisk-Users] Problem with MeetMe I don't have any of
[Asterisk-Users] chan_capi, chan_misdn and chan_modem
Could someone please comment on the current state of chan_capi, chan_misdn and chan_modem channel drivers in terms of functionality and stability. Specifically, which channel driver would be best for a passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that chan_capi distinguishes between junghanns and non-junghans cards, and that chan_misdn is better suited for general misdn compatibility. A second point I'd like some clarification on is the purpose of Junghann's BRIStuff patch. Is this patch only necessary for chan_capi or also for chan_misdn? Does this patch add functionality to asterisk or is it only intended to smooth chan_capi integration into asterisk? Thanks in advance! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceBlue GSM
On Thu, 2005-05-12 at 17:31 +0400, Jean-Michel Hiver wrote: 2-) You can buy a voip2GSM Gateway. To which you no longer need hardware, you just register the voip2GSM devise to Asterisk and then it is ready to receive and send calls just like any other sip phone. Cost of this is around 400 USD / UNIT That is interesting. What is the make and the model that you are referring to? Is there a website with more info? I currently use Quescom IP400 GSM but they are expensive (although they support up to 12 GSM channels). Can these GSM modules work as proxies so when you are local your GSM handset can be used on the VoIP network? When you walk away from the nanocell/picocell transmitter it autoswitches to the real provider. The way GSM auth works this technically would not be that difficult to do, and infact there is equipment that does not interface to anything that does proxy call setup/tear down info (IMSI catchers for example), so what I am asking for is not that far out. This would also make it easier for call shops, set up one of these and people can use their mobile as a voip phone. In the office your GSM calls are sent to asterisk to your desk phone and/or your mobile, outbound calls go over asterisk for least cost routing, etc. But the handset is your mobile (and in theory on your person at all times). If anyone knows of a device that integrates to asterisk that does that I would *greatly* appreciate hearing from you regarding a vendor, make/model, even a supplier. If you are a supplier I grant you permission to use my contact info to directly contact me about that issue only so long as you dont add me to any lists. Thanks -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Inbound ANI DNIS format
I believe *ANI*DNIS That's how Asterisk sends it when I set my t1 line to featd. In /etc/asterisk/zapata.conf signalling=featd not much to go on, but a little! -Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: Thursday, May 12, 2005 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Inbound ANI DNIS format Hello, Being totally fed up with the lack of quality and reliability from both VoicePulse and BroadVoice, We are switching to a direct IP connection to Global Crossing. We've installed a local point-to-point T1 into their CO, and they will give/take SIP g729a directly and act as the gateway for us. In setting up the inbound SIP service, they are asking the question, In what format do I want my ANI DNIS presented? They provided examples, such as *ANI*DNIS, etc. Does anyone out there know how Asterisk expects to see this information on inbound calls? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incoming calls picked-up then simply hanged-up
Thanks I will give that a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Thursday, May 12, 2005 9:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Incoming calls picked-up then simply hanged-up You should put your asterisk into verbose mode using asterisk -c or if you are using a server asterisk -r and you can trace out what happens and it will be in the log file called full in the /var/log/asterisk directory and then you can probably figure out what happened. Your incoming call context must eventually dial an extension, but I am sure you will see what is going on if you debug like that. on Thursday 05/12/2005 fhunter([EMAIL PROTECTED]) wrote I don't think my first posting went thru. I am trying to set up Asterisk for the first time. I am new to this. I am using [EMAIL PROTECTED] I have a TDM400P with one FXO and one FXS The system is working for outgoing calls and if I test incoming calls using . But when doing an actual call the system seems to answer the call and then immediately hang up. I made a small test following some instructions and made changes to the from-pstn context to look like this: [from-pstn] exten = s,1,Answer() exten = s,2,Wait(4) exten = s,3,Playback(goodbye) exten = s,4,Hangup() The incoming calls are set up to go from the PSTN to the Digital Receptionist. But I get the same behavior if I have incoming call send to the extension I have set up. Has anyone else seen this behavior? Any ideas as to what I should try? Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Broadvoice is finally starting to give answers
On Thu, 2005-05-12 at 09:08 -0500, Dan Perik wrote: trixter http://www.0xdecafbad.com wrote: They paid 100% of the *UNDISPUTED* charges but nothing is said about the disputed ones. Typo or intentional? It also sounds to me like its an access charge issue, but I may be reading too much into this. Sounds like BroadVoice paid their bill according to their interpretation of their contract with the carrier. And that the carrier interpreted the contract differently and billed them a significantly larger amount (thus the use of the word undisputed). It also sounds like this dispute went on for quite a while. The carrier finally pulled the plug. It also sounded like the carrier is initiating a lawsuit against BroadVoice. So to sum up, it seems like a basic contract dispute. Yes, I think its over access charges but do not know. Undisputed charges left unpaid during the dispute is not grounds to cancel service in america. Thus while the undisputed charges are paid broadvoice cannot claim that BV was behind on payments to terminate service. Not making it clear in the open letter makes me think that they are just hoping that people understand the law (ha!) and know that service cannot be terminated for this or something else. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] beginner in Asterisk configuration
Hello Sorry for english speaking peaple, but I just help this beginner in our natural language : French ;-) Je suis Français aussi, si tu as besoin d'un peu d'aide tu peux me joindre directement par mail Pour tester ta config : asterisk -gc Bonne chance -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Tutu Lord Envoyé : jeudi 12 mai 2005 09:58 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] beginner in Asterisk configuration hello, i am french student and i want configure a Asterisk server. when I want launch the server with the command safe_asterisk -vcf the server answer : Asterisk ended with exit status 1 Asterisk died with code 1 what is the signification of it please ? thank you lucas _ MSN Hotmail : antivirus et antispam gratuits http://www.imagine-msn.com/hotmail/default.aspx?locale=fr-FR ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ${BLINDTRANSFER} variable
I've found on wiki that there is a variable called ${BLINDTRANSFER}
which should contain the channel (or a number) of user that made a blind
transfer of
a call to another extension.
