[asterisk-users] SkyHost is set to expire

Hi all, I'm actually using Asterisk 1.4.26 and Skype For Asterisk (last version). Everything was going fine, but yesterday I've got this messege when I've tried to restart asterisk (and skypeforasterisk): [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found license 'S4A-XXX' providing 5 conc

[asterisk-users] codecs and volume

Hi, Does using a different codec affect the volume of the voice? i was testing g711 and g729, voice seems to be softer on g729 compared to g711. sorry not really familiar on how codecs work. regards Ron ___ -- Bandwidth and Colocation Provided by ht

[asterisk-users] CDR is "NO ANSWER" when it should be "ANSWERED"

Hi, I'm having trouble with dialing out on analog lines. Asterisk can't seem to detect "answers". I have two zap groups. Group 1 is connected to an external analog PSTN provider. This group seems to work fine, especially with "answeronpolarityswitch". Group 2 is a group of "GSM gateways", ie

Re: [asterisk-users] Off Topic: Aastra BLF limit...

Hi Carlos, It's simply not possible due to a firmware limitation when general SIP and not Aastra proprietary mode (not enougth memory capacity). Don't lack your time by searching a non exisiting solution. Best Regards, Francois -Message d'origine- De : asterisk-users-boun...@lists.digi

Re: [asterisk-users] SkyHost is set to expire

Hi Daniel, Are you using a demo/beta version of Skype for Asterisk ? If yes, this status is normal, the demo/beta program is terminated from a while. I am using the real commercial (not free) and not getting that message. Best Regards, Francois -Message d'origine- De : asterisk-users

Re: [asterisk-users] SkyHost is set to expire

Hi, no I'm using the real commercial once. I've installed it in November 2009. Regards, daniel F6HQZ ha scritto: > Hi Daniel, > > Are you using a demo/beta version of Skype for Asterisk ? > If yes, this status is normal, the demo/beta program is terminated > from a while. > > I am using t

[asterisk-users] T.38 and Linksys SPA8000

Hello everyone. I'm trying to set up a SIP DID on a customer, which uses T.38 for faxing. Voice is working great, but I never configured anything using T.38 in Asterisk so I'm kinda lost. On my googling I found out that would be best letting the Linksys SPA8000 (for those that don't know, it's

[asterisk-users] set box IP from which send sip traffic

Hello, I've asterisk (asterisk 1.6.0.6) box with two network interfaces (two public IP: IP1 and IP2). SIP binds on 0.0.0.0 . Is it possible configure SIP peer/user to receive/send traffic from one of these IP? For example one client sends/receives traffic from IP1, other client send/receives t

Re: [asterisk-users] SkyHost is set to expire

Hi, I would contact customer support then. My licenses expire all 2029 (20 years after buying). regards, Philipp Daniel Grotti wrote: > Hi, > no I'm using the real commercial once. > I've installed it in November 2009. > > Regards, > daniel > > > > > F6HQZ ha scritto: > >> Hi Daniel, >> >>

Re: [asterisk-users] cheap ip phone with auto-answer

On Mon, 28 Dec 2009, Tim Nelson wrote: > - "Leif Neland" wrote: >> I want some cheap ip-phones with auto-answer, to work as paging system >> >> at dinnertime. >> Options, please. >> >> Leif >> > > I've had great luck using the BT201 phones from Grandstream for this > purpose. In fact, this i

Re: [asterisk-users] CDR

If asterisk enters the answered state at any point in the call, then the call disposition becomes answered. Thank you and have a nice day, Anthony Francis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Szasz Szabolcs Sent: Tuesday, D

[asterisk-users] ReceiveFAX G.711 + Realtime

Hello, We're trying to receive G.711 (aLaw) faxes on the asterisk and convert them to tif. With T.38, we have several issues, so we are trying to use G.711, since the gateway is located in the same LAN, so there's no bandwidth/packet-lose issue. We also use on the same Asterisk Real-Time proce

Re: [asterisk-users] SkyHost is set to expire

Hi, My license expires at 2029, but this isn't a license problem I think. I've downloaded my S4A from the following link: http://downloads.digium.com/pub/telephony/skypeforasterisk/asterisk-1.4/x86-32/skypeforasterisk-1.4_1.0.6-x86_32.tar.gz As Digium documentation says. Regards, daniel Phi

Re: [asterisk-users] ReceiveFAX G.711 + Realtime

Cyprus VoIP wrote: > My question: > > Is the following syntax for disabling T.38 support correct? > vm*CLI> -- Executing Set("SIP/Proxy-", "t38pt_udptl=no") > vm*CLI> -- Executing Set("SIP/Proxy-", "SIP_CODEC=aLaw") > vm*CLI> -- Executing Answer("SIP/Proxy-", "

Re: [asterisk-users] SkyHost is set to expire

Daniel Grotti wrote: > Hi, > > My license expires at 2029, but this isn't a license problem I think. > I've downloaded my S4A from the following link: > > http://downloads.digium.com/pub/telephony/skypeforasterisk/asterisk-1.4/x86-32/skypeforasterisk-1.4_1.0.6-x86_32.tar.gz > > As Digium documentat

