Re: [asterisk-users] Collect digits from the Callee after the Call is connected.

Doug Lytle wrote: > Simith Nambiar wrote: > >> Hello Doug, >> Thank you for your response, if you see my >> e-mail above, i wanted the Read to happen after the Call is connected >> (i.e after Dial >> > > > Pen and paper? > > Seriously, I thought you were just h

Re: [asterisk-users] CDR Desgin

>On Mon, Nov 24, 2008 at 6:56 PM, Steve Murphy <[EMAIL PROTECTED]> wrote: > For the moment, let's not worry about the implementation. Let's > get consensus on the spec first. In the scenario, where A calls B, > B xfers A to C, C xfers A to D, or some such similar scenario, > half the world wants a

[asterisk-users] OSLEC build errors on DAHDI [was: Re: HPEC performance]

On Wed, Nov 19, 2008 at 03:46:35PM -0700, Joseph L. Casale wrote: > >Not trivial but not as voodoo as before: > > > > http://docs.tzafrir.org.il/dahdi-linux/#_oslec > > Tzafrir, > I pulled down linux-2.6.28-rc5.tar.bz2 and followed the doc, now > when compiling I get the following: > WARNING: "os

Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem

On Tue, Nov 25, 2008 at 2:19 AM, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote: >> On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus <[EMAIL PROTECTED]> > wrote: >> > I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers >> > and

[asterisk-users] Set a specific registration expiry value to a given peer without touching defaultexpiry in sip.conf ?

Hi, I've got several trunks in my 1.6.0.1 setup. One of them is asking for 1800 sec registrations. You can provide this value setting defaultexpiry to 1800 in sip.conf but how can you specify this duration to this specific trunk and not affect the others ? An option to register statement in sip.c

[asterisk-users] Disabling Call-Waiting

Hello! I have a few sip devices and it is necessary for me to disable call-waiting and immediately return a busy signal if the sip's channel is busy on them. What is the procedure to do so in Asterisk 1.4? Thank you, Elliot ___ -- Bandwidth and Coloca

Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem

Hello! On Tue, Nov 25, 2008 at 2:25 AM, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > On Monday 24 November 2008 06:19:44 pm Tilghman Lesher wrote: >> On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote: >> > On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus <[EMAIL PROTECTED]> >> >> wrote: >>

Re: [asterisk-users] Disabling Call-Waiting

You can try call-limit = 1 in sip.conf for each phone. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Elliot Murdock Sent: martes, 25 de noviembre de 2008 11:04 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Disabling Call-Waiting Hello! I have a few

Re: [asterisk-users] Disabling Call-Waiting

Hello Sebastian, Thanks! However, I tried using call-limit and the phones are still getting the calls. Maybe I am doing something wrong. Elliot On Tue, Nov 25, 2008 at 3:14 PM, Sebastian <[EMAIL PROTECTED]> wrote: > You can try call-limit = 1 in sip.conf for each phone. > > > > > > *From:* [EMAI

Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem

I think call-limit is for 1 outbound and 1 inbound so if you get a call you wont get other but if you do an outbound call an incoming call will be allowed. Maybe you can configure 1 line in your phone or try Check device state before make the call in extensions. -Original Message- From:

Re: [asterisk-users] Disabling Call-Waiting

On Tue, 25 Nov 2008, Elliot Murdock wrote: > Hello! > > I have a few sip devices and it is necessary for me to disable call-waiting > and immediately return a busy signal if the sip's channel is busy on them. > > What is the procedure to do so in Asterisk 1.4? Read the phone manual and work out h

[asterisk-users] AsteriskNOW 1.5 upgrade from 1.4 to 1.6

This link (http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/ ) seems to indicate that in order to upgrade AsteriskNOW v1.5 from Asterisk 1.4 to 1.6, it's as easy as installing an upgrade package. Does anyone know where to find that upgrade package? If it does

Re: [asterisk-users] The sound is played but I did not hear

there is no answer ??? 2008/11/12 jhon digital21 <[EMAIL PROTECTED]> > Hello, > > I have another little problem with my ZAPs channels, in fact, when I > received a call, I heard no sound while in the CLI, sound is played: > > -- Starting simple switch on 'Zap/4-1' > -- Executing [EMAIL PROTECTED]

Re: [asterisk-users] The sound is played but I did not hear

jhon digital21 wrote: > there is no answer ??? > Try putting a Wait(1) before playing the sound effect. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