Also I've found a patch for chan_sip to add support for ${BLINDTRANSFER},
but it's not working at all (chan_sip crashing), so I guess it is intended
for CVS-HEAD version.
Has anyone tried to backport it to STABLE (1.0.7 preferably :) ).
Thanks,
Ivan
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with MeetMe
What I have discovered is that my motherboard only supports usb-ohci and not usb-uhci. Reading on the wiki, it says that ztdummy requires usb-uhci. To make things worse, I slapped in a TDM22B just to get timer support, only to discover that the machine kept crashing because of a hardware conflict with my RAID controller. Really weird! Anyway, my only three other options are: 1) Compile kernel 2.6, which I'd hate to do 2) Replace either the motherboard or the RAID controller, which is worse than option 1 3) Setup a separate machine where I can install the TDM22B and dedicate it just for MeetMe and may be a couple of other things. I may give this a shot. I still need to figure out how to do this, so if you guys can provide any sample configs I'd appreciate it. Any other suggestions you guys may have? Thanks, Daniel On May 12, 2005, at 10:08 AM, Chris wrote: It sounds like you don't have USB support compiled in the kernel. Chris - Original Message - From: Daniel Salama [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2005 11:55 PM Subject: Re: [Asterisk-Users] Problem with MeetMe Chris/BJ, I am running REL3 with kernel 2.4.21-27.0.4.ELsmp. I enabled USB devices in the BIOS. Here are the problems I'm seeing: [EMAIL PROTECTED]: ~ modprobe zaptel [EMAIL PROTECTED]: ~ modprobe ztdummy /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb- uhci.o failed /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod ztdummy failed [EMAIL PROTECTED]: ~ lsmod Module Size Used byNot tainted zaptel183104 0 (unused) soundcore 7044 0 (autoclean) iptable_filter 2412 0 (autoclean) (unused) ip_tables 16544 1 [iptable_filter] e1000 77884 2 floppy 57552 0 (autoclean) sg 37388 0 (autoclean) usbcore81152 1 ext3 89992 2 jbd55156 2 [ext3] 3w-9xxx 570016 3 sd_mod 13936 6 scsi_mod 115240 2 [sg 3w-9xxx sd_mod] [EMAIL PROTECTED]: ~ modprobe -r zaptel [EMAIL PROTECTED]: ~ modprobe usb-uhci /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb- uhci.o failed /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod usb-uhci failed [EMAIL PROTECTED]: ~ modprobe usb-ohci [EMAIL PROTECTED]: ~ modprobe zaptel [EMAIL PROTECTED]: ~ modprobe ztdummy /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb- uhci.o failed /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod ztdummy failed [EMAIL PROTECTED]: ~ lsmod Module Size Used byNot tainted zaptel183104 0 (unused) usb-ohci 23176 0 (unused) soundcore 7044 0 (autoclean) iptable_filter 2412 0 (autoclean) (unused) ip_tables 16544 1 [iptable_filter] e1000 77884 2 floppy 57552 0 (autoclean) sg 37388 0 (autoclean) usbcore81152 1 [usb-ohci] ext3 89992 2 jbd55156 2 [ext3] 3w-9xxx 570016 3 sd_mod 13936 6 scsi_mod 115240 2 [sg 3w-9xxx sd_mod] It still won't load ztdummy. I can't get usb-uhci to work. I read on the wiki that ztdummy requires uhci. What's the difference between ohci and uhci? Thanks, Daniel On May 11, 2005, at 9:16 PM, Chris wrote: I forgot because I haven't moved to a 2.6 kernel. Chris - Original Message - From: BJ Weschke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2005 6:56 PM Subject: Re: [Asterisk-Users] Problem with
Re: [Asterisk-Users] VoiceBlue GSM
Hi, That is interesting. What is the make and the model that you are referring to? Is there a website with more info? As for the models, we sell them as OEM. You may contact me offlist if interested. Better priced and more powerful than existing devises out there. I currently use Quescom IP400 GSM but they are expensive (although they support up to 12 GSM channels). Quescomm, are good but have drawback. 12 SIMs model you need to unscrew to change the SIMs. This is not very practical. I recommand other brands like Vierling, who are more practical in everyday life of a corporate grade carrier devise. Can these GSM modules work as proxies so when you are local your GSM handset can be used on the VoIP network? When you walk away from the nanocell/picocell transmitter it autoswitches to the real provider. The idea is brilliant. I haven't seen such product on market. Tellular have something similar, but you still need some workout. I have other solution for this functionality: In europe you can have two SIMs for same phone number. In Belgium for instance (proxi-Duo). Then, you plug one SIM in the GSM Gateway and another one in the GSM devise. When you get home you switch your GSM and automatically the other sIM becomes active, when you walk away you switch your GSM phone on and it will automatically become active. The way GSM auth works this technically would not be that difficult to do, and infact there is equipment that does not interface to anything that does proxy call setup/tear down info (IMSI catchers for example), so what I am asking for is not that far out. Some GSM network operators do this. Last SIM registered is one to be considered and others become inactive. This would also make it easier for call shops, set up one of these and people can use their mobile as a voip phone. Hmm. What is added value of customers using their cell phone in callshops? They have walked all the way long to reach the place. You may explain further In the office your GSM calls are sent to asterisk to your desk phone and/or your mobile, outbound calls go over asterisk for least cost routing, etc. But the handset is your mobile (and in theory on your person at all times). If anyone knows of a device that integrates to asterisk that does that I would *greatly* appreciate hearing from you regarding a vendor, make/model, even a supplier. If you are a supplier I grant you permission to use my contact info to directly contact me about that issue only so long as you dont add me to any lists. Other idea is that poeple Thanks -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: GXP 2000 Conference Button and ILBC
-Original Message- snip I just downloaded the recent firmware for GS GXP 2000 and I must say the phone works great but... How do you make the conf button work?? Anybody done that? ---I just put in a system with 25 of these and have the same issue. Looks like the conf button will come out in some later firmware. Also, the speakerphone has a nasty speaker-mic feedback loop. For a +-$120 phone seems to be a good UA. Also, with great dissapointment I must ask, where is ILBC support? GS web page mentions it and the manual says it supports it almost using bolds :) soo where is it ---I am using ulaw so I haven't played with any of the codecs that they have introduced. At a quick glance it really only looked like they added 729 to the ulaw/alaw that they had though. Jason Kawakami www.optellabs.com Salt Lake City, UT ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cellsocket with @home
I am posting this in case someone need
help.
