[asterisk-users] How do you hangup a call without terminating your session?

Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap]

[asterisk-users] How do you hangup a call without terminating your session?

Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap]

[asterisk-users] How do you hangup a call without terminating your session?

Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap]

[asterisk-users] How do you hangup a call without terminating your session?

Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap]

[asterisk-users] How do you hangup a call without terminating your session?

Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap]

[asterisk-users] How do you hangup a call without terminating your session?

Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap]

Re: [asterisk-users] Press twice *

Depending upon what you want to do, you could add ** to features.conf e.g.: [featuremap] disconnect => ** Hugh 2010/6/4 Anahi Ludueña > Hi people, I need to detect when the user presses twice *... > In the dialplan I added the following, but it doesn't work. > Could you help me with that? > >

[asterisk-users] How do you hangup a call without terminating your session?

Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap]

[asterisk-users] Macros, GoSub & StackPop

Hi - I have a Macro that contains a GoTo. The documentation indicates: If you GoTo out of the Macro context, the Macro will terminate and control will return at the location refered to by the Goto. I thought I might convert the Macro to a GoSub routine, but the documentation doesn't mention wha

Re: [asterisk-users] Trouble getting feature codes to work

ng an IVR. Cheers! H On Fri, Jan 22, 2010 at 7:02 AM, Karsten Wemheuer wrote: > Hi, > > Am Donnerstag, den 21.01.2010, 21:08 -0500 schrieb hugolivude: > > Hi, > > > > I'm having trouble getting feature codes to work in Asterisk 1.4.21.2. > > Features.conf cont

Re: [asterisk-users] Trouble getting feature codes to work

sed to put this: include => featuremap in the context containing the Dial command right? Thanks in advance, H On Thu, Jan 21, 2010 at 9:31 PM, C. Chad Wallace < cwall...@lodgingcompany.com> wrote: > > At 9:08 PM on 21 Jan 2010, hugolivude wrote: > > > The call works fine

[asterisk-users] Trouble getting feature codes to work

Hi, I'm having trouble getting feature codes to work in Asterisk 1.4.21.2. Features.conf contians this: blindxfer=## atxfer=*2 automon=*1 disconnect=** I'm really most interested in getting disconnect to work so that I hear "Goodbye" when I press ** during a call connected this way in my dial pl

Re: [asterisk-users] Caller hang up not detected

) > > > > help this helps J > > > > Steven Davison > > Net Technial Solutions > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *hugolivude > *Sent:* 21 January 2010 13:47 > > *

[asterisk-users] Caller hang up not detected

Hi, I'm having trouble getting Dial to exit when the caller hangs up in Asterisk 1.4.21.2. I use a POTS line to call into the DiD given to me by VOIP service provider. When the call comes in, I have the VOIP provider send it to another POTS line. All this works fine however when the caller (me)

[asterisk-users] Feature codes not detected

Hi, I'm having trouble getting feature codes to work in Asterisk 1.4.21.2. Features.conf contians this: blindxfer=## atxfer=*2 automon=*1 disconnect=** I'm really most interested in getting disconnect to work so that I hear "Goodbye" when I press ** during a call connected this way in my dial pl

Re: [asterisk-users] Sendmail using SMTP authorization

sh -o /etc/mail/authinfo.db')dnl Don't forget to use smtp-rog.mail.yahoo.com in authinf and not smtp.broadband.rogers.com! Yours, H On Sun, Nov 9, 2008 at 12:15 PM, hugolivude <[EMAIL PROTECTED]> wrote: > > http://groups.google.ca/group/comp.mail.sendmail/browse_thr

Re: [asterisk-users] Sendmail using SMTP authorization

ing, and takes 30 seconds to setup. > > Thanks, > Matt G > > : http://www.voipphreak.ca > : http://www.ratemydialplan.com > : http://www.asterisk-jobs.com > >> -Original Message- >> From: [EMAIL PROTECTED] [mailto:asterisk-users- >> [EMAIL PROTECTED] O

[asterisk-users] Sendmail using SMTP authorization

Hi - OK not really an Asterisk question but it is affecting one of my favorite features - emailing voice mail! I've posted on some Linux forums and sendmail.org but no response so I'm hoping someone will take pity on me ;-) My ISP requires SMTP authorization and I'm having a heck of a time getti