Re: [asterisk-users] Disabling Call-Waiting

Sorry for hijacking this thread, but I need something similar in opposite way the original poster wants: I have real incoming phone number and all calls on this phone number are redirected to local extension: 1234567 -> 515 where "1234567" is a phone number and "515" is a local extension in ast

Re: [asterisk-users] AsteriskNOW 1.5 upgrade from 1.4 to 1.6

Jason Lixfeld wrote: > This link > (http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/ > > ) seems to indicate that in order to upgrade AsteriskNOW v1.5 from > Asterisk 1.4 to 1.6, it's as easy as installing an upgrade package. > > Does anyone know where to fin

Re: [asterisk-users] Disabling Call-Waiting

On Tue, 25 Nov 2008, Mikhail (Plus Plus) wrote: > Sorry for hijacking this thread, but I need something similar in > opposite way the original poster wants: > > I have real incoming phone number and all calls on this phone number are > redirected to local extension: 1234567 -> 515 > where "1234567

Re: [asterisk-users] Disabling Call-Waiting

You can try: exten => 1234567,1,Dial(SIP/515) exten => 1234567,101,Dial(SIP/516) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mikhail (Plus Plus) Sent: martes, 25 de noviembre de 2008 02:08 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussi

Re: [asterisk-users] Disabling Call-Waiting

Hello! Thanks for the responses. I'll look into the phone devices themselves. I am wondering, if call-limit did not solve my problem, what is the call-limit parameter supposed to do anyway? Later, Elliot On Tue, Nov 25, 2008 at 6:29 PM, Sebastian <[EMAIL PROTECTED]> wrote: > You can try: > >

Re: [asterisk-users] Disabling Call-Waiting

On Tuesday 25 November 2008 10:07:34 Mikhail (Plus Plus) wrote: > Sorry for hijacking this thread, but I need something similar in > opposite way the original poster wants: > > I have real incoming phone number and all calls on this phone number are > redirected to local extension: 1234567 -> 515 >

Re: [asterisk-users] CDR Design

We are suggesting the same thing, what you describe is multidimensional. If you think of the cdr's as being in a database, and say you wanted to have it show you all the calls today and all the calls that are associated with that call. Your select grabs the first dimension, a list of all calls.

Re: [asterisk-users] Disabling Call-Waiting

On Tuesday 25 November 2008 10:46:49 Elliot Murdock wrote: > Thanks for the responses. I'll look into the phone devices themselves. > > I am wondering, if call-limit did not solve my problem, what is the > call-limit parameter supposed to do anyway? The call-limit is actually kind of deprecated.

Re: [asterisk-users] CDR Desgin

On Tue, 2008-11-25 at 08:06 +, Grey Man wrote: > >On Mon, Nov 24, 2008 at 6:56 PM, Steve Murphy <[EMAIL PROTECTED]> wrote: > > For the moment, let's not worry about the implementation. Let's > > get consensus on the spec first. In the scenario, where A calls B, > > B xfers A to C, C xfers A to

Re: [asterisk-users] CDR Design

On Mon, 2008-11-24 at 09:12 -0700, Anthony Francis wrote: > It is my belief that before redesigning the CDR engine some time should > be spent looking at current PSTN CDR formats and what information is > kept in them. > The main problem that I feel we face is that calls can be complicated, > bu

[asterisk-users] MOH Realtime

Anybody was able to set it up?? I can't make it work, any idea?? Ast 1.6.0.1 Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists

Re: [asterisk-users] Disabling Call-Waiting

If is deprecated how do you treat a queue (realtime), that has to have just one call for agent?? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: martes, 25 de noviembre de 2008 03:37 p.m. To: Asterisk Users Mailing List - No

Re: [asterisk-users] Disabling Call-Waiting

Try looking at the DEV_STATE function (available separately on Asterisk-1.4). It will tell you the status of the phone before you call the Dial() application. __Yehavi: 2008/11/25 Sebastian <[EMAIL PROTECTED]> > If is deprecated how do you treat a queue (realtime), that has

Re: [asterisk-users] CDR Design

Yes, I know we are suggesting the same thing... I just thought you are suggesting putting this multidimensional CDR in one row (which of course requires data structure other than a simple comma separated row - XML perhaps). I did not understand you were referring to a conceptual multi-dimensiona

Re: [asterisk-users] Disabling Call-Waiting

But in this case I'm using queue not dial. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yehavi Bourvine Sent: martes, 25 de noviembre de 2008 05:33 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Disabling Call-Waiting Try