=
YOU THA
MAN!
No sure how I will repay you, but
anything you need, just let me know!
Thank you, thank you, thank you
-- Executing GotoIf(SIP/2007-12c7,
0?4) in new stack
-- Executing SetCallerID(SIP/2007-12c7,
2007) in new stack
-- Executing Goto(SIP/2007-12c7,
6) in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing SetGroup(SIP/2007-12c7,
OUT_4) in new stack
-- Executing CheckGroup(SIP/2007-12c7,
1) in new stack
-- Executing SetVar(SIP/2007-12c7,
DIAL_NUMBER=2831234#) in new stack
-- Executing SetVar(SIP/2007-12c7,
DIAL_TRUNK=4) in new stack
-- Executing AGI(SIP/2007-12c7,
fixlocalprefix) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed,
returning 0
-- Executing Dial(SIP/2007-12c7,
ZAP/4/2831234#) in new stack
-- Called 4/2831234#
-- Zap/4-1 answered SIP/2007-12c7
-- Hungup 'Zap/4-1'
== Spawn extension (macro-dialout-trunk, s, 11)
exited non-zero on 'SIP/2007-12c7' in macro 'dialout-trunk'
== Spawn extension (from-internal, 2831234, 1)
exited non-zero on 'SIP/2007-12c7'
-- Executing Macro(SIP/2007-12c7,
hangupcall) in new stack
-- Executing ResetCDR(SIP/2007-12c7,
w) in new stack
-- Executing NoCDR(SIP/2007-12c7,
) in new stack
-- Executing Wait(SIP/2007-12c7,
5) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited
non-zero on 'SIP/2007-12c7' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited
non-zero on 'SIP/2007-12c7'
pbx*CLI
-Original Message-
From: Tha Man!!
Sent: Thursday, May
12, 2005 8:34 AM
To: Manny Wise
Subject: Re: [Asterisk-Users] Cellsocket
help needed
There is no real easy way (that I
know of) to add the # at the end without getting your hands a bit
dirty.
Go into AMP and click on
Maintenance.
Then click on phpMyAdmin.
That will open a new window with the
phpMyAdmin mysql interface.
On the left, click on the dropdown
box and select asterisk
Under the box it will show the
structure of the DB.
Click on extensions
Now on the right near the top, click
on Browse
This is where AMP keeps all your
extension info.
This will be the hardest part
because you are going to have to do the identification.
Typically, the dial commands are
kept near the bottom and start with outrt
You're going to want to find the
name of the outbound route that will be using the cellsocket.
Look for the entries for that route
that contain dialout-trunk, those will be the ones you want to
edit.
Click on the little pencil icon for
that line.
Go to the dialout-trunk
line and add a # on to the end
example:
dialout-trunk,1,${EXTEN:}#
Then click on go at the bottom
to say the changes.
This will save the change and take
you back to the previous list.
Make sure you have all the dialout-trunk's
in the list for that outrt modified.
Once done, close the phpMyAdmin
window.
Now go back to AMP. We need to force
it to regen its configs.
Go under setup.
Go under extensions.
Click on any extension on the right.
Don't change anything and just click
on Submit Changes
This will pop up the red bar at the
top.
Click on it to apply changes.
After all this, you should be good to
go.
Good luck!!
- Original Message -
From: Manny Wise
To: The Best!!!
Sent: Wednesday, May
11, 2005 6:29 PM
Subject: Re:
[Asterisk-Users] Cellsocket help needed
I have made some progress, but still doesn't workI
created the trunk, * is dialing the ttrunk... I see from telenet asterisk -r
that the trunk pass the call to the cellsocketI see that it says Zap/4-1
answered.but to make it work I have to press the # in the cisco phone, then
I see the cellphone dialing, I followed the instruction from the other post
and have tried several convinations and still don't put the # at end, Do you
have some samples?? thanks again!!
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gnugk
Hi
I've a problem with a gnugkv2.0.7
I've compiled gnugk successfully
I've installed PWlib-1.6.6 and openh323-1.13.5 libraries successfully
When i run gnugk i have this error:
error while loading shared libraries liboh323_linux_x86_r.so.1.13.5 cannot
open shared object file No such file or directory
I try to use command export:
export LD_LIBRARY_PATH=${HOME}/openh323/lib:${HOME}/pwlib/lib
in directory where i have this libraries
Have you suggestions?
