Re: [asterisk-users] cmd SET time value

Nope Tim, had tried that already, duznt work. Here's the cli output === Executing Set("SIP/4000-097afc90", "fwdTime=*|mon-tue|*|*") in new stack Sep 7 12:22:20 WARNING[24964]: pbx.c:6062 pbx_builtin_setvar: Ignoring entry 'mon-tue' with no = (and not last 'options' entry) Sep

[asterisk-users] Re: [asterisk-dev] UUI in calls

Hello, I need pass user to user information when I use originate a call from a sip phone to a queue. I need retrieve that uui when the calls rings in the sip phone logged in the queue. Any idea? Thank you First: let's move this to asterisk-users, where it belongs. This is not a developmen

Re: [asterisk-users] Native Chinese speaker needed

Steve,   You're right. Things seem to flow naturally when you don't think about it too much. But when it comes to defining a set of rules to determine what to say and when, it's so easy to miss (at least for me) until you actually listen to the result. :-)   On 9/6/06, Steve Underwood <[EMAIL PROTE

[asterisk-users] Query on Call Forward Feature codes for SIP users..

All, Could any one help me in configuring the feature codes for Call forward feature in asterisk..? How to configure the feature code *XX for activation /deactivation of call forward for SIP users ? Would appreciate , if somebody can help me more in detail . Thanks & Regards, -Sathish

Re: [asterisk-users] cmd SET time value

Single quotes -> ' <- work when I set other variables that contain special characters. Give that a try, -Tim On September 6, 2006 23:18, Benjamin Jacob wrote: > Hello ppl, > > Ive a couple of macros defined to call fwd based on time to a > number/voicemail. > Very elementary. > > ===

Re: [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

Dan Serban wrote: I have a system running Asterisk 1.2.10 (Debian packaging) and about 50 Linksys SPA-942 phones, after the initial config and mass deployment of the phones everything looks like it's configured well. When an incoming call is answered and then attempted to be "xfer'ed" via the so

[asterisk-users] cmd SET time value

Hello ppl, Ive a couple of macros defined to call fwd based on time to a number/voicemail. Very elementary. = 11. [macro-dialexten] 12. exten => s,1,Dial(SIP/${ARG1}) ; 1. [macro-stdpbx1exten] 2. exten => s,1,Set(fwdedNum=${DB(CFWD/${ARG1})}) 3. exten => s,n,GotoIf

[asterisk-users] the sounds quality of IAX2 channels are not good as SIP channels?

I use both IAX2 channels and SIP channels. IAX2 channels reduce bandwidth effectively. But sometime my cli show NOTICE[1281]: chan_iax2.c:1628 iax2_destroy: Avoiding IAX destroy deadlock WARNING[1281]: chan_iax2.c:708 jb_warning_output: Resyncing the jb. last_delay 28, this delay 1227, threshold

[asterisk-users] How to check which rtp ports my firewall let through?

I thought with iptable -L |grep udp I will find out which ports are open for the rtp stream, but I cannot get this info from here, or at least I cannot interpret it: # iptables -L |grep udp ACCEPT udp -- anywhere anywherestate RELATED,ESTABLISHED LOG

Re: [asterisk-users] using SIP to connect remote other VoIP server

Could you be more specific? Do you want to set up linking between two asterisk servers? Is this to a service provider? A single SIP registration and peer entry will handle multiple channels, and can also handle different numbers at the destinations. Try to get away from thinking of things

[asterisk-users] Polycom new firmware and bootrom

Well, it seems that Polycom has release new firmware 2.0.1 and bootrom 3.2.2. I've proceded to upgrade all my ip430 phones because they were essentially broken with the original firmware. All the phones boot up fine now, grab their files. They just won't talk to the asterisk server any more. I

Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel

Hi Leo, Sorry mate, I thought I had done some research but I only found one reference to it somewhere and it stated that if you have more than one VM running inside VMWare server v1.0, then there are timing issues where the clock seems to vary randomly. I figured that didn't apply to me since I h

Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel

HI Mojo, thanks for that. Sounds like a hidden option. It doesn't show up when I do a "tab" after I type "show translation" on the CLI. But to respond to your comment, I thought that's what it was, as in calculated based on the current load of the system but the fact is that there is absolutely

Re: [asterisk-users] Asterisk 1.2.11 and # key

The idea was to use single # as regular # and ## as blind transfer. And it looks like it was changed recently.On 9/7/06, Matt < [EMAIL PROTECTED]> wrote:Exactly... why WOULD you use two?   We use 1 # for blind transfer. If I need to enter # to an external call I enter it twice.   (## willsend a sin

Re: [asterisk-users] Digium G.729 codec binaries updated

Kevin P. Fleming wrote: As of a few minutes ago, the Digium FTP servers at ftp.digium.com contain a new set of Linux X86-32 and X86-64 G.729 codec binaries, along with a new registration utility. The new codec binaries were produced using GCC 4.1, and are more highly optimized than the previous

Re: [asterisk-users] How to test TE405P T1

Hi, If I just want to briefly test the T1, what is the basic config. I need to setup? Thanks! Andy Garth van Sittert wrote: Andy Chung (Power-All) wrote: Hi all, I have connected a T1IDA-P to the Digium TE405P. Checked with the Telco, and confirmed the T1 is up and connected. However, I h

[asterisk-users] using SIP to connect remote other VoIP server

    How to using SIP to connect remote other VoIP server? is it only running one line voice if I registered a one SIP account?    anybody can give me some sample configuration files? thanks a lot!   ___ --Bandwidth and Colocation provided b

Re: [asterisk-users] app_rxfax Only Receives One Page

Marco Mouta wrote: It has happened to me, with a x100p card, problems receiving fax that i've solved adjusting the gains. Don't understand quite well why you say that... Because the DTMF decoders and FAX modems work over a very wide range of signal levels. If you need to boost the signal to

Re: [asterisk-users] Native Chinese speaker needed

Steve Hsieh wrote: John, My patch, as it is now, would do: two thousand six year nine month ten two day two thousand six year nine month two day Is "couple" used instead of "two" anywhere else? You use it for day and minute. Is it ever used for

[asterisk-users] Re: Really bad phone line.. possible causes?

Mojo with Horan & Company, LLC wrote: What codec are your sip phones using? We'd have a similar, though immediate, degradation in audio quality using G.729 when zaptel was built with MMX optimizations. We use an AMD CPU. When zaptel was rebuilt without MMX optimizations we were back in busin

[asterisk-users] Garbled (quality probs) IAX2 & SIP calls Asterisk-to-Asterisk

I'm an almost 3 year Asterisk user now, since the pre-0.9 days, and I administer two Asterisk boxes. One of them is a small office with 16 users mostly using ATA phones hooked into Asterisk via a Rhino channelbank and a T1 card and the other is an even smaller office (mine) with just a couple

Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel

Ranj, Sigh :(. It would have save you and us a lot of time if you'd mentioned this fact earlier: Oh also, note that this system is running inside of a Virtual Machine with 768 RAM and a 3.4GHz CPU although NO other VM is active on this VM server. When running in a VM (like VMWare), the timing

[asterisk-users] Digium's response to posting of G.729 and G.723 source code

On September 4, 2006 an anonymous poster sent a message to these mailing lists containing a link to a package of source code claiming that it was "Digium's G.729 and G.723 codecs". As far as we can tell, that statement was not accurate. While the code posted appears to contain some of the same

[asterisk-users] Digium G.729 codec binaries updated

As of a few minutes ago, the Digium FTP servers at ftp.digium.com contain a new set of Linux X86-32 and X86-64 G.729 codec binaries, along with a new registration utility. The new codec binaries were produced using GCC 4.1, and are more highly optimized than the previous versions. In addition,

Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel

RR wrote: Also, at every start of (*), the "show translation" command shows different transcoding times without changing a single thing in the system in the way of config etc. Why is that? Because I believe they are calculated based on current system load. Whenever you'd like to see them recal

Re: [asterisk-users] Cisco MWI

Michiel van Baak wrote: On 17:04, Wed 06 Sep 06, Doug Lytle wrote: Or switch the phone to SCCP. It will give you a lot of extra power :) And segfaults, and phone lockups and grief. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Sa

[asterisk-users] faktortel

Is there anyone on this list who uses faktortel?   Anyone having problems with incoming iax today?     Cheers, Dean   ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update opt

Re: [asterisk-users] Cisco MWI

If you want to leave the @default off, if I remember correctly, you have to set searchcontexts=yes in voicemail.conf. On Wed, 2006-09-06 at 23:16 +0200, Michiel van Baak wrote: > On 17:04, Wed 06 Sep 06, Doug Lytle wrote: > > Steve Kennedy wrote: > > >But my mailbox (5200) is in default. > > > >

Re: [asterisk-users] Cisco MWI

On 17:04, Wed 06 Sep 06, Doug Lytle wrote: > Steve Kennedy wrote: > >But my mailbox (5200) is in default. > > > > I'm pretty sure that you'll still need to include the @context for the > MWI to work correctly. Or switch the phone to SCCP. It will give you a lot of extra power :) -- Michiel va

Re: [asterisk-users] Cisco MWI

Steve Kennedy wrote: But my mailbox (5200) is in default. I'm pretty sure that you'll still need to include the @context for the MWI to work correctly. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty

Re: [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

I use canreinvite=yes in my config files, and it does work, so maybe its a spa 941 misconfiguration. I think if nat=no sometime it has problems if you are behind NAT, but under same network it must not fail. Which firmware are you running on spas? Dan Serban escribió: Alberto Sagredo wrote:

Re: [asterisk-users] Cisco MWI

On Wed, Sep 06, 2006 at 03:36:53PM -0400, Doug Lytle wrote: > Steve Kennedy wrote: > >Phone itself. > >[S-5200] > This is incorrect. It should be: > [5200] > >mailbox=5200 That bit seems to work, phones registers ok and can receive and make calls. > You're missing the @context on your mailbox l

Re: [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

Alberto Sagredo wrote: > I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works > fine. Are you canreinvite=yes ?. > > I have not been notice any problem related to transferring calls (blind > and attended) > Thank you for your response, it gave me a nudge to check the configurati

[asterisk-users] Volume events causing talk off on Asterisk with Digium 411P

  We are experiencing random talk off events when we hear a loud volume event on the PSTN side of our calls.  We do not always hear the spurious DTMF, but I can see it in the console when I have the debug and verbose levels turned up.  We do however always have the associated brief periods

Re: [asterisk-users] app_rxfax Only Receives One Page

It has happened to me, with a x100p card, problems receiving fax that i've solved adjusting the gains.   Don't understand quite well why you say that...     On 9/6/06, Steve Underwood <[EMAIL PROTECTED]> wrote: Marco Mouta wrote:> Try to increase your rxgain, and check you have echocancel disabled

Re: [asterisk-users] Different MOH between waiting calls and transfer

equis software wrote: I know SetMusicOnHold command, but when a call is in the queue waiting, the queue function play the MOH configured 'myMOH', see the example below: I don't think you can with the dialplan. Maybe with the manager interface. You'd have to set it before the call is answere

Re: [asterisk-users] Different MOH between waiting calls and transfer

I know SetMusicOnHold command, but when a call is in the queue waiting, the queue function play the MOH configured 'myMOH', see the example below:exten => 90,1,Answerexten => 90,2,Ringingexten => 90,3,SetMusicOnHold(myMOH) exten => 90,4,Queue(myQueue|tw|||300)When the agent who answer the call tran

Re: [asterisk-users] Conditional IF based on IP address?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Hsieh wrote: > Thanks, Russ! > > Any suggestions on how to apply a subnet mask so that I can match an IP > that belongs to 192.168.0.0/23 , for example? Or > would the only way be to match the string using REGEX? > > >

Re: [asterisk-users] Conditional IF based on IP address?

Thanks, Russ!   Any suggestions on how to apply a subnet mask so that I can match an IP that belongs to 192.168.0.0/23, for example? Or would the only way be to match the string using REGEX?   On 9/6/06, Rushowr <[EMAIL PROTECTED]> wrote: -BEGIN PGP SIGNED MESSAGE-Hash: SHA1Rushowr wrote:>

Re: [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works fine. Are you canreinvite=yes ?. I have not been notice any problem related to transferring calls (blind and attended) Regards Dan Serban escribió: I have a system running Asterisk 1.2.10 (Debian packaging) and about 50 L

Re: [asterisk-users] Native Chinese speaker needed

John,     My patch, as it is now, would do:two thousand six year nine month ten two daytwo thousand six year nine month two day Is "couple" used instead of "two" anywhere else?  You use it for day andminute.  Is it ever used for year, month, hour, or second?     For year, it should

[asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

I have a system running Asterisk 1.2.10 (Debian packaging) and about 50 Linksys SPA-942 phones, after the initial config and mass deployment of the phones everything looks like it's configured well. When an incoming call is answered and then attempted to be "xfer'ed" via the soft button on the pho

Re: [asterisk-users] Conditional IF based on IP address?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rushowr wrote: > Steve Hsieh wrote: >> Greetings, > >> Is it possible to create a conditional IF inside extensions.conf based >> on the source IP address of a SIP phone (as opposed to extension)? What >> I would like to do is the following: > > >>

Re: [asterisk-users] Cisco MWI

Steve Kennedy wrote: Phone itself. [S-5200] This is incorrect. It should be: [5200] mailbox=5200 You're missing the @context on your mailbox line. i.e. my phones are in the from-sip context, so: [EMAIL PROTECTED] Doug -- Ben Franklin quote: "Those who would give up Essenti

[asterisk-users] Call parking and ringbacks

Greets all and TIA Here is my description short and simple: Call comes in --> Gets parked --> parking time expires --> rings back person who parked the call Is there a way for me to change the extension when ringing back? Normally Asterisk does this: 12125551212 --> AsteriskPBX --> exte

Re: [asterisk-users] Really bad phone line.. possible causes?

What codec are your sip phones using? We'd have a similar, though immediate, degradation in audio quality using G.729 when zaptel was built with MMX optimizations. We use an AMD CPU. When zaptel was rebuilt without MMX optimizations we were back in business. Jeff Turner wrote: Hi, I was wo

Re: [asterisk-users] Conditional IF based on IP address?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Hsieh wrote: > Greetings, > > Is it possible to create a conditional IF inside extensions.conf based > on the source IP address of a SIP phone (as opposed to extension)? What > I would like to do is the following: > > > 1. If SIP phone IP

[asterisk-users] Call parking and RTP traffic

Greetings I've noticed something odd while messing around with a test system and I'm not sure if this is a bug or not. I have three phones connected to an asterisk system in a remote office over a point to point T1 (no nat) all set up with canreinvite=yes. The phones are a Polycom 601, 501 an

Re: [asterisk-users] Native Chinese speaker needed

> Chinese uses a prefix for ordinal numebrs. However ordinal number are > not used in dates. In Chinese today (its the 6th September here) is > expressed as > zero six year nine month six day It is currently programmed to say "two thousand six year" for the year part. > You can skip the "

Re: [asterisk-users] Dell Poweredge SC430 and Digium cards compatability enquiry

I'm using a 400SC with two TDM400P cards and a TrixBox install.  This is an older version of the SC430 and it's been working like a champ.  I also have an SC430, but I'm not using it for Asterisk, so I can't give you insight on that one.  Sorry. On 9/5/06, Matthew Thompson <[EMAIL PROTECTED]> wrote