Re: [asterisk-users] pick up IAX2 calls

I think it doesn't work across channel types. So it works (if I recall correctly) in IAX or in SIP or in ZAP, but not in mixture. I think that if you have a Dial() that rings several extens, then any of the technologies involved can pickup with *8 So if you have Dial(IAX/fred&SIP/bill&zap/mark)

[asterisk-users] half channel audio after upgrade to 1.4.18

I upgraded from 1.2 to 1.4.18 After upgrading I get half channel audio on SOME phones. I have Cisco 7960 that works, I have a wireless polycom 8002 phone that works. However, my polycom 501's are getting half channel audio on EXTERNAL calls. Internal calls are OK. I have enabled nat=yes on all

Re: [asterisk-users] Disabling Call-Waiting

Hello, I am wondering if a queue feature that blocks call-waiting should be submitted. As opposed to a regular line, where a call-waiting feature makes sense, for a queue, call-waiting just doesn't. Most queue agents are taking phone calls in a strict order and accordingly, an annoying call-wait

Re: [asterisk-users] two sip listening ports for single asterisk

Hi guys, I told my network admin to do what was advised in this thread. It works very well for incoming calls but outgoing calls hangup exactly after 20 secs everytime while displaying the following message on cli: v[Nov 19 10:26:26] WARNING[18617]: chan_sip.c:1910 retrans_pkt: Maximum retries exc

Re: [asterisk-users] CDR Desgin

Steve Murphy <[EMAIL PROTECTED]> writes: > I'll modify my RFC to reflect this line of thinking. Yes, it is a > bit of shift. And, yes, the implementation will make this new > approach optional, and not default. I believe the whole approach is sound and I just want to voice my support for it. /B

Re: [asterisk-users] Disabling Call-Waiting

"Elliot Murdock" <[EMAIL PROTECTED]> writes: > I am wondering if a queue feature that blocks call-waiting should be > submitted. Doesn't Queue() already disregard busy phones? I must admit that we run with callwaiting turned off, so it isn't something I get to test very often, but I could have sw

[asterisk-users] Connecting AS5350XM with Asterisk

Hello, everybody! I need help connecting my Cisco AS5350 to Asterisk. What i want to do is forward all outgoing calls from Asterisk server to Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP. How could this be done? Thanks in advance Atif Shahzad _

[asterisk-users] Connecting AS5350XM with Asterisk

Hello, everybody! I need help connecting my Cisco AS5350 to Asterisk. What i want to do is forward all outgoing calls from Asterisk server to Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP. How could this be done? Thanks in advance Atif Shahzad _

[asterisk-users] Connecting AS5350XM with Asterisk

Hello, everybody! I need help connecting my Cisco AS5350 to Asterisk. What i want to do is forward all outgoing calls from Asterisk server to Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP. How could this be done? Thanks in advance Atif Shahzad _

Re: [asterisk-users] Connecting AS5350XM with Asterisk

Set up a SIP dial peer and an outbound POTS dial peer. Bear in mind that the gateway shunts calls POTS->VOIP and VOIP->POTS by default, so you can use the same destination pattern matching for both in this simple scenario, but if it gets any more complicated than that, some degree of translati

Re: [asterisk-users] Disabling Call-Waiting

Benny Amorsen wrote: > "Elliot Murdock" <[EMAIL PROTECTED]> writes: > > >> I am wondering if a queue feature that blocks call-waiting should be >> submitted. >> > > Doesn't Queue() already disregard busy phones? I must admit that we > run with callwaiting turned off, so it isn't something I

Re: [asterisk-users] half channel audio after upgrade to 1.4.18

Jerry Geis wrote: > I upgraded from 1.2 to 1.4.18 > > After upgrading I get half channel audio on SOME phones. > > I have Cisco 7960 that works, I have a wireless polycom 8002 phone that > works. > However, my polycom 501's are getting half channel audio on EXTERNAL calls. > Internal calls are OK.