Thanks
Ale
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice mail - Extension at vs Phone Number OGM
The good thing about gsm files and the fact that they are headerless is that you can simply cat files together. You just need to find the right sound files to do so. Then program your dialplan to play the message before sending the person to voicemail. I would zero out the unavailable and busy messages in the voicemail directory as you are recreating them. Then use the s option when sending the person to VM. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris CoulthurstSent: Thursday, May 12, 2005 5:16 AMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] Voice mail - "Extension at" vs "Phone Number" OGM Is there a way to make an outside call hear The person at phone number is unavail, but when an internal extension calls another extension, they hear The person at extension number is unavail? I swear Ive read this somewhere before but Im not typing in the right search. I probably found it before by complete accident Of course, we want the outside caller to hear a phone number seven digits long, while an extension hears just that, an extension. Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] switch in extensions.conf
Can anyone provide more information on switch or point me to where I can find more about it? The only I've been able to find on the wiki is: http://www.voip-info.org/tiki-index.php?page=Asterisk+-+dual+servers and towards the bottom of (section Forwarding to another Asterisk): http://www.voip-info.org/wiki-Asterisk+config+extensions.conf Some of the questions I have are: 1) If I have an asterisk machine being used only as a VoIP gateway to the PSTN, and I have multiple asterisk machines behind it waiting to make or receive calls through/from the gateway, can I have multiple switch statements in the same context, so that if the gateway tries to contact asterisk machine 1 and it's not available, try the next one and so on? 2) How to configure the other asterisk machines to use the VoIP gateway for all outbound calls? I read somewhere that you cannot have circular references using switch, but I'm not sure if it refers to what I'd like to do 3) Assuming the VoIP gateway cannot contact any of the other asterisk machines, can the voip gateway put calls on hold and continue trying and only disconnect or play busy tone after some pre-defined period of time? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incoming calls picked-up then simply hanged-up
This is what I got:
May 12 11:12:53 VERBOSE[1376]: -- Starting simple switch on 'Zap/4-1'
(Note that the line went dead on the calling phone before this next stuff
ever appeared)
May 12 11:13:01 WARNING[1376]: CallerID returned with error on channel
'Zap/4-1'
May 12 11:13:01 VERBOSE[1376]: -- Executing
[1;36;40mAnswer[0;37;40m([1;35;40mZap/4-1[0;37;40m, [1;35;40m[0;37;40m)
in new stack
May 12 11:13:01 DEBUG[1376]: Took Zap/4-1 off hook
May 12 11:13:01 DEBUG[1376]: Enabled echo cancellation on channel 4
May 12 11:13:01 DEBUG[1376]: Engaged echo training on channel 4
May 12 11:13:01 VERBOSE[1376]: -- Executing
[1;36;40mWait[0;37;40m([1;35;40mZap/4-1[0;37;40m, [1;35;40m5[0;37;40m)
in new stack
May 12 11:13:06 VERBOSE[1376]: -- Executing
[1;36;40mPlayback[0;37;40m([1;35;40mZap/4-1[0;37;40m,
[1;35;40mgoodbye[0;37;40m) in new stack
May 12 11:13:06 DEBUG[1376]: Scheduling timer at 160 sample intervals
May 12 11:13:06 VERBOSE[1376]: -- Playing 'goodbye' (language 'en')
May 12 11:13:07 DEBUG[1376]: Scheduling timer at 0 sample intervals
May 12 11:13:07 DEBUG[1376]: Scheduling timer at 0 sample intervals
May 12 11:13:07 VERBOSE[1376]: -- Executing
[1;36;40mHangup[0;37;40m([1;35;40mZap/4-1[0;37;40m,
[1;35;40m[0;37;40m) in new stack
May 12 11:13:07 VERBOSE[1376]: == Spawn extension (from-pstn, s, 4) exited
non-zero on 'Zap/4-1'
May 12 11:13:07 DEBUG[1376]: cdr_mysql: inserting a CDR record.
May 12 11:13:07 DEBUG[1376]: cdr_mysql: SQL command as follows: INSERT INTO
cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration
,billsec,disposition,amaflags,accountcode) VALUES ('2005-05-12
11:13:01','\Francine Walker\ 201','201','s','from-pstn',
'Zap/4-1','','Hangup','',6,6,'ANSWERED',3,'')
May 12 11:13:07 DEBUG[1376]: Hangup: channel: 4 index = 0, normal = 18,
callwait = -1, thirdcall = -1
May 12 11:13:07 DEBUG[1376]: disabled echo cancellation on channel 4
May 12 11:13:07 DEBUG[1376]: Set option TDD MODE, value: OFF(0) on Zap/4-1
May 12 11:13:07 DEBUG[1376]: Updated conferencing on 4, with 0 conference
users
May 12 11:13:07 VERBOSE[1376]: -- Hungup 'Zap/4-1'
May 12 11:13:26 DEBUG[1376]: Manager received command 'Command'
May 12 11:13:26 DEBUG[1376]: Manager received command 'Command'
May 12 11:15:26 DEBUG[1376]: Manager received command 'Command'
May 12 11:15:26 DEBUG[1376]: Manager received command 'Command'
May 12 11:17:26 DEBUG[1376]: Manager received command 'Command'
May 12 11:17:26 DEBUG[1376]: Manager received command 'Command'
May 12 11:17:26 DEBUG[1376]: Manager received command 'Command'
Any ideas, I don't see any errors reported. Just that callerID warning.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Thursday, May 12, 2005 9:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Incoming calls picked-up then simply hanged-up
You should put your asterisk into verbose mode using asterisk -c or if
you are using a server asterisk -r and you can trace out what happens
and it will be in the log file called full in the /var/log/asterisk
directory and then you can probably figure out what happened. Your incoming
call context must eventually dial an extension, but I am sure you will see
what is going on if you debug like that.
on Thursday 05/12/2005 fhunter([EMAIL PROTECTED]) wrote I don't
think my first posting went thru.