Re: [asterisk-users] pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use

number I simply got another instance to run. Thanks again for your help. Howard On Thu, Sep 18, 2008 at 6:38 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Thu, Sep 18, 2008 at 05:31:08AM -0500, Anthony Messina wrote: >> On Wednesday 17 September 2008 09:18:58 pm hugolivude w

[asterisk-users] pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use

On Tue, Sep 16, 2008 at 2:21 AM, Michiel van Baak <[EMAIL PROTECTED]> wrote: > On 22:46, Mon 15 Sep 08, hugolivude wrote: >> I have two Asterisk servers running on the same LAN. One starts fine, >> but when I start the other I get: >> >>pbx_dundi.c:4582 load_

Re: [asterisk-users] pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use

On Tue, Sep 16, 2008 at 2:21 AM, Michiel van Baak <[EMAIL PROTECTED]> wrote: > On 22:46, Mon 15 Sep 08, hugolivude wrote: >> I have two Asterisk servers running on the same LAN. One starts fine, >> but when I start the other I get: >> >>pbx_dundi.c:4582 load_

[asterisk-users] Problems with 2 Asterisk servers on same LAN

OS = CentOS 5 Asterisk = 1.4.21 Router = WhiteRussian 0.9 Not sure whether I have a problem w/ Asterisk or White Russian config, so I'm posting to both lists. I have 2 Asterisk servers running behind a Linux router w/ White Russian. I'm having a lot of trouble with REGISTER. The servers are set

[asterisk-users] Realtime & sip.conf

Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=>:@/ and use realtime realtime (funny name!) for

[asterisk-users] sip.conf & realtime

Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=>:@/ and use realtime realtime (funny name!) for pee

Re: [asterisk-users] Couple installation questions

Once again guys thanks so much! Baji - Took a look at your instruction page. Thanks for putting that together. I've bookmarked it!! Howard On 8/1/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > On Wed, Aug 01, 2007 at 05:31:59PM -0400, hugolivude wrote: > > Hi, > &g

[asterisk-users] Couple installation questions

Hi, I'm installing * 1.4.9 and a couple questions have come up: 1) I read here( http://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x) that if you are using E1 cards you need to install LIBPRI. I'm not us

Re: [asterisk-users] Problems building zaptel 1.4.4

Just had to install the linux kernel source... All better now! Thanks for responding everyone! I have a few installation questions, but I'll post them in a separate thread. Hugh On 8/1/07, hugolivude <[EMAIL PROTECTED]> wrote: > asterisk-dev:/ # rpm -qa | grep kernel &g

Re: [asterisk-users] Problems building zaptel 1.4.4

asterisk-dev:/ # rpm -qa | grep kernel kernel-default-2.6.16.13-4 Thanks, Howard On 8/1/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Wed, Aug 01, 2007 at 06:41:42AM -0400, hugolivude wrote: > > > Right. autoconf.h is not necessarily generated. Please ignore that > >

Re: [asterisk-users] Problems building zaptel 1.4.4

> ./configure Works > make menuselect Works > make > make install Does not work. Errors already posted. > menuselect select the modules to compile > > do you have problems with this? > I can select modules but make fails. Thanks, Hugh ___ --Band

Re: [asterisk-users] Problems building zaptel 1.4.4

> Right. autoconf.h is not necessarily generated. Please ignore that > warning. To be fixed in upcoming versions of Zaptel. > > (This is not related in any way to GNU autoconf, that is used to > generate the ./configure script. The program autoconf itself is not > needed to build Zaptel, unless you

Re: [asterisk-users] Problems building zaptel 1.4.4

On 7/31/07, hugolivude <[EMAIL PROTECTED]> wrote: > ./configure works: > > configure:: *** Zaptel build successfully configured *** > > but make fails: > > make -C /lib/modules/2.6.16.13-4-default/build > SUBDIRS=/usr/src/packages/SOURCES/Asterisk/zaptel-1.4.4 modules

Re: [asterisk-users] Problems building zaptel 1.4.4

On 7/31/07, hugolivude <[EMAIL PROTECTED]> wrote: > That's right: > > find / -name 'autoconf.h' > > still comes up empty... > > > On 7/31/07, Matt Riddell <[EMAIL PROTECTED]> wrote: > > -BEGIN PGP SIGNED MESSAGE- > > Hash: SH

