similar to: maximum simultaneous calls

Please remove this email from your mailing list. UNSUBSCRIBE Thank you. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Thursday, March 29, 2007 9:14 AM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 32, Issue 118 Send asterisk-users

Hi, is there a way to limit an account to do simultaneous calls in sip and iax? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070713/841cbb5f/attachment.htm

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

I am having difficulty configuring SIP behind NAT (using latest CVS). Using sip.conf: [general] port=5060 ; Port to bind to externip=ww.xx.yy.zz bindaddr=0.0.0.0 nat=yes register=>[userid]:[password]@voiptalk.org/2000 [voiptalk.org] nat=yes externip=ww.xx.yy.zz type=friend secret=[password] nat=yes reinvite=no canreinvite=no I fail to register. SIP Debug gives: SIP

Trying now to set up the final part of my * switch. I must admit I've had great fun over the last week or so playing with it, and would like to thank you guys for all the assistance (past and present, since I've been trawling a lot of old posts!!!). Scenario - using voiptalk.org to supply the incoming gateway, tied to an 0845 number for convenience in testing. Internal 7960 -> 7960

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded

I have several of incoming numbers on IAX from voiptalk and magrathea but have a problem with IAX.conf. If I follow the example from voiptalk [VoIPTalk Incoming Number] type=friend username=VoIPTalk Incoming Number context=[XXXXXXXX] and make incoming entries in IAX.conf for the numbers like below with a different entry for each number pointing to a different context, incoming numbers always

Why is it that newcomers always feel like inserting 'Answer' is a necessary step in their extension.conf entries? >[voiptalk.org] >;forwards any calls starting with an "8" thru voiptalk.org >exten => _8.,1,Answer >exten => _8.,3,SetCIDNum(55555555) >exten => _8.,4,SetCIDName(My Name And Surname) >exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g)

Hi! i compiled rpms like postfix with mysql support. My problem is when i compile programs i can't distinguish the difference between postfix w/o mysql and postfix w/ mysql because they have the same filename(postfix-2.x.x.i386.rpm), what i like to happen is when i compile postfix w/ mysql support it will show something like postfix-mysql-2.x.x.i386.rpm. so when i install and try to query it,

Hi Guys, Im having a problem compiling zaptel 1.2.4 on CentOS 4.3, anyone encountered this problem before? Here's the error i got: make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /usr/src/zaptel-1.2.4/zaptel.o /usr/src/zaptel-1.2.4/zaptel.c:384: error: syntax error before

Hi Guys, Im having a problem compiling zaptel 1.2.4 on CentOS 4.3, anyone encountered this problem before? Here's the error i got: make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /usr/src/zaptel-1.2.4/zaptel.o /usr/src/zaptel-1.2.4/zaptel.c:384: error: syntax error before

Hi list, i noticed from the cli my asterisk box is accepting unauthenticated calls how can i prevent this? CLI: -- Accepting UNAUTHENTICATED call from 192.168.0.2: > requested format = gsm, > requested prefs = (), > actual format = ulaw, > host prefs = (g729|ulaw|alaw), > priority = mine -------------- next part -------------- An HTML

Hi i have an asterisk pbx with E1 port connected to another PBX. Im trying to send the DNID/DNIS to the PBX here's my dialplan exten => 8881111111,1,Dial(ZAP/g2) exten => 8881111111,n,Hangup() The PBX just get the number 2 as it's DNIS when i change it to ZAP/1 or ZAP/g1 the PBX get the number 1. What should i add to send the extension number as DNID/DNIS? Thanks! --------------

what's a workaround for this? -bash: /bin/mv: Argument list too long -- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/register&r=19441 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20060615/6946660e/attachment-0002.html>

Is there a problem with apt-get repo? i get 404 not found when i try to update the repositories? im using rpm http://mirror.centos.org for my repositories and here's the error of apt. Ign http://mirror.centos.org centos/4.1/apt/i386 release Err http://mirror.centos.org centos/4.1/apt/i386/os pkglist 404 Not Found [IP: 64.92.174.74 <http://64.92.174.74> 80] Ign http://mirror.centos.org

Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as:

Hi, We're deploying a small VoIP solution for a group of teleworkers. Naturally, this exposes us to all sorts of fun, most of which we seem to have working properly. However, some NAT issues are still bugging us and we have noticed that often these situations didn't exist when users were connected directly to our VoIP provider, voiptalk.org. They have something which they call a

Hi, I use VoIPTalk as my provider and unsure of a minor issue. When people call me they get a US ring tone instead of UK. Is this a Asterisk configuration issue or one for VoIPTalk ? I am running 1.6.2.0 and IAX2 trunks. Thanks - Phil

I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have

Hi, I need to create a simple number checking for authorizing the calls. if a person dial 91800555121212345 where 12345 is the authorization code. If the authorization code is correct the call will go through if not it will play something saying wrong authorization code or just hangup. This my dialplan to get the authorization code AUTH=12345 exten => _9.,1,Answer() exten =>