I am trying to set up Asterisk for the first time. I am new to this. I
am using [EMAIL PROTECTED] I have a TDM400P with one FXO and one FXS
The system is working for outgoing calls and if I test incoming calls
using . But when doing an actual call the system seems to answer
the call and then immediately hang up.
I made a small test following some instructions and made changes to the
from-pstn context to look like this:
[from-pstn]
exten = s,1,Answer()
exten = s,2,Wait(4)
exten = s,3,Playback(goodbye)
exten = s,4,Hangup()
The incoming calls are set up to go from the PSTN to the Digital
Receptionist. But I get the same behavior if I have incoming call send to
the extension I have set up.
Has anyone else seen this behavior? Any ideas as to what I should try?
Thanks in advance.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Your life is like a penny. You're going to lose it. The question is: How
do you spend it?
John Covici
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice Recognition - Cases of success
Hi Folks, I am planning to make a little project of voice recognition. I already browsed Voip Wiki and found some solutions. Before putting my hands on it to just do a little demo menu, I would like to hear from the list any succesful case using voice recognition and Asterisk. Best Regards, Isamar Maia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM gateway for Asterisk
Folks! I am looking at a couple of models of Fixed GSM Gateways for the Purpose of VOIP connectivity and specifically to work with Asterisk. I found that these can be imported into USA for about $99.99 or about that. This is a one channel unit just like tellular, one of them has GPRS. FCT11M: 1)freq: GSM network,900/1800/1900Mhz, 2)provides reversal signal for payphone/billing 3)supports PBX and VOIP 4)for voice (no fax) 5)battery(optional) 6)can be used for the remote area where signal is weak. FCT11G: 1)freq: GSM network,900/1800/1900Mhz, 2)provides reversal signal for payphone/billing 3)supports PBX and VOIP 4)for voice and GPRS (no fax) 5)battery(optional) 6)can be used for the remote area where signal is weak. I am pasting an image of the network diagram here: Specifications in text are below. I would appreciate for any feedback of their usability. Seshu Description: --- This unit can conveniently access to the available GSM system network. This system possesses such a high receiving sensitivity and a large transmitting power that it expandsthe effective coverage of the cellular network to a larger geographic area (upto 15 miles). The unit has been extensively used in the fixed access to the cellular network to solve the wired communications problems in the rural areas. It can also be used to develop fast radio public telephone services to satisfy the communications for the time being and work as the CO relay tosimplify the registration s and lower the cost. Furthermore it can meet the requirement of mobile communications onboard vehicles, ships, trains, etc. All these enlarge the number of the network subscribers considerably so that it can utilize the resources better. General Instructions How to link with a charger, Office PBX and VOIP Main functions -- Payphone Caller ID Pin number locked (Optional) Block prefix number (Optional) Support OfficePBX Support VOIP Office PBX VOIP Description/ Unit Specifications UP MHz 890~915 1710~1755 1850~1910 WorkingFrequency DOWN MHz 935~960 1805~1850 1930~1990 Transmitting power dBm 33 Receiving sensitivity dBm -104 Atmosphere Kpa 86~106 Power Specifications Power mode: AC to DC a. Switch adaptor (without battery) 110-220V to 5V or 7-12V, 50/60Hz, 1.25A b. Switch adaptor (with Ni-MH battery) 110- 220V to 7.5V, 50/60Hz, 1.0A Backup battery: Standby: 20Hrs(Appr.) Continued Talking: 3Hrs(Appr.) Note: a. The battery will give the power when the normal power is off, and the battery power will be off when the normal power is On. B. The battery is for back up power only, It is not designed for normal power use. Quick Installation 1. Take off the cover of the SIM holder, then put in a SIM card into the holder. Receiving sensitivity dBm -104 2. Plug in a phone into the phone socket RJ-11 3. a. Install the antenna first, please screw the antenna tightly into the connector, and put the antenna in the purpose place. b. Connect the power, and put power switch ON. Power Specifications Power mode: AC to DC a. Switch adaptor (without battery) 110-220V to 5V or 7-12V, 50/60Hz, 1.25A b. Switch adaptor (with Ni-MH battery) 110- 220V to 7.5V, 50/60Hz, 1.0A Antenna information --- Frequency range: A:890 960MHz B:1710 1880MHz Banwidth: A:70MHz B:170MHz Gain: 2.15dBi or 5.5dBi (optional) Impedance: 50Ù Max Power: 50W Connector Type: SMA Size: Longth:30cm , 60cm and 100cm (optional) Weight: 120g Other Specifications Plastic cover: light blue or black Size 183mm 124mm 32mm(l\w h) G.Weight(complete set) 1.2Kg Circumstances: a temperature -20 ~50 b relative humidity 5%~95% Switches on the Box: --- ANT ON OFF SET WORK LOAD RJ11 Antenna Power Switch for power Switch for set or work Set for factory only by now Please put the switch on work Load/USB for the factory only Rj11 for phone line Gain: 2.15dBi or 5.5dBi (optional) Impedance: 50Ù Max Power: 50W Connector Type: SMA Size: Longth:30cm , 60cm and 100cm (optional) Weight: 120g NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queue Priorities