Re: [asterisk-users] Problems building zaptel 1.4.4

I should have also mentioned that I saw in another post that I may need to install autoconf: ftp://ftp.gnu.org/gnu/autoconf I downloaded this, did ./configure, make, make install but it did not fix my problem. I still cannot find autoconf.h... Thanks, H On 7/31/07, hugolivude <[EMAIL PROTEC

[asterisk-users] Problems building zaptel 1.4.4

Hi, I'm having trouble compiling zaptel 1.4.4 on SUSE 10.1. I'm really only interested in getting ztdummy to work because this is a dev machine with no zaptel h/w. SUSE 10.1 is a 2.6 kernel: asterisk-dev:/home/hugh # uname -r 2.6.16.13-4-default It seems that my problem is related to autoconf.

[asterisk-users] Problem with asterisk-addons - checking for mysql_init in -lmysqlclient... no

I'm trying to build the MySQL components in asterisk-addons but no luck so far. I hope that you can help. I have MySQL installed. rpm -qa indicates: MySQL-server-5.0.22-0 MySQL-devel-5.0.22-0 MySQL-client-5.0.22-0 rpm -ql MySQL-devel | grep client indicates: /usr/lib/mysql/libmysqlclient.a /usr

Re: [asterisk-users] MySQL components in asterisk-addons not being built

define HAVE_MEMORY_H 1 | #define HAVE_STRINGS_H 1 | #define HAVE_INTTYPES_H 1 | #define HAVE_STDINT_H 1 | #define HAVE_UNISTD_H 1 | #define HAVE_CURSES 1 | #define HAVE_NCURSES 1 | /* end confdefs.h. */ | | /* Override any GCC internal prototype to avoid an error. |Use char because int might match th

Re: [asterisk-users] MySQL components in asterisk-addons not being built

07, Nasir Iqbal <[EMAIL PROTECTED]> wrote: > Hi, > > please see your ./configure output especially few last lines. > > and note missing thins. > > Regards > > Nasir Iqbal > > On Tue, 2007-07-24 at 10:45 -0400, hugolivude wrote: > > Thanks Tahir. I already

Re: [asterisk-users] MySQL components in asterisk-addons not being built

com/pub/asterisk/releases/asterisk-addons-1.4.2.tar.gz > > 1. Extract them > 2. Make > 3. Make Install > 4. cdr_addon_mysql.so will be installed including all other modules. > > Regards > > Nasir Iqbal > > > On Tue, 2007-07-24 at 08:17 -0400, hugolivude wrote:

[asterisk-users] MySQL components in asterisk-addons not being built

I'm trying to add MySQL CDR recording in Asterisk 1.4.6. I'm following the instructions posted here: http://www.voip-info.org/wiki-Asterisk+cdr+mysql I have MySQL installed and it works fine - starts on stratup, I can create DBs, tables and so on and I can connect through php. rpm -qa indicates

[asterisk-users] cdr_addon_mysql.so is not created

I'm trying to add MySQL CDR recording in Asterisk 1.4.6 I'm following the instructions posted here: http://www.voip-info.org/wiki-Asterisk+cdr+mysql but after running make, the cdr_addon_mysql.so is not created. I don't get any compile errors. In fact it just seems to skip the compile altogth

Re: [asterisk-users] Installing AJAM

8088/asterisk/mxml?action=status&username=foo&secret=bar you get the same error message. I also found that the demo works now: http://localhost:8088/asterisk/static/ajamdemo.html Cheers, H On 7/2/07, hugolivude <[EMAIL PROTECTED]> wrote: > I discoved a real problem this time! &g

Re: [asterisk-users] Installing AJAM

) but there is no rawman installed. Any idea how to fix this? Hugh On 7/2/07, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote: > or copy /usr/src/asterisk-1.4.6/configs/http.conf.sample /etc/asterisk > and edit, because make samples I believe wipes out existing configs >

[asterisk-users] Installing AJAM

Hi, I just installed Asterisk 1.4.6. I didn't see http.conf in /etc/asterisk. Is there a seperate install for AJAM? I dug around a little and found only _one_ reference that refers to installation of AJAM: http://astrecipes.net/?n=217 In accordance with the instructions on this site I perfor