Re: [asterisk-users] SkyHost is set to expire

Hi men, I am sure this is the demo version, not the correct actual licensed one. Fro mthe CLI, enter that : fax show version My Asterisk reply that : Fax For Asterisk Components: Applications: 1.6.1_1.0.15 Digium Fax T.38 Driver: 1.6.1_1.0.11 (optimized for c3_2_32) Digi

Re: [asterisk-users] Looking at Asterisk for 8000sq/ft residential use

Leif Neland wrote: I can't believe anyone would use RJ-11 any more. You can multi-purpose RJ-45 jacks to work with POTS lines. Run everything down to a central panel and send pots over the jacks that you need to. That way if you decide you need/want to go IP in the future, you're all set.

Re: [asterisk-users] ReceiveFAX G.711 + Realtime

> I have no idea where you got the idea that such a thing is possible... > it's not. sip.conf settings for SIP endpoints are not channel variables, > and cannot be modified from the dialplan unless the CHANNEL() dialplan > function has been specifically extended to support them. I was actually HOP

[asterisk-users] asterisk 1.6.2.0 sip channel to sip channel call dtmf inband not work.

we tested asterisk 1.6.2.0, found that when call from one sip_channel to another sip_channel , -- exten => _X.,1,Noop() exten => _X.,n,Dial(SIP/${EXTEN},50,TtgM) --

Re: [asterisk-users] Does A2Billing has mial list?

Hi Sucan, A2Billing doesn't have a mailing list but you may ask your specific question on A2billing Forum or maybe even here. This may be of intrest to you if you have an installation question: A2Billing automated install script : http://a2billing2asterisk.googlepages.com

Re: [asterisk-users] SIP Issue

You should set the ddwhome variable with the Set function or declare it on the global context. Try the Dial command with the dial string directly, before using the variable. Fro debugging purposes you should set debug and verbose at least to 10 and check the logs. Regards, Juan James A. Shigley

Re: [asterisk-users] Does A2Billing has mial list?

A2billing forum has a lot of information and questions are answered very fast. Try searching on the forum before posting, cause the answer may be there already. forum.asterisk2billing.org/ Regards, Juan Bruce Nik wrote: Hi Sucan, A2Billing doesn't have a mailing list but you may ask

Re: [asterisk-users] Any good dialplan code out there to implementvertical service codes?

It depends on what flavor of Asterisk and trunks (SIP/Zapata/DAHDI) you are using. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Freeman Sent: Monday, December 28, 2009 8:14 PM To: asterisk-users@lists.dig

Re: [asterisk-users] ReceiveFAX G.711 + Realtime

> Now, referring to the error above, I see (in voip-info.org) that > t38passthrough is an R/O variable and not an R/W, but in any case, I got > 0 as a result, so it should have been OK, and it's not, as ReceiveFAX > still sends a T.38 reINVITE. If I can't modify it, what should I do? For the tes

Re: [asterisk-users] ReceiveFAX G.711 + Realtime

Cyprus VoIP wrote: > Now, referring to the error above, I see (in voip-info.org) that > t38passthrough is an R/O variable and not an R/W, but in any case, I got > 0 as a result, so it should have been OK, and it's not, as ReceiveFAX > still sends a T.38 reINVITE. If I can't modify it, what shou

Re: [asterisk-users] Any good dialplan code out there to implementvertical service codes?

At the moment, 1.6.0.20 realtime with Dahdi 2.2, TDM is a TE420, but that won't be customer facing. Thanks- Joe Danny Nicholas wrote: > It depends on what flavor of Asterisk and trunks (SIP/Zapata/DAHDI) you are > using. > > -Original Message- > From: asterisk-users-boun...@lists.digium

[asterisk-users] asterisk billing transferred calls

Hi, I'm looking for an application to show all the calls received/made including (this is very important!) transferred calls because I need to track all the time spent on the phone by all my employees. There is a list here but they are too many to try them all: http://www.voip-info.org/wiki/vie

Re: [asterisk-users] SkyHost is set to expire

Daniel- > no I'm using the real commercial once. > I've installed it in November 2009. Did you have the demo version installed before the commercial version? I.e. install the commercial over the top of the demo version? -Jeff > F6HQZ ha scritto: >> Hi Daniel, >> >> Are you using a demo/beta v

Re: [asterisk-users] Realtime mysql extensions mutiple queries for each priority?

> On Monday 28 December 2009 23:49:13 Tilghman Lesher wrote: >> On Monday 28 December 2009 18:09:15 JR Richardson wrote: >> > I turned on console debug to see the actual mysql queries and to my >> > surprise and concern, I see every query for an extension priority >> > repeated 3 or more times prio

Re: [asterisk-users] identifying channel for softhangup

On 12/29/2009 1:01 AM, Jeremy Kister wrote: > e.g., in the first call, below, the channel name is > "SIP/vgw1-0075" -- the second call (on the same FXO port after a > soft hangup on the CLI) is "SIP/vgw1-0077" > > How can I extract this information in the dialplan so that I can use > th

Re: [asterisk-users] identifying channel for softhangup

Most of the asterisk-dev members read this discussion (In My Experience). ${EXTEN} in the case you state would be SIP/vgw1-0075. Perhaps this link would be helpful http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List -Original Message- From: asterisk-users-boun...@lists

Re: [asterisk-users] identifying channel for softhangup

On 12/29/2009 3:23 PM, Danny Nicholas wrote: > Most of the asterisk-dev members read this discussion (In My Experience). > ${EXTEN} in the case you state would be SIP/vgw1-0075. > Perhaps this link would be helpful > http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List Thanks f

Re: [asterisk-users] identifying channel for softhangup

You could do a System(core show channels) and grep out 911 and kill everything else; probably easier as an AGI call that a dialplan function, but both can be done. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf O

Re: [asterisk-users] identifying channel for softhangup

On 12/29/2009 3:54 PM, Danny Nicholas wrote: > You could do a System(core show channels) and grep out 911 and kill > everything else; probably easier as an AGI call that a dialplan function, > but both can be done. great idea; thanks! -- Jeremy Kister http://jeremy.kister.net./

Re: [asterisk-users] identifying channel for softhangup

Un-top-posting... > On 12/29/2009 1:01 AM, Jeremy Kister wrote: >> e.g., in the first call, below, the channel name is "SIP/vgw1-0075" >> -- the second call (on the same FXO port after a soft hangup on the >> CLI) is "SIP/vgw1-0077" >> >> How can I extract this information in the dialpl

[asterisk-users] Context Switches and Load Average spike - Asterisk Version 1.4.22

I am running Asterisk V 1.4.22 Twice during the last two days the Context Switches on our box has gone from about 7K to 80K in 2.5 hours. The load average would spike to 17, drop to 0.35 then spike again. When connecting to the console 'core show channels' will list the channels but not total ca

Re: [asterisk-users] identifying channel for softhangup

On 12/29/2009 3:54 PM, Danny Nicholas wrote: > You could do a System(core show channels) and grep out 911 and kill > everything else; probably easier as an AGI call that a dialplan function, > but both can be done. my end result just feels ugly. the loop is due to the fact that I have more than

Re: [asterisk-users] Looking at Asterisk for 8000sq/ft residential use

Before I start I am a Panasonic certified dealer AND I have installed over 100 Asterisk systems that are in production. That said for your application use Panasonic, DONT use Asterisk. Use the Panasonic KX-TDA50G. Supports up to around 50 ports. In addition to Analog and their proprietary Digital

Re: [asterisk-users] Looking at Asterisk for 8000sq/ft residential use

On Mon, Dec 28, 2009 at 5:42 PM, John Novack wrote: > > > Rick Huebner wrote: >> My brother-in-law is finishing up his McMansion and I've done all of the >> low voltage wiring and am starting the trimout. We are batting around >> what to do for a phone system and I'm torn between a Panasonic >> T

Re: [asterisk-users] Looking at Asterisk for 8000sq/ft residential use

On Mon, Dec 28, 2009 at 11:45 PM, Doug wrote: > At 16:13 12/28/2009, Rick Huebner wrote: > >My brother-in-law is finishing up his McMansion and I've done all of the > >low voltage wiring and am starting the trimout. We are batting around > >what to do for a phone system and I'm torn between a

Re: [asterisk-users] Looking at Asterisk for 8000sq/ft residential use

On Tue, Dec 29, 2009 at 11:30:21PM -0500, C F wrote: > Before I start I am a Panasonic certified dealer AND I have installed > over 100 Asterisk systems that are in production. > > That said for your application use Panasonic, DONT use Asterisk. > Use the Panasonic KX-TDA50G. Supports up to around

[asterisk-users] Auto-provisioining Polycom 430 wth dd-wrt router

Hi all, I'm trying to use a wrt54gl router running dd-wrt as a provisioning server for a remote installation. I've got dhcp working and I have provisioning files ready to go. I understand that I need to set bootp option 66 to point to the tftp/ftp/http server. In fact, I have this working co

[asterisk-users] Skype for Asterisk

Hi Sir, We have integrated Skype with Asterisk (skype user id:- rexesbposolutions). Each call which is coming to skype account is getting transfered to Asterisk Queue. It has following two cases: case 1: When we call from normal skype account to skype account (rexesbposolutions), everything is w

Re: [asterisk-users] Inquiry:Problem with Call Parking

On Wed, Sep 9, 2009 at 4:02 AM, Jeff LaCoursiere wrote: > > On Wed, 9 Sep 2009, hadi motamedi wrote: > > > Thank you for your message . But I tried to find it on my server , as the > > followings : > > #find / -name sip.cfg -print > > But it didn't return any result . Can you please let me know w