Take a look at the weight option in queues.conf. Available in CVS only I believe. Callum McGillivray wrote: Hi All, We have been playing around with call queues and asterisk and now have everything set up the way that we want it, bar 1 thing. When we have a scenario of an agent logged into several queues, we want to prioritize the queue so that calls in that queue are answered before all the rest. While we can prioritize calls within a queue, this seems to have no affect when an agent is logged into several queues (it probably has effect within the queue itself, but that's not quite what we are after). Does anyone know how to prioritize one queue over another, rather than just the calls within the queue? Thanks, Callum ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming context problem
Hello All, Can anyone u pls tell me the context pattern i need to add on sip.conf and extension.conf for incoming calls ... the senerio is i have a provider who routes a UK DID to my IP previously i was using ATA186 and calls were coming on ATA186 via sip and phone was connected to port 1 .. i didn`t had to do anything.. i want to use asterisk to attend the call and forward to a extension. how shld write the context for sip and extension.conf ? best wishes Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inbound ANI DNIS format
huh? That's a TDM/RBS type question. I've not seen most implementations of SIP interconnections doing things like that? On 5/12/05, Adam Robins [EMAIL PROTECTED] wrote: Hello, Being totally fed up with the lack of quality and reliability from both VoicePulse and BroadVoice, We are switching to a direct IP connection to Global Crossing. We've installed a local point-to-point T1 into their CO, and they will give/take SIP g729a directly and act as the gateway for us. In setting up the inbound SIP service, they are asking the question, In what format do I want my ANI DNIS presented? They provided examples, such as *ANI*DNIS, etc. Does anyone out there know how Asterisk expects to see this information on inbound calls? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best CPU config for dual-Xeon?
I have some beefy dual-Xeon servers that I will be using for Asterisk VoIP applications (i.e. no Zaptel cards). Using 2.6.11-1.14_FC3smp as the kernel (Fedora Core 3), and currently with Asterisk STABLE. My question is concerning the CPU setup, as I've seen conflicting or out-of-date suggestions: given the above config, should I have hyper-threading turned on or off? Turned on appears like 4 CPUs, and turned off will, I assume, appear as 2 CPUs. It's not clear to me what the issues with HT are/were, and whether they only relate to the use of hardware, interrupts, etc. or what, so any advice would be much appreciated. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceBlue GSM
Sorry for my delayed response Selon , I am setting up a test Asterisk box in our company to replace our current switchboard and well - GSM connection was one of the requirements for me to do to allow asterisk to replace our switchboard. (The others are not going to well... or they are finally done) Some digging around produced a Falcon 2D GSM modem... but I can't find nor imagine how to connect that device to asterisk (physically: a serial connection to the box is easy) but how asterisk will use it is a completely different question. And well the aim of the game is to impress - and to get asterisk to replace our PBX. 2-) You can buy a voip2GSM Gateway. To which you no longer need hardware... I was under the impression that you require hardware to connect your asterisk box up to the GSM network. Anyway, all that I require is a means to connect to a GSM provider (cellular network) to make and receive cellular calls. I have been googling a bit and the list has some info about it, but people were talking about using a Siemens Home station or the like... and that sounded a bit like a work-around to me. Then VoiceBlue... well, it looked good - I mean, it says it'll do everything that I will require and that it works with asterisk. (I presume as a SIP device or the like that you Dial). That was it - I could not find a price on the web you see - the South African Rand does not fair well against USD or the GBP... so I need to know if it is a solution for me. The added bit was the SMS capability - that I know asterisk can do but I need to see if the hardware supports it. To send - should be easy... to receive will be more difficult. But both is desired. (Asterisk+OpenXchange) just a thought. Kind Regards Etienne [EMAIL PROTECTED] wrote: Etienne, I am not sure I understand all what you require. Do you need to know the cost of the voiceblue of 2N or you need to find solution that can allow you send GSM calls ? There are several alternatives: 1-) Voiceblue as you mentioned; 2-) You can buy a voip2GSM Gateway. To which you no longer need hardware, you just register the voip2GSM devise to Asterisk and then it is ready to receive and send calls just like any other sip phone. Cost of this is around 400 USD / UNIT When you talk about sms capability, dyou want to originate or receive SMSs through the devise? Selon Etienne Pretorius [EMAIL PROTECTED]: Hello All * users. I have been looking for a way to allow GSM termination through Asterisk to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on asterisk with the ZAP channels via the Digium TDM 400P. I am unable to find any place that can tell me the cost of the VoiceBlue with a currency to I can calculate the cost of buying one. Alternativly - or just out of interist - I only really need to handle one GSM call @ a time and have a SMS capability... is there anyone that can suggest the best way to do so without doing a hack/patch to make a device to interact with asterisk? -- Kind Regards Etienne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr!
Good day all I installed asterisk-addons and now its logging nicely in my database But I want it to log in my usual log csv as well Please Let me know Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM gateway for Asterisk
On Thu, 2005-05-12 at 11:27 -0400, Kanuri, Seshu (Company IT) wrote: Folks! I am looking at a couple of models of Fixed GSM Gateways for the Purpose of VOIP connectivity and specifically to work with Asterisk. I found that these can be imported into USA for about $99.99 or about that. This is a one channel unit just like tellular, one of them has GPRS. Something like this is similar to what I was asking about in a different thread, however a SIP/GSM protocol converter would be more ideal. Passively passing all data from the GSM network to the mobile and vice versa, thus removing any requirement for a SIM in the GSM device that gets installed. Basically the mobile would register through this becuase the signal strength is stronger, outbound calls would be routed to the PBX via SIP (or other, SIP would make more sense as its more universal), inbound GSM calls would be transparently bridged to the real mobile, all auth data would be passed so the mobile would have the SIM and perform as if it were directly connected to the GSM network. A SIP IM to GSM SMS bridge would also be really ideal. The ability for the SIP interface to cause a call to be initiated to the GSM network would also be ideal (granted this would require the phone to accept the auth data and reply accordingly, which could be a bit tricky, but if the GSM mobile user attempted to place a call it should work, although routing for that would have to exist on the GSM protocol converter itself rather than via the PBX. This would effectively turn any GSM phone into a pbx extension and/or SIP phone, with the ability for calls to come into that phone from the GSM network. I strongly feel that SIP would be better than trying to tie in an Abis interface into the PBX (those do exist commonly as a nanocell or picocell transceiver). Because the protocol converter does not need to decrypt via A5 the GSM calls, GSM MoU approval should not be required. In theory one could buy gsm transceiver boards and make their own device using an embedded (or just nanoitx and non embedded) solution, slap on some supported operating system and asterisk in the unit itself. Granted it would not always need to be a full on asterisk implementation since it does a very limited subset of features, but could be. That could incrase the SIP to all supported protocols. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk starting problem
Bharat M. Sarvan wrote: Hello Everybody, I am having problems with starting Asterisk. The message what I am getting is; May 11 15:41:32 WARNING[5031]: res_musiconhold.c:728 moh_scan_files: Cannot open [cdr_addon_mysql.so]May 11 15:41:32 WARNING[5031]: loader.c:305 __load_resource: libmysqlclient.so.10: cannot open shared object file: No such file or directory May 11 15:41:32 WARNING[5031]: loader.c:463 load_modules: Loading module cdr_addon_mysql.so failed! I have configured the modules.conf for loading the cdr_addon_mysql.so. But still the problem persists. If you could please help me to figure as to whats wrong, it would be very kind of you. Regards, */Bharat M. Sarvan/* ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sounds like you are missing the mysql client libraries. -- Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice Recognition - Cases of success
Hi Isamar, There is a trial project underway for Asterisk and www.tellme.com but this is a commercial implementation of Speech Recognition using external resources and infrastructure. This will not be free. Let me know if you have a commercial application that has funding behind it. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-8307-3503 (Sydney in-dial) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Isamar Maia Sent: Thursday, 12 May 2005 11:27 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voice Recognition - Cases of success Hi Folks, I am planning to make a little project of voice recognition. I already browsed Voip Wiki and found some solutions. Before putting my hands on it to just do a little demo menu, I would like to hear from the list any succesful case using voice recognition and Asterisk. Best Regards, Isamar Maia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming calls picked-up then simply hanged-up
Are you sure you have context=from-pstn in your zapata.conf for the fxo channels? Julian. On 5/12/05, fhunter [EMAIL PROTECTED] wrote: I don't think my first posting went thru. I am trying to set up Asterisk for the first time. I am new to this. I am using [EMAIL PROTECTED] I have a TDM400P with one FXO and one FXS The system is working for outgoing calls and if I test incoming calls using . But when doing an actual call the system seems to answer the call and then immediately hang up. The incoming calls are set up to go from the PSTN to the Digital Receptionist. But I get the same behavior if I have incoming call send to the extension I have set up. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX to FWD?
Is anyone here able to make calls to FWD via IAX? I haven't beenable to for some while. I'd like to get to the bottom of the problem. There's been little response in the FWD support forum thus far. I can call my own number and it rings my server, but I cannot call any other number. It generates and error reporting everyone is busy at this time. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Something every TDMP user should know
Hi team, Not long ago a bunch of us were posting reports of a strange phenomenon where voice quality would pack up completely from time to time, typically resulting in loud crackling on the line and/or the voice channel breaking up completely. With our installation it would occur from time to time, typically when the * server was at it's busiest. Most of the time this problem would result in all users having to terminate their calls and re-establish them. After a lot of (very frustrating) troubleshooting we have have now gone two weeks without a re-occurrence of the problem and we are hoping that we may have finally resolved it altogether. I wanted to post a quick summary of the steps that we have taken to resolve this issue and what we think the problem turned out to be, as (from the number of responses to my last posts about this issue), it sounds like a few people have been experiencing it, so hopefully our experiences will help. The * server in question is based on a single-processor IBM xSeries 205 with a gig of RAM, SCSI 320 HDD's (RAID 1) and Red Hat ES 3. It uses ISDN (via CAPI and a four port Eicon Diva Pro Server card) and a mixture of SIP and analogue extensions. A TDM400P with four FXS ports supports the four analogue extensions (all Uniden cordless phones) and the SIP handsets consist of a mixture of BT102's and SNOM190's. Our turning point with this issue came when we bit the bullet and purchased a support incident from Digium. By this stage we had spent dozens and dozens of hours trying unsuccessfully to research and diagnose the problem and still had no accurate idea of what was causing it. Several people replied to our posts to this list saying that they were having a very similar issue as well, but no one had a clue what was causing it. Digium support zeroed in on the issue fairly quickly and we got the *distinct* impression that they have seen this problem many times before. They instantly got us to look at the output of zttest and we found that this was (in their words) 'extremely low', with 'best' and 'worst' readings of 99.975586% and 99.963379% respectively. They told us that we needed to be getting at least 99.98% and recommended that we: Check that the TDMP is on it's own IRQ (much to our embarrassment our card wasn't at the time, so we had to play with it a bit to get it to occupy a unique IRQ). Disable hyper threading on the Xeon CPU. Uninstall our SCSI hardware and replace it with IDE hardware. Upgrade to the latest stable releases of Asterisk, Zaptel and Libpri. We made changes 1 and 2 in the above list and are prepared to make changes 3 and 4 if we find the problem hasn't gone away. It hasn't happened in over two weeks now (after occuring many times per day for a while), so we hopefully won't have to throw out our SCSI hardware. After we made each change (1 and 2 were made about two weeks apart from each other) we found that the quality improved, with the incidence of the issue halving after '1' and disappearing (hopefully for good) after '2'. Incidentally the results of zttest *did not* noticeably improve after making these changes (it is still below 99.98%). Apparently our problem is related to the fact that the TDMP generates massive amounts of IRQ requests and that it becomes extremely upset if a suitable number of those IRQ requests are not honoured. Dispite the fact that a PCI device has to be able to share an IRQ in order to meet the PCI specification, it appears that having a TDMP sharing an IRQ with *anything* is a really really bad idea. I haven't been able to get an explanation about why hyper threading is a bad thing, but apparently high-performance devices such as SCSI adapters can cause resource contention issues with the TDMP, resource issues that the TDMP becomes very upset about. So hopefully we have seen the back of this problem and I have to say that I have been pretty dissappointed to find out that this issue appears to be relatively well known by Digium, but seemingly not publicised in the slightest. We searched for days to find anything relating to our issue but to no avail. Hopefully the next time someone has this issue they might find this mail and save themselves some of the frustration that we had. When we challenged Digiums advice about retarding the CPU (i.e. disabling hyper threading) and slowing I/O (by throwing out our SCSI RAID controller and replacing with IDE) they fell strangely silent - after getting prompt and meaningful responses to our requests they suddenly stopped responding at all. I think that this issue constitutes a pretty major flaw in the design of the TDMP and we will strongly avoid putting these cards into any * servers from now on. This is a real shame, as we as a company really want to reward Digium for all of their good work by actually buying their products, but we no longer have any faith in the design and suitability for production use of this product. Maybe it's time for Digium to think about
Re: [Asterisk-Users] IAX.CC/SixTel
This seems to be par for the course: You'll get a DID and poof, it's gone! Nobody answers the phone and nobody responds to tickets. For example: http://www.sixtel.net/tickets/view.php?ticket=xojnikrapqofyaspej -Steve Wiley Siler wrote: Anyone have an opinion about these guys and their recent performance? I need some local DIDs and they provide for my area code. Thanks, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming context problem
Hello All, Can anyone u pls tell me the context pattern i need to add on sip.conf and extension.conf for incoming calls ... the senerio is i have a provider who routes a UK DID to my IP previously i was using ATA186 and calls were coming on ATA186 via sip and phone was connected to port 1 .. i didn`t had to do anything.. i want to use asterisk to attend the call and forward to a extension. how shld write the context for sip and extension.conf ? best wishes Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice Recognition - Cases of success
Interactive Intelligence has a commercial Speech recognition API for this purpose. Check http://www.inin.com Or the specific Vocalite engine page at: http://www.inin.com/Products/vocalite/vocalite.asp Seshu Kanuri NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr!
As far as I can see in my installation, it does both. Nathan Altus Snyman wrote: Good day all I installed asterisk-addons and now its logging nicely in my database But I want it to log in my usual log csv as well -- - Nathan E. Pralle Give the director a serpent deflector. www.nathanpralle.com - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best CPU config for dual-Xeon?
Hi Tony, check out my recent post regarding our experiences with Hyperthreading and * with Zaptel cards. We have a few machines in the wild that *do* run Hyperthreading but no Zaptel cards and these work absolutely fine. My understanding is that the Hyperthreading problems are purely related to HW interrupts with Zaptel. My advice would be to leave HT turned on and just turn it off if you have problems with it - something that takes are few seconds in the BIOS and doesn't require any software changes. HT does provide significant performance improvements over non-HT... performance that could come in handy if your * server has a lot of calls in progress (and hence a lot of CODEC's to process). Cheers, Damian. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Tony Mountifield wrote: I have some beefy dual-Xeon servers that I will be using for Asterisk VoIP applications (i.e. no Zaptel cards). Using 2.6.11-1.14_FC3smp as the kernel (Fedora Core 3), and currently with Asterisk STABLE. My question is concerning the CPU setup, as I've seen conflicting or out-of-date suggestions: given the above config, should I have hyper-threading turned on or off? Turned on appears like 4 CPUs, and turned off will, I assume, appear as 2 CPUs. It's not clear to me what the issues with HT are/were, and whether they only relate to the use of hardware, interrupts, etc. or what, so any advice would be much appreciated. Cheers Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 Can't be unlocked
Odd problem here--I just got a couple of Cisco 7960s from Ebay that are not functioning as expected.. These 7960s can't seem to be unlocked for manual configuration via any mechanism that I can find. If you go to settings, there is no option 9 (unlock). Available options stop at 4 (Status). **# has no effect. The Phones report that thier current firmware version is 3.1 MF.G2. When plugged into a known good DHCP/TFTP server, the phones will *sometimes* get a DHCP lease that is reflected in SettingsNetwork Configuration, but at no point will they grab new firmware via TFTP. DHCP server logs show the phones trying acquire a lease and then immediately requesting a new one. If anyone has encountered a similar situation, please advise. Thanks, John Mensel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