Re: [asterisk-users] Fwd: Problems with zap following 1.4.6 install

ight track!! H On 6/30/07, hugolivude <[EMAIL PROTECTED]> wrote: > Thanks for responding Russell! > > > What output do you get if you run "module unload chan_zap.so" > > == Unregistered application 'ZapSendKeypadFacility' > > > and then &q

Re: [asterisk-users] Fwd: Problems with zap following 1.4.6 install

Thanks for responding Russell! > What output do you get if you run "module unload chan_zap.so" == Unregistered application 'ZapSendKeypadFacility' > and then "module load chan_zap.so" ? == Registered application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found [Jun 30

[asterisk-users] Fwd: Problems with zap following 1.4.6 install

Hi, Just upgrading to 1.4.6 from 1.2. SIP channels work OK, but not zap. I have a TDM400 w/ an FXO & 2 FXS. I built libpri 1.4.0 first then zaptel 1.4.3. Menuselect had a * beside chan_zap and I loaded the wcusb & wctdm before building asterisk. In the CLI "zap show channels" returns "no suc

[asterisk-users] SIP RE-INVITE after an Answer()

Hi, I managed to get SIP re-invite working. If a call comes into my * box from my ITSP on a DiD, I can handle the call by calling Dial() in my dial plan and the call will get transferred and the media does not pass through my * box after the call is bridged. However, if I Answer() the call befor

[asterisk-users] SIP Re-Invite behind a NAT

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to

[asterisk-users] Fwd: Trouble using 2 IAX2 DiDs provided by different ITSPs

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both

[asterisk-users] Trouble using 2 IAX2 DiDs provided by different ITSPs

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both

[asterisk-users] Trouble using 2 IAX2 DiDs provided by different ITSPs

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both

[asterisk-users] Trouble using 2 IAX2 DiDs provided by different ITSPs

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both

[asterisk-users] Bad Voice Quality - IAX2 redirect

Asterisk 1.2.7 RedHat 9.0 Hi, I've run into some voice degradation problems with IAX2: I frequently have calls come in on a DiD provided by an ITSP. I often have to redirect these calls back out to the PSTN (i.e. to a cell phone). When this happens, I don't want my server in the media path, I

Re: [asterisk-users] Hairpinning problems using IAX2 and SIP

use notransfer=yes and it worked perfectly, even behind NAT. > > Also you should contact your ITSP, maybe they don't allow this? > > > > On 11/5/06, hugolivude < [EMAIL PROTECTED] > wrote: > > Thanks for responding. > > > > Yes I am doing pretty much e

Re: [asterisk-users] Hairpinning problems using IAX2 and SIP

7; -- Hungup 'IAX2/-1' On 11/4/06, Andrew Joakimsen <[EMAIL PROTECTED]> wrote: When you say you answer the call, I assume you have something like this: exten => 5551212,1,Answer exten => 5551212,1,Dial(SIP/provider/10005551212) Try to not answer the call and see

Re: [asterisk-users] Call Quality Issues with IAX?

Funny you mention this because I've run into some voice degradation problems with IAX2 myself recently... When I have an external call come in on a DiD I frequently have to send it back out to the PSTN (i.e. to a cell phone). When this happens I don't want my server in the media path, I want to

[asterisk-users] Redirect problems using IAX2 and SIP

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is refe

[asterisk-users] Redirect problems using IAX2 and SIP

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is refe

[asterisk-users] Hairpinning problems using IAX2 and SIP

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is refe

Re: [asterisk-users] Understanding NAT Traversal

Thanks Moj! The RTP packet problem makes sense. Still unclear on some of the other points: I think the biggest problem with SIP is the RTP ports. The initial SIP request goes out (for example) to port 5060, and FROM port 5060 as well. The response needs to get back to the SIP UA on that por

[asterisk-users] Understanding NAT Traversal

Quick question re. NAT traversal.  I understand how sitting behind a NAT could cause problems for a SIP UA.  The SIP UA would create SIP mesages using IP addresses from inside the network (i.e. 192.#.#.# or 10.#.#.#) and these IP addresses are of course unnavigable for the recipient. What I don'