Re: [asterisk-users] Submenus

In fact, change the n in "exten => 1,n,Background(Met_Instructions)" to 3 like so: "exten => 1,3,Background(Met_Instructions)" and then add immediately after that line: exten => 1,4,WaitExten(3000) exten => 1,5,Goto(3) So if they don't answer immediately they are read the instructions again. Moj

Re: [asterisk-users] Asterisk 1.2.11 and # key

Exactly... why WOULD you use two? We use 1 # for blind transfer. If I need to enter # to an external call I enter it twice. (## will send a single # to the called party). On 9/6/06, Michael Strelnikov <[EMAIL PROTECTED]> wrote: So, I don't understand the reason to use ## then. I think you ar

Re: [asterisk-users] Is asterisk's mgcp support(NAS) Network access server package

i am out of office until 11.09.2006. If urgent please contact me on my mobile phone 00385 91 1988 815 davor ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.

[asterisk-users] Is asterisk's mgcp support(NAS) Network access server package

hi Is asterisk's mgcp support(NAS) Network access server package __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation pr

Re: [asterisk-users] app_rxfax Only Receives One Page

Marco Mouta wrote: Try to increase your rxgain, and check you have echocancel disabled Better still, try leaving the gains alone. If the gain controls were removed completely from Asterisk, support issues would decrease dramatically. Steve pls post your results On 9/6/06, *Steve Totaro*

Re: [asterisk-users] app_rxfax Only Receives One Page

Doug Lytle wrote: Steve Totaro wrote: I used the asterfax install script which installs iaxmodem. I will look into HylaFAX but I would love to find the solution to the one page problem while I pursue HylaFAX. Any errors being displayed on the Console? Have you run in in debug mode while a

[asterisk-users] Conditional IF based on IP address?

Greetings,   Is it possible to create a conditional IF inside extensions.conf based on the source IP address of a SIP phone (as opposed to extension)?  What I would like to do is the following:     1. If SIP phone IP belongs to 192.168.0.0/24 subnet, set CALLERID= 2. Else, set CALLERID=   T

Re: [asterisk-users] Cisco MWI

Steve Kennedy wrote: I have Asterisk 1.2.11 running and a Cisco 7960 (SIP v7.3). I cant seem to get the message waiting indicator working. I did try changing the MIME type as suggest, but then the phone kept continuously ringing. Any pointers? Steve Lets see your sip.conf entry for the p

Re: [asterisk-users] app_rxfax Only Receives One Page

Steve Totaro wrote: I used the asterfax install script which installs iaxmodem. I will look into HylaFAX but I would love to find the solution to the one page problem while I pursue HylaFAX. Any errors being displayed on the Console? Have you run in in debug mode while a fax in coming in?

Re: [asterisk-users] app_rxfax Only Receives One Page

Try to increase your rxgain, and check you have echocancel disabledpls post your resultsOn 9/6/06, Steve Totaro < [EMAIL PROTECTED]> wrote:Doug Lytle wrote:> Steve Totaro wrote: >> I am trying to setup a fax server and all I get is the first page of>> a multipage fax.  The first page is perfect qua

Re: [asterisk-users] app_rxfax Only Receives One Page

Doug Lytle wrote: Steve Totaro wrote: I am trying to setup a fax server and all I get is the first page of a multipage fax. The first page is perfect quality. I am not sure how to debug this. I have an HP DL320 with a quad Sangoma T1 board. You'll be much happier moving it over to HylaFAX

Re: [asterisk-users] Submenus

Try breaking up the contexts. Contexts are what you call 'submenus'. For example: [MetarMain] exten => 1,1,answer exten => 1,n,Background(Met_welcome) exten => 1,n,Background(Met_Instructions) exten => 3575,1,set(airport=ekrk) ; non-ambiguous exten => 3575,n,goto(Metar_Process,s,1) exten => 35

Re: [asterisk-users] Budgetones - multiple phones losing IP addressduring day

So the phones aren't loosing the IP address any longer?Just confirming what you meant by "It's running now". Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep

[asterisk-users] Cisco MWI

I have Asterisk 1.2.11 running and a Cisco 7960 (SIP v7.3). I cant seem to get the message waiting indicator working. I did try changing the MIME type as suggest, but then the phone kept continuously ringing. Any pointers? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 /

Re: [asterisk-users] asterisk t.38 fax failed

Ricardo,I found compilation error below, any thought?chan_sip.c:3895: error: `UDPTL_ERROR_CORRECTION_REDUNDANCY' undeclared (first use in this function)chan_sip.c:3898: error: `UDPTL_ERROR_CORRECTION_FEC' undeclared (first use in this function)chan_sip.c:3901: error: `UDPTL_ERROR_CORRECTION_NONE' u

RE: [asterisk-users] Cisco PIX firewall and nat=yes

Thanks I will check into this. I don't actually have access to the PIX (I have to talk to like 3 people to get to the person who actually manages this for the client) ...but that makes sense too I currently have it registering at 60 secs -Original Message- From: [EMAIL PROTECTED] [mailto

Re: [asterisk-users] Cisco PIX firewall and nat=yes

There is a "Timeout SIP" in the config. What is it set to? If it is less than the the qualify interval, which I believe is 60 seconds, then the PIX will close the inbound hole for qualify traffic. We've got lots of phones at several remote sites all running behind PIX's and all being NAT'd t

RE: [asterisk-users] Budgetones - multiple phones losing IP addressduring day

Its running now   tethereal -f "host 12.20.121.2 and icmp" Capturing on eth0     From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jessee J Holmes Sent: Wednesday, September 06, 2006 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Re: [asterisk-users] SIP client with video???

I've never experienced any of those problems. I can send video to other eyeBeam versions, 3.0 is the only version that supports video on the lite side. I've never lost any SIP information and only one registration isn't a big deal to me. If you need more than one buy the full eyebeam version. On 9/

Re: [asterisk-users] Budgetones - multiple phones losing IP address during day

Garth,This may be a silly question, but are you running the latest firmware on the phones from Grandstream? If not, try upgrading one phone to see if it helps solve the problem.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP pro

RE: [asterisk-users] Cisco PIX firewall and nat=yes

Title: Message As a follow up those commands helped with the outbound calls but inbound still had issues.  Asterisk would still show the peer UNREACHABLE.  Turning off qualify has fixed the problem!   Bill   From: Bill D'Anjou [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 2

[Asterisk-Users] Which SIP hardphone with embedded VPNClient ?

Hi,Snom offers sip hardphones with embedded VPN client (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/162680/match=hardphone+vpn+client ).Beside that, I've read Zultys used to provide one (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/162680/match=hardphone+vpn+c

Re: [asterisk-users] app_rxfax Only Receives One Page

Steve Totaro wrote: I am trying to setup a fax server and all I get is the first page of a multipage fax. The first page is perfect quality. I am not sure how to debug this. I have an HP DL320 with a quad Sangoma T1 board. You'll be much happier moving it over to HylaFAX and iaxmodem. Ver

Re: [asterisk-users] Has anyone tried to install both digital card and analog card in one machine

We are currently working with two TDM2400P, with 32 FXS ports and one TE205P all on the same machine, and it works fine, (haven't done any stress testing yet though, maybe if someone could share his/her experience with high load) Xue Liangliang escribió: Hi, all, I am not sure whether we

Re: [asterisk-users] How to notify an ACD agent before he/she picks up

Anyone knows if I can get to know the operator ACD choosed to send the call by using Realtime Queue, or maybe via the manager? Hi, I need to send a message to an agent when the ACD starts to ring on he/she. I have and application already built that sends such a message (just like a cti),

Re: [asterisk-users] asterisk t.38 fax failed

In sip.conf add to [general] context and to every peer context that you want to register in Asterisk to use T.38 the following lines: t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no In udptl.conf file I have the following configurations: [general] udptlstart=4000 udptlend=4999 T38FaxUdpEC = t38UDPRedu

Re: [asterisk-users] Native Chinese speaker needed

Hi, John Williams wrote: No, I'm not looking for a voice talent. I have been deploying an IVR in my company's China office, and our people there complained about the way asterisk spoke the date in Chinese. After discussing it with them, I have submitted a patch, which can be found on the Digi

[asterisk-users] app_rxfax Only Receives One Page

I am trying to setup a fax server and all I get is the first page of a multipage fax. The first page is perfect quality. I have googled and found people with the same problem but no good answers. The one answer given was to use a multipage tiff viewer which I did and confirmed that only one

Re: [asterisk-users] asterisk t.38 fax failed

Ricardo,Thanks, could you please share some of your t.38 passthrough configuration in sip.conf and also udptl.conf?Thanks,Ricardo Carvalho <[EMAIL PROTECTED]> wrote: No, T.38 doesn't work with Asterisk. Only works with Asterisk t38passthrough patch that you can find at URL: http://bugs.digium.com/f

[asterisk-users] Asterisk video support

Hello to all I used SER for SIP calls with video, but now Im trying the same in Asterisk and It doesnt work. I m using X-Lite 3.0 (the same that worked with SER). Do Asterisk needs any special configuration to allow SIP calls with video between its clients? Regards Joao Pereira Asterisk's su

Re: [asterisk-users] SIP client with video???

The problems with X-Lite 3 are: - just accepts one SIP registration - doesnt send video to other X-Lite or eyeBeam versions - sometimes loses the SIP informations when you reboot the PC . Regards Joao Pereira Blake Krone wrote: What's wrong with X-Lite 3.0? I haven't had any issues with it

Re: [asterisk-users] Deadlock

On 9/5/06, Michael Welter <[EMAIL PROTECTED]> wrote: (I'm getting 404 Not Found from the search engines) I have a system that gets a deadlock every week or so. On the logs I have many "channel.c:787 channel_find_locked avoided deadlock for 0x837730" messages. The system has an Eschelon T1 with

Re: [asterisk-users] Unable to make calls from CallManager to Asterisk

What version of call manager?On 9/5/06, Anantha Padmanabha.M.L <[EMAIL PROTECTED]> wrote: HI,I have successfully integrated CallManager and Asterisk and was able to make call from one of Asterisk phone to CallManager Phone.But Could not able to make call from CallManager to asterisk.I have also tr

Re: [asterisk-users] How to test TE405P T1

Andy Chung (Power-All) wrote: Hi all, I have connected a T1IDA-P to the Digium TE405P. Checked with the Telco, and confirmed the T1 is up and connected. However, I have no idea how to test the T1 is really work, because the Asterisk server not yet be configure. Anyone has the method on how to

Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

Hi Marco, found drivers versions :) Our PBX working without msns parameter has: - ISDN line provided by fastweb italy - ISDN driver: install-misdn version 0.2.1-rc13 Other PBX NOT working without msns parameter has: - ISDN line provided by telecom italy - ISDN driver: install-misdn

Re: [asterisk-users] Asterisk + Samsung OffServ 500

Eugeniy Khvastunov wrote: What signaling method i should use for connecting Asterisk(Gentoo, Tormenta 2) + Samsung OffServ 500 by PRI flow? What parametrs in zaptel.conf, zapata.conf? --- Какой метод сигнализации нужно использовать при подключении Asterisk(Gentoo, Tormenta 2) + Samsung OffServ

Re: [asterisk-users] includes in realtime ??

lol RR. will def do some RnD on this one, and wil get back. have put this on the back burner for now. thanks again. cheerz Ben RR wrote: I use rtcachefriends=yes and any changes I make in my database become effective immediately along with also getting the MWI functionality. Even though wha

Re: [asterisk-users] macros in Realtime

Thanks Simon, will try and get back on this one . Ben. Simon Woodhead wrote: Hi Ben, Yes it is but you need to remember to still include [macro-stdpbx1exten] switch => Realtime/ in your extensions.conf Simon On 9/6/06, * Benjamin Jacob* <[EMAIL PROTECTED] > wro

[asterisk-users] cmd SET time value

Hello ppl, Ive a couple of macros defined to call fwd based on time to a number/voicemail. Very elementary. = 11. [macro-dialexten] 12. exten => s,1,Dial(SIP/${ARG1}) ; 1. [macro-stdpbx1exten] 2. exten => s,1,Set(fwdedNum=${DB(CFWD/${ARG1})}) 3. exten => s,n,GotoIf(

Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

Hi Marco, unfortunately I cannot make any further test on that machine because it has already been delivered to the customer but I can tell you that its extensions.conf is the same of our production PBX which has not that problem. The only differences are the telco and the ISDN driver. If you

Re: [asterisk-users] Answer Machine detection

We use app_amd in the VICIDIAL project and it works fairly well(better than 60% if you tune it right), although I still don't recommend that anyone use Answering Machine detection. http://bugs.digium.com/view.php?id=5959 When we usually do automated notification campaigns we will most often wait

RE: [asterisk-users] how to setup poxy sip server

Hi Ranjeet,If you just want a SIP proxy that can route SIP messages without taking care of media flows, you should consider using SER or OpenSer. Only use Asterisk is a PBX, so use it when you need PBX's functionalities like voicemail, IVR ... You can find documentation about Asterisk's configurati

Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

Also your problem could be related with the Answer() you weren't answering the calls on your previous extensions.confPls test both configs with and without answer and reply your results. On 9/6/06, Marco Mouta <[EMAIL PROTECTED]> wrote: Hi,Multiple Subscriber Number. This is a telephone number as

Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

Hi,Multiple Subscriber Number. This is a telephone number associated with an ETS 300 BRI line. Providers of ETS 300 often give you three MSNs with a BRI, although additional MSNs can be purchased. An ISDN terminal will "ring" (provide an alerting signal) only when calls are made to the MSN

[asterisk-users] Answer Machine detection

I use Asterisk mainly in the IVR world and sometimes do a few outbound based campaigns. The horrible subject of Answer Machine Detection as lifted its head again. To my knowledge there are 3 ways to deal with it. 1) When the call is answered, you please a message something like "There is an im

[asterisk-users] how to setup poxy sip server

Hi,   I want to setup proxy sip server. In case of asterisk I don’t know how to do that. Please help me doing this.   Thanks, Ranjeet   The information contained in, or attached to, this e-mail, contains confidential information and is intended solely for the use of the indi

Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

Hi Marco, it seems that msns=* is necessary to make Asterisk work correctly...I do not why... We have another PBX with a monoBRI but have not this problem, maybe is the different ISDN telco or the old misdn driver does not complainthe important is that now it works!! I have only to fix som

Re: [asterisk-users] Budgetones - multiple phones losing IP address during day

GarthSounds like a DHCP lease issue that the BT102's are not playing nicely with?RegardsRobOn 06/09/06, Brandon Galbraith < [EMAIL PROTECTED]> wrote:Garth,Are they all on the same switch? Possibly could be a network-level issue, and not something wrong with the phones. -brandonOn 9/6/06, Garth van

Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

msns is as far as i know, similar to DIDs but it includes the complete Dialed number (the number your customer has dialed to call you).put msns=*also test this:[outbound_isdn]exten=> _X.,1,Answer() exten => _X.,n,Playback(vm-goodbye)exten => _X.,n,Hangupexten=> s,1,Answer() exten => s,n,Playback(vm

[asterisk-users] mobile refusing call

Hi list,   I have a problem. I have an asterisk <--> Cisco Pots gateway. The problem is when i call via sip over the asterisk over the pots GW to a mobile phone and refuse th ecall on this mobile the sip phone is still ringing. it seems the cisco gw se on th eone site that the call ist busy/refu

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