similar to: Wondering why I can't post

Displaying 20 results from an estimated 1000 matches similar to: "Wondering why I can't post"

2007 Aug 29
5
Ringing sound doesn't work
Hi, I have these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension
2007 Sep 10
5
Asterisk Manager API - Originate command
Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Oct 01
1
mISDN NPI setting with b410p
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've just received the following mail: ======================================================================= SETUP Q.931 Message - - CALLED PARTY NUMBER - -- NUMBERING PLAN IDENTIFICATION (Octet 3) - --- 0000 (Unknown) Full Octet = 80'h, you have set to 81'h. TYPE OF NUMBER should also be unknown which I have been told you have
2007 Aug 20
1
Application for Home Delivery Restaurants
Hello All We have developed an application for Home Delivery Restaurants using Asterisk, Java (JSP/ JSF) and MySQL. Here is listing of its features. If someone is interested then we can provide him more details. - POP up window with caller data containing his/her name, address and transactions history. - In case of new customer, Pop up window with blank form to add customer data and
2007 Sep 12
2
Conference bridge.
Any recommendations for an affordable SIP conference bridge unit? I mean one that isn't crappy; something where the duplex and cancellation functions that are traditionally built into such devices actually work. I am referring to something that looks like this . . . http://www.hardware.com/products/cnet/I212272.jpg But not necessarily that. -- Alex Balashov Evariste Systems Web :
2007 Sep 26
1
Manager Originate Action and Cancel
I'm using the Originate Action on the Asterisk Manager to place calls between two extensions in async mode. Is there any way to cancel the Originate Action before I get the OriginateResponse action? I'm unable to perform a Hangup because I can't know the channel name before I get the response... thanks in advance! -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel.
2007 Sep 26
2
SIP Panel?
Dear List, Has anyone found or written a status panel application, windows or linux, that uses SIP notifies and subscriptions, to gather the status of SIP extensions from Asterisk? And displsy nicely on a GUI? -- Terence C. Giufre-Sweetser Technical Support & Network Engineering SkyMesh Pty Ltd Licensed Telecommunications Carrier ABN 62 113 609 439 47 Baxter Street FORTITUDE VALLEY Q
2007 Sep 28
1
odd audio problem
I am having an odd audio problem. See setup diagram below. When a call comes in it get routed through the 1st asterisk box (currently running 1.2) through another asterisk box (running 1.4.11). All audio is good. When I upgraded the 1st asterisk box to 1.4.11. A call comes in, relays to the 2nd asterisk box. The AA answers the call and the audio is good. Once the call is forwarded to an
2007 Oct 16
1
Echoes & Asterisk connects too early
Hello, I have read the articles on echo cancellation (http://www.voip-info.org/wiki/view/Asterisk), but couldn't find a solution to my problem. We are running Asterisk 1.4.12 together with an EICON DIVA SERVER BRI-2M PCI (current driver from EICON) and some SNOM 300/360. There are few clients where we recognize echoes on both sides when we call them via ISDN. With some of these clients
2007 Oct 22
2
NAT traversal packet loss measurement
How can one measure the effect of NAT traversal packet loss? We currently have no solution for NAT traversal for our SIP clients. There is no doubt that packets are getting lost. What is not clear is how much damage this does. On the face of it, everything seems fine. Could this be so? Perhaps we're suffering a degradation in quality or our call setup times could be improved. How can we
2007 Sep 01
3
Zaptel modules are being installed in different directory
Hi: Iam running kernel is 2.6.8.1-12mdk but the modules of zaptel are being installed to /lib/modules/2.6.8.1-12mdkcustom how can i fix this up, any one have an idea? Best Regards; Wissam _________________________________________________________________ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://www.toolbar.live.com
2007 Aug 30
2
Unknown connection error: (2006) MySQL server has gone away
Hi, I get the following after a call has finished: ERROR[6862]:mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away Does this error message only appear when asterisk makes a new connection to mysql, because the old connection was stale (and dropped) ? If so, is there a way to get asterisk to stop reporting this as an error seeing it seems to write the CDR to
2007 Sep 12
3
Agent Callback Login in 1.4
Awhile back I had heard some talk, in this list I believe that Agent callback login was going to be deprecated in 1.4, I see it is still there. Does anyone know what is happening with this? -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 voip at rockynet.com
2007 Aug 30
3
Testing Framework
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, So, now that we've all complained about the state of testing of Open Source versions of Asterisk, lets do something about it. I propose we start with a list of things that we think should be tested in Asterisk, and means to test them. Maybe we could run certain tests based on the changes between minor versions? Anyway lets start. Call
2007 Aug 15
2
Load balancing SIP trunks?
I have 10 SIP trunks that I'd really like to round-robin load balance. Currently I have a macro that switches between available lines, but there really must be a function in Asterisk to do this on its own. So my question is just that, are there any easy ways for Asterisk to either balance between SIP trunks or even just a built in function to find the next available SIP trunk. I think using
2007 Aug 01
7
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.h - I cannot find that file: asterisk-dev:/home/hugh # find / -name 'autoconf.h'
2007 Oct 24
1
Asterisk Shutting Down
We've experienced the same problem twice now in the past month. The asterisk pid stops responding. We aren't able to connect to the CLI and all calls are dropped. The lots are pretty bare as well. This is the message log: Oct 24 09:12:35 WARNING[20711] channel.c: Avoided initial deadlock for '0x8444a70', 10 retries! Oct 24 09:12:35 WARNING[20711] channel.c: Avoided initial
2007 Oct 24
2
How to tune Asterisk AMD - Answering Machine Detection "hacks"
Hello Everyone, Can someone point me to reliable links on how to tweak Asterisk AMD I am calling a number and have to two files to play depending if it is a real person or an answering machine. Most everytime Asterisk calls it thinks it is an Answering Machine and it starts playing the AMD message, instead of the delivering the "1st real message" Any hints? Thanks in advance, -C
2007 Sep 17
7
Why does everyone seem to dislike *now?
Greetings, Last week I began researching Asterisk for the first time. I did what most noobs would do; downloaded an image that seemed simple and straightforward and had some credibility (*now). I also downloaded the TFOT version 1 as a guide. As questions arose, I tossed a few out in #asterisckNOW channel..and found it to be a ghost town. Only later did i start to ask a few
2007 May 21
2
MoH WAY too loud
Hi folks! I'm having a problem where my music on hold is just blaring to my callers. I've tried several different formats (converting using mpg123 and sox) and adjusted my musiconhold.conf to use quietmp3, to no avail. Every file plays way too loud. I did notice that sox has a -v flag for adjusting volume, but danged if I can find documentation online that'll tell me what