Re: [asterisk-users] Bandwidth requirements

So much detail!  Thanks very much guys, I'm sure that all this excellent info will be valuable to others as well. Gratefully yours, HOn 10/5/06, Brian Candler <[EMAIL PROTECTED]> wrote: On Thu, Oct 05, 2006 at 10:53:14AM +0200, raphael Jacquot wrote:> Brian Candler wrote:>> >However on ADSL, you h

[asterisk-users] Bandwidth requirements

Hi, Age old question it seems but I haven't been able to get a handle on it yet.  Let's assume I'm using a g729 codec.  If I wanted to handle 20 simultaneous calls, how much bandwidth would I need?  Is there a general formula for this? I tried this caluclator: http://www.voip-calculator.com/calc

[asterisk-users] Bandwidth requirements

Hi,Age old question it seems but I haven't been able to get a handle on it yet.  Let's assume I'm using a g729 codec.  If I wanted to handle 20 simultaneous calls, how much bandwidth would I need?  Is there a general formula for this?I tried this caluclator: http://www.voip-calculator.com/calculato

Re: [asterisk-users] caller id problem

You cannot set callerid on POTs lines. You my have more luck if you place your call via a T1 - but it's still up to your carrier. Some VoIP providrs also allow you to set callerid on SIP calls, but you need to check. I fear you'll have a hard time finding a carrier that will allow you to set CA

Re: [asterisk-users] Need help

I have these DID's coming to my asterisk box and I have a dial plan and all I want is that it should ring on my cell phone for 20 seconds and if I don't pickup it would then ring on my home phone. It does the ringing and goes to the next phone in 20 seconds and every thing works fine but If I pick

Re: [asterisk-users] VoiceMail being cutoff when leaving message

Are the callers that are cut off "low talkers"? Try dialling _down_ the silencethreshold value. 128 is the default, but I was having people get cut off during voicemails so I dialled it down to 64. A "low talker" called today and successfully left a message without being cut off. silencethres

Re: [asterisk-users] Cannot dial out through SIP provider

Woops, sorry the first part of my response is wrong: Shouldn't the line: exten => _,1,Dial(SIP/[EMAIL PROTECTED],,) be: exten => _,1,Dial(SIP/[EMAIL PROTECTED],,) note the .dk in the second one... What I said here is incorrect, looks to me you have it right. You may still

Re: [asterisk-users] Cannot dial out through SIP provider

Shouldn't the line: exten => _,1,Dial(SIP/[EMAIL PROTECTED],,) be: exten => _,1,Dial(SIP/[EMAIL PROTECTED],,) note the .dk in the second one... Also I don't see a register line in your sip.conf. In the [general] section I would have expected something like: register=>:@musim

Re: [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution

Note that you have: [teliax] context=default but you do not have a "default" context in extensions.conf for this. Change the above to: [teliax] context=general **OR** in extensions.conf change [general] exten => 3031234567, 1, Answer() exten => 3031234567, 2, Dial(SIP/105,15) t

Re: [asterisk-users] Correct syntax for Set(CALLERID(all)...

uot;, "ANI", "DNID", "RDNIS". > > In any case you were just told in the original post that there is one, > how can you lie and say you don't know? > > > On 8/10/06, Don <[EMAIL PROTECTED]> wrote: > > dunno that there is an "all&

Re: [asterisk-users] Correct syntax for Set(CALLERID(all)...

the original post that there is one, > > how can you lie and say you don't know? > > > > > > On 8/10/06, Don <[EMAIL PROTECTED]> wrote: > > > dunno that there is an "all" datatype is there? > > > > > > Set(CALLERID(name)=John Doe) > > &g

[asterisk-users] Correct syntax for Set(CALLERID(all)...

Asteris 1.2.7 Redhat 9 Hi, Is this the correct syntax for setting CALLERID: exten => _1,n,Set(CALLERID(all)=The Smiths <212-876-9345>) I'm able to get the number to change but the name is always "Unknown Name". I've tried numerous combinations of quotes, but just cannot get the name... Thank

Re: [asterisk-users] Setting CALLERID on a residential telco line

That's what I feared. I could do it if I had a T1 is that right? Thanks, H On 8/4/06, Steven Ringwald <[EMAIL PROTECTED]> wrote: hugolivude wrote: > Redhat 9 > Asterisk - 1.2.7 > TDM 400 - 1 FXO, 2 FXS > > I'm using a standard residential PSTN line on my ZAP

[asterisk-users] Setting CALLERID on a residential telco line

Redhat 9 Asterisk - 1.2.7 TDM 400 - 1 FXO, 2 FXS I'm using a standard residential PSTN line on my ZAP channel and curious whether I can override the caller ID my telco has for me with one of my choosing. I've tried this: exten => s-ZAP,n,Set(CALLERID(all)=My Name <999-999-999>) exten => s-ZAP,n

[asterisk-users] Help troubleshooting "deadlocked" Asterisk

I'm having a heck of a time keeping my Astertisk box up an running: Redhat 9 Asterisk 1.2.7.1 Digium TDM400 w/ 1 FXO + 2FXS 1 g729 codec I have my Sip.conf set up to renew registrations every hour: maxexpirey=3600 defaultexpirey=3600 When I look in the /var/log/asterisk/messages tho, I se

[Asterisk-Users] zt hook failed

Asterisk 1.2.7.1, Red Hat 9.0, TDM 400 2 FXS, 1 FXO I'm getting the following warning from time to time: "zt hook failed: Device or resource busy" It seems that once I get this error I can no longer use my zap channel. Interestingly it seems to affect SIP as well as I can no longer dial ou

[Asterisk-Users] Re: Analogue phone w/ TDM400

No definitely not. These tones are generated by the phone, not Asterisk. H On 5/28/06, T.S <[EMAIL PROTECTED]> wrote: Sure that's not the message waiting "stuttering" indicator? Terrelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On

[Asterisk-Users] Analogue phone w/ TDM400

Hi, I'm running * 1.2.7.1 on Red Hay 9.0 w/ a TDM 400 2 x FXS, 1 x FXO. I'm using a VTech cordless that makes three short beeps when someone another extension is picked up, presumably this lets you know if someone is trying to listen in.. Everything works, except the VTech now makes the three be

Re: [Asterisk-Users] Silent Attendant

great stuff guys, thanks a lot!! H On 5/6/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: Wes Baehr wrote: > Simply generate or record a 5-second sample of ringing. Then use > Background() to play that ringing file - if someone presses 9, they will be > routed accordingly, or otherwise se

[Asterisk-Users] Silent Attendant

I'd like to set up a "silent attendant". By this I mean when someone calls me I'd like for them to hear the comfort ringing tones, but for the first 5 seconds I'd like to give them the option of pressing 9 to send the call to an alternate extension; if they don't press 9, the call goes to a defa

[Asterisk-Users] REGISTER that isn't a register

Anyone come across this message: WARNING[2203]: chan_sip.c:9633 handle_response_register: Got 200 OK on REGISTER that isn't a register I get it from time to time but don't know what to do about it! Thanks, h ___ --Bandwidth and Colocation provided by

[Asterisk-Users] Re: Running Asterisk as non-root

Oh boy - thanks. I use g729... H On 5/5/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: If you run Asterisk as non root, you may have problems installing G729 licenses. The digium registration utility has certain hard coded stuff, and doesn't behave well when things aren't installed in the sta

[Asterisk-Users] Re: Running Asterisk as non-root

Thanks for the swift response Luki. Have you checked out: http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root recently? Does it seem up to date to you? It indicates it was updated in March of this tear, but I did this once and don't want to go thru all the hassle if the directions

[Asterisk-Users] Running Asterisk as non-root

I saw where one should not run Asterisk as root: http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root How important is this? Thanks, just curious whether it's worth the trouble. H ___ --Bandwidth and Colocation provided by Easynews.com --

[Asterisk-Users] Insights on SIP channel usage in * 1.2.7.1 are welcome!

I've had a heck of a time getting a SIP channel to work in Asterisk 1.2.7.1 (Redhat 9.0). I've done it successfully a number of times on pre 1.2 versions of Asterisk so perhaps it's version related. Any insights are welcome! At first I wasn't able to dial out on the SIP channel _whenever_ I sta

[Asterisk-Users] Re: Hi...Please help me

ant. Howard On 5/2/06, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: On Tuesday 02 May 2006 16:42, hugolivude wrote: > We share SIP phones at the office in a 1:4 ratio. You're probably > asking – how do you know when a ringing phone is for you? Well, > everyone in our offi

Re: [Asterisk-Users] Hi...Please help me

First off, I agree w/ Gonzalo – softphones didn't work out for me either. One thing that did work great tho was a combo. We share SIP phones at the office in a 1:4 ratio. You're probably asking – how do you know when a ringing phone is for you? Well, everyone in our office gets an XLite softph

[Asterisk-Users] WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '????'

Hi, Red Hat 9.0 Asterisk 1.2.7.1 ** Apologies if you notice this posted multiple times, I'm just not seeing it on the boards ** Whenever I start Asterisk, I am unable to call out on my SIP channel: -- Executing Dial("Zap/1-1", "SIP/[EMAIL PROTECTED]||t|") in new stack Apr 30 11:02:00 WARNING

[Asterisk-Users] WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '????'

Hi, Red Hat 9.0 Asterisk 1.2.7.1 Whenever I start Asterisk, I am unable to call out on my SIP channel: -- Executing Dial("Zap/1-1", "SIP/[EMAIL PROTECTED]||t|") in new stack Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No such host: 6477235412 Apr 30 11:02:01 NOTICE[12814]: a

[Asterisk-Users] WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '????'

Hi, Red Hat 9.0 Asterisk 1.2.7.1 Whenever I start Asterisk, I am unable to call out on my SIP channel: -- Executing Dial("Zap/1-1", "SIP/[EMAIL PROTECTED]||t|") in new stack Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No such host: 6477235412 Apr 30 11:02:01 NOTICE[12814]: a

[Asterisk-Users] Intermittent problem dialling out on a SIP channel

Hi, Red Hat 9.0 Asterisk 1.2.7.1 I'm having a bit of an intermittent problem with my SIP account. Often (but not always) when I start * or RELOAD my dial plan from the CLI I get this message: Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822 add_realm_authentication: Format for >authenticati

[Asterisk-Users] Re: Pattern matching problem

Sorry meant to say thanks Andrew!! I appreciated your comments too Kevin ;-) Oy no wonder i have problems w/ dial plans H On 4/26/06, hugolivude <[EMAIL PROTECTED]> wrote: > Thanks Kevin, your explanation resonated better with me. Sorry for > doubting you Eric ;-) > > S

[Asterisk-Users] Re: Pattern matching problem

D]> wrote: > Hi, > > Same dial pattern on my extension.conf, But it's work great. The Asterisk > only match 7 digits number. My * version is 2.1.6. > > Kevin > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of hugo

[Asterisk-Users] Re: Pattern matching problem

this case) appears broken to me. H On 4/26/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: > This is just the way Asterisk works. > > hugolivude wrote: > > Thanks, but the problem's with the first extension: > > > > exten => _NXX,1,NoOp(

[Asterisk-Users] Re: Pattern matching problem

, > Background, etc) and not when dialing from a phone. > > BTW, this works just like the Telco. You can dial as many extra digits > as you want, and the telco will ignore the extra ones, which is why you > can dial 1-800-PROGRESSIVE it will work (assuming such a number exists). >

[Asterisk-Users] Pattern matching problem

I'm running Asterisk 1.2.7.1 on Red hat 9 and have a strange pattern matching problem: I have the following in my dial plan: exten => _NXX,1,NoOp(Number dialed ${EXTEN}) exten => _NXX,n,Dial(Zap/1/${EXTEN}) Unless I'm missing something, I wouldn't expect the pattern above to match a 10

[Asterisk-Users] Re: Some questions re. T1 cards & QoS

On 4/24/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: > hugolivude wrote: > > > 1) Will I need a digital or analogue interface card? I expect digital > > is the answer, but the Digium web site said something about analogue > > cards being able to support "pro

[Asterisk-Users] Some questions re. T1 cards & QoS

I've been asked to assess the cost of implementing Asterisk with a single T1 line in one of our offices. I've had plenty of experience w/ TDM400 cards, but T1 is new for me so a couple of questions: 1) Will I need a digital or analogue interface card? I expect digital is the answer, but the

[Asterisk-Users] Re: no audio

Wierd timing - I'm struggling with exactly the same issue. My problem was with ZAP - ZAP. The phones ring, but no audio. Turns out there's a bug with the version I'm running. It has to do w/ the system date. When I changed my system date to 1-Jan-06, everything worked!! Here's what I found fr

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