Re: [asterisk-users] Connecting AS5350XM with Asterisk

The procedure you explain is for outbound or inbound for Asterisk or Can you tell me the procedure for only outbound from my Asterisk server to Cisco 5350? On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov <[EMAIL PROTECTED]>wrote: > Set up a SIP dial peer and an outbound POTS dial peer. Bear in mi

Re: [asterisk-users] Connecting AS5350XM with Asterisk

Alex, 1 more thing my gateway is configured with H.323 so tell me how can I configure it with SIP? On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov <[EMAIL PROTECTED]>wrote: > Set up a SIP dial peer and an outbound POTS dial peer. Bear in mind > that the gateway shunts calls POTS->VOIP and VOIP->

Re: [asterisk-users] Connecting AS5350XM with Asterisk

Use only the second dial peer. A T I F wrote: > The procedure you explain is for outbound or inbound for Asterisk or Can > you tell me the procedure for only outbound from my Asterisk server to > Cisco 5350? > > On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov > <[EMAIL PROTECTED]

Re: [asterisk-users] Connecting AS5350XM with Asterisk

My attention to my dial peer. It has nothing about H.323 and much about SIP. A T I F wrote: > Alex, > > 1 more thing my gateway is configured with H.323 so tell me how can I > configure it with SIP? > > On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov > <[EMAIL PROTECTED]

Re: [asterisk-users] Connecting AS5350XM with Asterisk

1. dial-peer voice 500 voip I use this configuration for inbound to asterisk. 2. dial-peer voice 510 pots description Fancy PRI - Outgoing huntstop destination-pattern .T direct-inward-dial forward-digits 10 And use this configuration for outbound from asterisk to Cisco 535

Re: [asterisk-users] Connecting AS5350XM with Asterisk

A T I F wrote: > 1. dial-peer voice 500 voip > > I use this configuration for inbound to asterisk. > > 2. dial-peer voice 510 pots > description Fancy PRI - Outgoing > huntstop > destination-pattern .T > direct-inward-dial > forward-digits 10 > > And use this configuration

Re: [asterisk-users] CDR Desgin

> > To me the obvious answer is to provide a CDR for every call leg so for > > A calling B via Asterisk there would be two CDRs produced. It's far > > far easier to disregard the unwanted CDRs than it is to try and > > generate the missing ones and in some cases it's virtually impossible. > > If it

Re: [asterisk-users] Connecting AS5350XM with Asterisk

Alex, I am new to *5350* my senerio is this; *1. ASTERISK ---outgoing-->CISCO5350 (both have live IP configured) 2. ASTERISK <-incomingCISCO5350* I need only configurations for Cisco for both in coming n outgoing to asterisk. IF you need configuration of my Cisco Gateway I will pro

Re: [asterisk-users] Connecting AS5350XM with Asterisk

This is not the homework list. A T I F wrote: > Alex, > > I am new to _*5350*_ my senerio is this; > > *1. ASTERISK ---outgoing-->CISCO5350 (both have live IP configured) > > 2. ASTERISK <-incomingCISCO5350* > > I need only configurations for Cisco for both in coming n outgoing t

Re: [asterisk-users] Connecting AS5350XM with Asterisk

I mean to say that I have assigned this task from my management to configure it, I am familiar with Asterisk but first time I am using Cisco 5350. On Tue, Nov 25, 2008 at 4:19 PM, Alex Balashov <[EMAIL PROTECTED]>wrote: > This is not the homework list. > > A T I F wrote: > > > Alex, > > > > I a

Re: [asterisk-users] Connecting AS5350XM with Asterisk

I provided you all the information you need in an elementary sense. Configuring the device comprehensively is a rather lengthy subject and cannot be meaningfully addressed in the scope of an email thread. The answer basically boils down to, "learn the Cisco VFC platform." There are plenty of s

Re: [asterisk-users] pick up IAX2 calls

hi I'm using only IAX extensions and inserted callgroup=1 and callpickup=1 for all IAX extensions in iax.conf. Didn't works for while. thanks Tim Panton wrote: I think it doesn't work across channel types. So it works (if I recall correctly) in IAX or in SIP or in ZAP, but not in mixture. I

[asterisk-users] bridging - Didn't get a frame from channel

Hi, I am having a difficulty with getting two realtime user’s to bridge on answer. I have managed successfully to bridge the same two users/channels via the Bridge Manager api command and confirm that the two communicate directly bypassing the asterisk server (I confirmed this with Wireshar

[asterisk-users] Ring/Off-hook in strange state 6 channel X

Greetings List I have connected my asterisk box with x100 2xfxo and xorcom 8xfxo and all of them give me the error "Ring/Off-hook in strange state 6". Whenever the caller hangup, the call continue to execute until it hits the hard coded hangup. I changed chan_dadhi busydetect=no and callprogress=

Re: [asterisk-users] The sound is played but I did not hear

same result [?] 2008/11/25 Doug Lytle <[EMAIL PROTECTED]> > Try putting a Wait(1) before playing the sound effect. > > Doug > > -- > > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > ___