[Asterisk-Users] IAX2 Problem and Question
Title: IAX2 Problem and Question Dear Asterisk Users. I have been setting up IAX between two servers, one in the USA and the other in UK so that I can pass help desk and general calls from one call center to another. I seem to be having an issue. When I set up IAX between my two servers I get into trouble when doing a reload on the CLI. Almost as if the system had gone into a loop reading the configuration. I also have a question, if I use the switch command i.e. switch = IAX2/brunswick:[EMAIL PROTECTED]/sip and switch = IAX2/dornoch:[EMAIL PROTECTED]/sip can I point to the same context? Or does the context on each pbx need to be unique. Thanks and Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney NSW 2089
[Asterisk-Users] sip no sound?
Good day all So I've installed asterisk with my openline4 card and I've setup sip and I can do sip on the local network,we are using soft clients,x-lite. But... When a call comes in from the outside(PSTN) and the you dial the extension it forwards the call the the client and you see incoming call on x-lite,you accept he callBUT there is no sound.It shows there is a call and you are connected but there is no sound Any Idea Please Help Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * INSTRUCTIONS FOR NEW MEMBERS OF THE COMMUNITY * Please read
Welcome to the Asterisk users community! It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. ** The mailing list is growing The lead programmer of Asterisk, Mark Spencer at Digium, inc, writes: The Asterisk community is growing at a remarkable pace. I know there are thousands of you out there -- in fact there are over eight *thousand* subscribers to asterisk-users alone, and almost one *thousand* registered users on the bug tracker. This means that everything anyone write to this mailing list, is sent to over 8.000 mailboxes that is already flowing over with messages. I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org project is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list. You'll find it on http://lists.digium.com, which is the address where you manage your subscription to this list as well. ** Reporting bugs If you think you have found a bug, report it. We need bug reports. Read this document http://www.digium.com/bugtracker.html and then go to the bugtracker http://bugs.digium.com to file a report. If you are unsure, find a bug marshal on the IRC channel to help you. They're appointed to support you with how to handle bugs. The bug tracker is also a place where you add your contribution to Asterisk. If you have coded extra functionality, make sure you give it back to the project so it can be added to the code base. This is how Asterisk grows, free contributions and consultants that are paid to add functionality on a case by case basis. ** Remember: It's Open Source, it's volountary Asterisk.org is a Open Source project. This means you can't request help from people, demand new functions or support. However, there are many individuals and companies out there that are offering services based on Asterisk, from VoIP service providers to consultants all over the world. Of course, this is also part of Digiums business, so you have plenty of help if your willing to pay. Digium is to be found at http://www.digium.com. Service providers and consultants are listed on the wiki, where you'll find companies all over the globe that are willing to set up your PBX and get you connected to either the PSTN or the growing telephony network on the Internet. Again, welcome to the Asterisk.org Open Source PBX Project! Meet you on the IRC channel :-) /oej - PS. This message will be sent regurlarly. If you have any corrections or additional information that needs to be included, mail me * off list *. Thank you! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Call Queuing App?
Title: Message here you go :) http://bugs.digium.com/bug_view_page.php?bug_id=214 Ta SJ -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric KirklandSent: 05 April 2004 04:25To: [EMAIL PROTECTED]Subject: [Asterisk-Users] New Call Queuing App? I thought I saw online in a list somewhere that theres an improved Call Queuing app out; supposedly it has the capability to tell the caller how far down the queue they are, etc? I saw one post about it somewhere but then no mention of it anywhere else Andy, [EMAIL PROTECTED] ---Incoming mail is certified Virus Free.Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.648 / Virus Database: 415 - Release Date: 3/31/2004 ---Outgoing mail is certified Virus Free.Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.648 / Virus Database: 415 - Release Date: 3/31/2004
RE: [Asterisk-Users] Unabled to exit console
Use the shell command ! to exit to shell. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Parlee Sent: Sunday, April 04, 2004 3:56 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Unabled to exit console No matter what I try, Asterisk won't let me out of the console. If I CTRL+C, of course, the process will terminate. I started asterisk like so: /usr/sbin/asterisk -gc and here's what I get when I 'exit': *CLI exit The QUIT and EXIT commands may no longer be used to shutdown the PBX. Please use STOP NOW instead, if you wish to shutdown the PBX. *CLI *CLI CTRL+D doesn't work either. What's going on? I am getting a bunch of these: Apr 3 20:06:08 WARNING[6151]: chan_sip.c:5632 sipsock_read: Recv error: Resource temporarily unavailable Apr 3 20:06:08 WARNING[6151]: chan_sip.c:5632 sipsock_read: Recv error: Resource temporarily unavailable Apr 3 20:06:09 WARNING[6151]: chan_sip.c:5632 sipsock_read: Recv error: Resource temporarily unavailable Apr 3 20:06:10 WARNING[6151]: chan_sip.c:5632 sipsock_read: Recv error: Resource temporarily unavailable Apr 3 20:06:11 WARNING[6151]: chan_sip.c:5632 sipsock_read: Recv error: Resource temporarily unavailable Apr 3 20:06:15 WARNING[6151]: chan_sip.c:5632 sipsock_read: Recv error: Resource temporarily unavailable Here's what I get wtih CTRL+D *CLI *CLI ^D Use STOP NOW to shutdown Asterisk *CLI Thanks in advance! -Ryan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 3-4 port FXO card recommendations
On Wed, 31 Mar 2004, Senad Jordanovic wrote: Angus Berry wrote: A quick search on eBay turned up this 4 port FXO external box for US$299: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3087347715category=51279 ...anyone know if it's compatible with Asterisk? Yes.. I can confirm I had it setup and it is working great. asterisk can use it to receive calls from PSTN and dial out to the PSTN? product description of the webswitch 100 G4 makes it sound like it's just a device to interface legacy PBX FXS interfaces to VOIP. If I can receive calls from PSTN into webswitch 100 G4 and route it (h.323) into PC with asterisk, that would be perfect. -Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gnophone installation problems
Fran Boon wrote: Gavin Hamill wrote: I'm using Mozilla 1.7a installed from a tarball. The needed libraries are just there: You've answered your own question. You installed Mozilla from a tarball. RPM therefore doesn't know about it. You need to install a recent Mozilla RPM :) Why do I need to install from RPM when I already included the Mozilla lib directories in /etc/ld.so.conf and issued a 'ldconfig' command? The system should know where to look for the needed libraries already... or use --nodeps F That wasn't a good move either: --- gnophone Registering Enlightened Sound version 0 Loaded and activated '/usr/lib/gnophone/modules/audio-esd.so' New input space: 0 of 40 64 byte fragments (0 bytes left) New output space: 40 of 40 64 byte fragments (2560 bytes left) Registering ALI 5451 (DUPLEX) on /dev/dsp0 Loaded and activated '/usr/lib/gnophone/modules/audio-oss.so' Registering Mozilla/5.0 Loaded and activated '/usr/lib/gnophone/modules/html-mozilla.so' Loaded and activated '/usr/lib/gnophone/modules/audio-phone.so' iax.c line 654 in iax_init: Started on port 5036 Listening on port 5036 Initialized phone core New input space: 0 of 40 64 byte fragments (0 bytes left) New output space: 40 of 40 64 byte fragments (2560 bytes left) Segmentation fault No bytes to read Error reading voice data on ALI 5451 (DUPLEX) on /dev/dsp0 --- Any ideas now? TIA, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?
Brian Cuthie wrote: Let's say that I have a call coming in to Asterisk through a TDM400P and going out through SIP to someone on the Internet. Is there any configuration option that would allow me to do silence suppression on the RTP stream generated by Asterisk on behalf of the TDM400P connected user? SIP phones allow me to do this easily, but I'd like to be able to conserve upstream bandwidth on calls that don't emanate from a SIP phone here at my location. Asterisk SIP does not support silence suppression. In fact, using Silence suppression on an inbound RTP stream will lead to problems, since Asterisk takes timing from inbound RTP streams. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unabled to exit console
jc wrote: Use the shell command ! to exit to shell. Use screen... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Registration Errors
Larry Keyes wrote: Hi...I've got two Grandstream phones attached to my Asterisk on the same subnet. The phones have fixed IP addresses. Asterisk is generated an error for one of them only, even though both appear to be registered correctly. The current state of the sip.conf is included below. Anyone know what is going on here? Both appear to be working fine between each other and between themselves in and outbound to an X100p card. Any ideas regarding the config file would be appreciated. -- Larry NOTICE[1125350192]: File chan_sip.c, Line 5297 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.0.162' NOTICE[1125350192]: File chan_sip.c, Line 3557 (register_verify): Peer '1001' isn't dynamic Read what it says. Peer '1001' is defined as a fixed IP address, not dynamic. So it is not allowed to register. The host= setting defines how we're going to contact the peer when we want to deliver a call to the phone. host=dynamic - Make the device register with asterisk so we know the current IP address host=ip address - No registration, we already know the IP address and the address doesn't change. For mobile devices, like soft phones on a laptop, use registration. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on FreeBSD
Hello, Asterisk in FreeBSD ports is currently FORBIDDEN due to security issues raised in pwlib (H323). As I just want to test Asterisk internally at this point I commented out the FORBIDDENs and compiled it with no problems. Unfortunately though, I can't seem to get any SIP softphones to register with Asterisk. I have tried SJPhone on Windows and KPhone and Linphone under FreeBSD. At the Asterisk console I've turned the sip debug on, but don't see anything at all. (no SIP traffic). I have followed the quickstart quide and configured things in sip.conf, but with no success. At this stage, and someone please confirm that Asterisk is really working on FreeBSD ? :) The version I installed is 0.7.2, running on FreeBSD 4.7. many thanks, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
Richard Airlie wrote: At this stage, and someone please confirm that Asterisk is really working on FreeBSD ? :) Yes, it's working with some limitations. See http://www.voip-info.org/tiki-index.php?page=Asterisk%20freebsd /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Brian Cuthie wrote: Let's say that I have a call coming in to Asterisk through a TDM400P and going out through SIP to someone on the Internet. Is there any configuration option that would allow me to do silence suppression on the RTP stream generated by Asterisk on behalf of the TDM400P connected user? SIP phones allow me to do this easily, but I'd like to be able to conserve upstream bandwidth on calls that don't emanate from a SIP phone here at my location. Asterisk SIP does not support silence suppression. In fact, using Silence suppression on an inbound RTP stream will lead to problems, since Asterisk takes timing from inbound RTP streams. Yeah, funny thing is I saw this problem just last night while messing around with music on hold. I had VAD on the SIP phone on and the MOH would stop unless I talked. I thought it was quite weird when it happened; now it makes sense. I've heard that Asterisk derives its timing in strange ways, but I've been wondering why it doesn't use the machine's clock (real-time interrupt, not wall-clock). -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gnophone installation problems
On Mon, 5 Apr 2004, Martin Mielke wrote: Why do I need to install from RPM when I already included the Mozilla lib directories in /etc/ld.so.conf and issued a 'ldconfig' command? The system should know where to look for the needed libraries already... The system might (depending on how you define the system), but RPM definitely does not. RPM does not check the filesystem for what may or may not be installed. It has a database for that purpose, which is updated when you install/remove packages. If you installed something, and did not use an RPM to do so, RPM does not know anything about the components that you installed. This is RPM 101. Somebody else wrote: or use --nodeps And Martin replied: That wasn't a good move either: I'm not surprised. Using --nodeps on an RPM package install is just plain wrong -- any software that requires it to install is broken. Run (don't walk) from any maintainer that tells you to use it to install their package. This (IMHO) should also be in RPM 101. :) Your ways forward in this case are: 1) Find an RPM of Mozilla to suit your distro (accepting that you might not get the version of Mozilla that you want), then try re-installing the gnophone RPM 2) Use a suitable .spec file to build an RPM of Mozilla to suit your distro (if you really must have a version of Mozilla for which there is no existing RPM), then try re-installing the gnophone RPM 3) Build and install gnophone from source (if you want to keep your existing Mozilla install) I'd suggest taking further questions on this away from this list, as it is really not on-topic for Asterisk... Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote: Richard Airlie wrote: At this stage, and someone please confirm that Asterisk is really working on FreeBSD ? :) Yes, it's working with some limitations. See http://www.voip-info.org/tiki-index.php?page=Asterisk%20freebsd Thanks for that, good to know. And now leads me to ask... why should my SIP softphones be unable to register? They are on the same subnet as asterisk. If i have sip debug turned on, shouldn't I at least be seeing some action on the Asterisk console when they try to register? thanks, Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] The maximum capacity of MeetMe
Hi !! I know that a conference room can be made infinitely. but, I think that there is actually a limit. For example, how many conference rooms can be made from CPU 866 [MHz] and RAM 256 [MB]? Is there any person who tried someone? I am studying MeetMe now. Please tell me a hint!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Redhat 9 OVER, Fidora Support, comments please.
Hi *ers, I recently got an Email from Redhat about the dropping of support for Redhat 9 on the 30 of April and that Fedora Project is the recommended future, otherwise, RedHat enterprise ($$$). Considering this, I would like some feed back on the Fedora Project from users who may be using it, and how its going with Asterisk? Are there any problems? Is the Asterisk development team got Fedora Project in mind and fully supported? Thanks, James Gardiner -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fran Boon Sent: Saturday, 3 April 2004 1:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel Nicolas Gudino wrote: http://sip.house.com.ar/operator Hi Nicholas, Agree with the other feedback - looks beautiful, the auto-refreshes are exceedingly smooth...definitely vindicates using Flash for client-side :) I also agree that more buttons would be very useful. (Although some of my labels get cut-off as-is, so I'd like a slightly smaller font even with current size) In fact I'll have so many that I think what I really want is the option to group them into different folders - ideally the user could even create their own folder! Aside from this, I note that the webpage states See at an glance: SIP registration status and reachability How does this work? I can't see any difference on my system between registered unregistered clients (makes a big difference for SoftPhones). I'd also like to have an option to disable the 'Talking to' part - in some situations this might be undesirable. Thanks a lot for the contribution - I would urge you to continue further :) Best Wishes, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
Richard Airlie wrote: On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote: Richard Airlie wrote: At this stage, and someone please confirm that Asterisk is really working on FreeBSD ? :) Yes, it's working with some limitations. See http://www.voip-info.org/tiki-index.php?page=Asterisk%20freebsd Thanks for that, good to know. And now leads me to ask... why should my SIP softphones be unable to register? They are on the same subnet as asterisk. If i have sip debug turned on, shouldn't I at least be seeing some action on the Asterisk console when they try to register? Turn on SIP debug and you'll be able to see what happens. Check with sockstat -l if Asterisk is listening to port 5060. Also, make sure you start asterisk with a lot of -v to get debug output. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?
At 8:34 AM -0400 on 4/5/04, Brian Cuthie wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Brian Cuthie wrote: Let's say that I have a call coming in to Asterisk through a TDM400P and going out through SIP to someone on the Internet. Is there any configuration option that would allow me to do silence suppression on the RTP stream generated by Asterisk on behalf of the TDM400P connected user? SIP phones allow me to do this easily, but I'd like to be able to conserve upstream bandwidth on calls that don't emanate from a SIP phone here at my location. Asterisk SIP does not support silence suppression. In fact, using Silence suppression on an inbound RTP stream will lead to problems, since Asterisk takes timing from inbound RTP streams. Yeah, funny thing is I saw this problem just last night while messing around with music on hold. I had VAD on the SIP phone on and the MOH would stop unless I talked. I thought it was quite weird when it happened; now it makes sense. I've heard that Asterisk derives its timing in strange ways, but I've been wondering why it doesn't use the machine's clock (real-time interrupt, not wall-clock). -brian Interestingly enough, Mark and I talked about this problem very briefly at dinner the other night. My recollection is that he seemed to think that taking timing from a Zap driver would be feasible, but there were many other things to do ahead of time. Perhaps others can program this or encourage it's development. Personally, I think VAD is a great service, as well as comfort noise generation to disguise when VAD is working. I'll always encourage methods that reduce bandwidth. Most major developers on Asterisk consider these technologies of low concern since their bandwidth is unlimited, as they typically sit in a co-lo somewhere (as many programmers of * are providers of service, not consumers.) The reality for most end users is that they are on very restricted pipes that are delivered via a WAN technology (especially for outbound, if you consider residential) and being able to put more customers into expensive bitstreams makes a lot of financial sense. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Redhat 9 OVER, Fidora Support, comments please.
|From: James Gardiner | Hi *ers, | I recently got an Email from Redhat about the dropping of support for Redhat | 9 on the 30 of April and that Fedora Project is the recommended future, | otherwise, RedHat enterprise ($$$). | Considering this, I would like some feed back on the Fedora Project from | users who may be using it, and how its going with Asterisk? Are there any | problems? I'm currently running it on 3 servers with no problems whatsoever. | Is the Asterisk development team got Fedora Project in mind and fully | supported? Fedora seems to far to be compatible enough with vanilla kernels. Matt Riddell ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Please help
I am only just starting out with * myself, but believe it or not I had the same problems not more than a couple of days ago. 1) With the X-Lite clients I was able to connect a call amongst them, but unable to hear a thing. (Same problem I suspect). The problem ended up being that the * server was not sending which audio protocol back to the client. (It was only send the DTMF protocol, which means that if you hit a number it would be heard.) I added the following lines to my sip.conf file and everything worked properly: [general] ; normal general settings go here disallow=all allow=ulaw allow=alaw allow=gsm 2) The scripts have been moved to the /usr/src/asterisk/contrib/scripts/ subdirectory. Once you run the script it will prompt you for the context, which I have left blank, and the extension. 3) I don't know because I haven't gotten that far. Hope this helps, Robert Jackson -Original Message- From: Marcias Martinez [mailto:[EMAIL PROTECTED] Sent: Sunday, April 04, 2004 5:28 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Please help Hi Guys, My name is Marcias, and I am setting up for the first time an Asterisk PBX, I am learning as I go along. I have been able to download and install Asterisk, Libpri and I have been able to get Asterisk up and running. I have several questions: 1 .I can call the Asterisk server from my Xten phone and it picks up. I have 3 computers (one of them being the asterisk server) I can seem to call from one computer to the other through the Asterisk server (All this is local within my network) but as soon as I pick up I cant hear anything on the other side. ANy ideas of whay this would be happening? 2. On another note, I have a website that Steve, was kind enough to re-direct me to. www.onlamp.com/lpt/a/3956 . Very nice website on how to setup everything, but it states that I am supposed to have an addmailbox script under /usr/src/asterisk. I don't seem to have this? So when I call in to listen to the voice mail that i have left I get the server informing me that I have voicemail, but I can't seem to listen to them. So I am not sure if I am missing something in the setup of the voicemail. 3. My next step is to learn how to change the welcome greetings and and then eventually how to hook up my server to another server or to hook a FWD number to it? Thanks for your help in advance, Marcias email: [EMAIL PROTECTED] FWD: 260032 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip no sound?
There was a question about this earlier. I had a similar problem and fixed it by specifying the audio protocol to be used in the general section of the sip.conf. -Original Message- From: Altus Snyman [mailto:[EMAIL PROTECTED] Sent: Monday, April 05, 2004 3:52 AM To: asterisk Subject: [Asterisk-Users] sip no sound? Good day all So I've installed asterisk with my openline4 card and I've setup sip and I can do sip on the local network,we are using soft clients,x-lite. But... When a call comes in from the outside(PSTN) and the you dial the extension it forwards the call the the client and you see incoming call on x-lite,you accept he callBUT there is no sound.It shows there is a call and you are connected but there is no sound Any Idea Please Help Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Redhat 9 OVER, Fidora Support, comments please.
James Gardiner wrote: Hi *ers, I recently got an Email from Redhat about the dropping of support for Redhat 9 on the 30 of April and that Fedora Project is the recommended future, otherwise, RedHat enterprise ($$$). Yup, this has been coming up for a while now.. Considering this, I would like some feed back on the Fedora Project from users who may be using it, and how its going with Asterisk? Are there any problems? I have started converting my systems to it and so far I have 3 servers and my desktop running FC1.. Is the Asterisk development team got Fedora Project in mind and fully supported? FC1 is basically what RHL10 would have been so compatibility is really the same as for RH9, the only issie is there appears to be an issue with the version of bison than comes with FC1 and Asterisk.. Installing the RH9 version of Bison solves the problem.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Redhat 9 OVER, Fidora Support, comments please.
Hi *ers, I recently got an Email from Redhat about the dropping of support for Redhat 9 on the 30 of April and that Fedora Project is the recommended future, otherwise, RedHat enterprise ($$$). Considering this, I would like some feed back on the Fedora Project from users who may be using it, and how its going with Asterisk? Are there any problems? Is the Asterisk development team got Fedora Project in mind and fully supported? I have four systems (one in production) running Fedora Core 1 without any obvious problems. Same rules apply for FC1 as for RH9 -- you have to install full versions of MPG123, Festival, libtiff, Postges, MySQL and some of the other packages that can be optionally used with Asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] The maximum capacity of MeetMe
two wrote: Hi !! I know that a conference room can be made infinitely. but, I think that there is actually a limit. For example, how many conference rooms can be made from CPU 866 [MHz] and RAM 256 [MB]? Is there any person who tried someone? I am studying MeetMe now. Please tell me a hint!! The answer is 1. That's right, only one conference. The answer is also probably some number greater than 30. Which answer applies to your situation, would depend on how many people are in each conference, what technologies they are using to connect, what kind of hardware is in your asterisk server, and how much bandwidth you have available. Now, go and figure out what you need asterisk to do, then test a configuration you have and figure out what your configuration needs to be. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Redhat 9 OVER, Fidora Support, comments please.
We had an issues with an Intel Zero Channel hardware RAID controller that wouldn't allow us to install either Fedora Core 1 or 2, so we couldn't test with *. Given that we didn't try to convert our 9 to Fedora, either. We got it running great under RH 9. HTH, Ryan Thrash On Apr 5, 2004, at 7:50 AM, James Gardiner wrote: Hi *ers, I recently got an Email from Redhat about the dropping of support for Redhat 9 on the 30 of April and that Fedora Project is the recommended future, otherwise, RedHat enterprise ($$$). Considering this, I would like some feed back on the Fedora Project from users who may be using it, and how its going with Asterisk? Are there any problems? Is the Asterisk development team got Fedora Project in mind and fully supported? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] avaya and linux
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Glen Ford Sent: Friday, April 02, 2004 2:48 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] avaya and linux Does anyone know if avaya voip product is running linux under the hood? Yes. The 5300 (even the non-voip featured ones) are a RedHat enterprise box with standard layer 2 switching hardware to connect the chassis together. Don't know about the other models, or even the current state of the 5300 platform, but the two or so year old ones I've been dealing with have the above config. Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 reload - how ?
Hi, My asterisk fails and stops after running the reload command ~20 times (I'm testing) - is this a kown problem ? Therefor I wil reload only sip, extensions and iax, it works with sip and extensions, but it seem that there are no reload for iax - or what ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] The maximum capacity of MeetMe
Hi !! Thank you for teaching!! A question is changed for a while. please tell me the information that the conference room was able to be made how many, by which spec. English cannot be used well and it is pardon!! English is also under study. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Problem and Question
On Mon, Apr 05, 2004 at 12:49:24AM -0400, Shad Mortazavi wrote: Dear Asterisk Users. I have been setting up IAX between two servers, one in the USA and the other in UK so that I can pass help desk and general calls from one call center to another. I seem to be having an issue. When I set up IAX between my two servers I get into trouble when doing a reload on the CLI. Almost as if the system had gone into a loop reading the configuration. I also have a question, if I use the switch command i.e. switch = IAX2/brunswick:[EMAIL PROTECTED]/sip and switch = IAX2/dornoch:[EMAIL PROTECTED]/sip can I point to the same context? Or does the context on each pbx need to be unique. Switch is not working right now. It will fowl up your connection. Switch is having some issues that need to be fixed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco QoS Howto
Can anyone point me to some sample Cisco QoS configurations suitable for IAX2? I've looked through Cisco's site, and get overwhelmed with the level of documentation (too much of a good thing). My PSTN gateway and PBX (both *) are connected via 2xT1 (per-packet load balancing) between a Cisco 7206 and a 3640. When the total bandwidth pushes much past 50%, I start getting some crazy distrotion (jitter?), making it impossible for one or both parties to understand the other. TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?
Personally, I think VAD is a great service, as well as comfort noise generation to disguise when VAD is working. I'll always encourage methods that reduce bandwidth. Most major developers on Asterisk consider these technologies of low concern since their bandwidth is unlimited, as they typically sit in a co-lo somewhere (as many programmers of * are providers of service, not consumers.) The reality for most end users is that they are on very restricted pipes that are delivered via a WAN technology (especially for outbound, if you consider residential) and being able to put more customers into expensive bitstreams makes a lot of financial sense. I agree fully. We need to implement a good timer in the SIP channel, both for VAD (but that's really in RTP, isn't it?) and for general SIP timers according to the RFC. Last week I also learned that DSL in the US is not as fat as DSL in general is over here. Anything below 384 upstream is nothing you can sell in Sweden :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco QoS Howto
Are you in control of both sides? What routing protocols are you using? Simply using Cisco CAR can help, but not a total solution. Are the 2 T1's carried by an ISP? Or are these private T's? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Troy Settle Sent: Monday, April 05, 2004 11:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco QoS Howto Can anyone point me to some sample Cisco QoS configurations suitable for IAX2? I've looked through Cisco's site, and get overwhelmed with the level of documentation (too much of a good thing). My PSTN gateway and PBX (both *) are connected via 2xT1 (per-packet load balancing) between a Cisco 7206 and a 3640. When the total bandwidth pushes much past 50%, I start getting some crazy distrotion (jitter?), making it impossible for one or both parties to understand the other. TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
Olle E. Johansson wrote: Richard Airlie wrote: On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote: Richard Airlie wrote: At this stage, and someone please confirm that Asterisk is really working on FreeBSD ? :) Yes, it's working with some limitations. See http://www.voip-info.org/tiki-index.php?page=Asterisk%20freebsd Thanks for that, good to know. And now leads me to ask... why should my SIP softphones be unable to register? They are on the same subnet as asterisk. If i have sip debug turned on, shouldn't I at least be seeing some action on the Asterisk console when they try to register? Turn on SIP debug and you'll be able to see what happens. Check with sockstat -l if Asterisk is listening to port 5060. Also, make sure you start asterisk with a lot of -v to get debug output. Since you see no SIP traffic with SIP debug on, is ipfw blocking SIP? Bob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation fault, exit status 139, ...
Hi!
I am running an * 0.7.2 on an X86 debian stable 2.4.25 (with
backports.org). The HW I am using is Digium's E100P on an HP DL 380.
Quite often it crashes, e.g. after a call has finished. Below some logs
form the * Console as well as from the /var/log/asterisk/messages
(Replaced some stuff with XXX).
Any idea what there could be the reason for this segmentaion fault?
What other indormation (e.g. configs) would be required to analyse
this problem further?
Thanx for you help!
cheers,
Bernie
* Console:
Apr 5 18:01:18 WARNING[24594]: app_dial.c:331 wait_for_answer:
Unable to forward voice
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
-- Executing Dial(SIP/xxx.switch.ch-0894ea58,
Zap/g1/04176XXX) in new stack
-- Called g1/04176
-- Zap/1-1 is ringing
-- Hungup 'Zap/1-1'
== Spawn extension (SIP, +4176XXX, 1) exited non-zero on
'SIP/xxx.switch.ch-0894ea58'
astra*CLI
/usr/sbin/safe_asterisk: line 6: 20873 Segmentation fault
asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
Disconnected from Asterisk server
[EMAIL PROTECTED]:/etc/asterisk$ /usr/sbin/safe_asterisk: line 6: 20905
Segmentation fault asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 6: 20925 Segmentation fault asterisk
${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: /dev/tty9: Input/output error
Asterisk ended with exit status 1
Asterisk died with code 1. Aborting.
* /var/log/asterisk/messages:
Apr 5 18:01:18 WARNING[24594]: Unable to forward voice
Apr 5 18:01:34 ERROR[1024]: Unable to load config iax1.conf
Apr 5 18:01:34 WARNING[1024]: Ignoring port for now
Apr 5 18:01:47 WARNING[16401]: Timeout, but no rule 't' in context 'SIP'
Apr 5 18:01:57 ERROR[1024]: Unable to load config iax1.conf
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Re: [Asterisk-Users] Cisco QoS Howto
Hi Troy, Troy Settle wrote: Can anyone point me to some sample Cisco QoS configurations suitable for IAX2? I've looked through Cisco's site, and get overwhelmed with the level of documentation (too much of a good thing). Take a look at this and see if you can use it for IAX2 as well: http://www.cisco.com/univercd/cc/td/doc/product/rtrmgmt/qos/qpm21/qpm21ug/ugvoip.htm [ snip ] HTH, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RPM packages
Andrey McRory built a RPM dist for * but I can't seem to find it anywhere.. Any hints where I might be able to find this package that has matching kernel? Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] The maximum capacity of MeetMe
I regret that I've only used MeetMe a few times, and only up to two users. Perhaps others that are using MeetMe could comment on the number of concurrent conferences and total users they have asterisk running with. The specs of the systems involved would be most helpful. If this is on the wiki, I apologize. If so, please post a url for the original poster. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RPM packages
Christopher C. Howard wrote: Andrey McRory built a RPM dist for * but I can't seem to find it anywhere.. Any hints where I might be able to find this package that has matching kernel? This is what I found for rpm. http://www.voip-info.org/wiki-Asterisk+RPM Hope this helps. Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] The maximum capacity of MeetMe
I regret that I've only used MeetMe a few times, and only up to two users. Perhaps others that are using MeetMe could comment on the number of concurrent conferences and total users they have asterisk running with. The specs of the systems involved would be most helpful. I have set up a conference with four people on a very low-end box. The voice quality was very good. All four were connected using VoIP. Three were using IAX2 clients, and one was using a SIP hardphone. I suspect the system could have easily scaled much further -- the CPU and memory usage were fairly low. Jeremy McNamera told me that he has run conferences with 184 users (two of the Quad-PRI cards) with out any problems. I also ran into a number of users at the VON show who are using Asterisk as a conference server (including the Pingtel team!). As for system specs, I believe that Jeremy's system is based on a dual-processor (P-IV 2.4 GHz) Dell 1750 server with several GB of RAM. My low end box was a 1.7 GHz Celeron with 256 MB. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] The maximum capacity of MeetMe
quote who=Andrew Thompson I regret that I've only used MeetMe a few times, and only up to two users. Well, the problem with giving general stats, is that it REALLY depends on the exact environment. Key points: (on a server dedicated for conferences only) o number of channels o types of channels o codecs used (and ratio) o number of conferences o number of channels in the conferences Then givin the interupt load, cpu load, i/o load, memory load and bandwidth for each of these variables, you can find what hardware will run the load you want. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Manager Originate
Thank you James for reply. Conole does not print any messages. When I trace SIP messages I can see that invitation is sent, and then it call is explicitly hung up. The phone starts to ring for a second and then goes quiet. The same thing happens if I originate on a Zap channel. On Zap channel * console just prints: -- Hungup 'Zap/1-1 Thank you Serge From: James Golovich [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problem with Manager Originate Date: Sun, 4 Apr 2004 15:02:42 -0400 (EDT) On Sun, 4 Apr 2004, Serge Mankovski wrote: Hi I am trying Manager interface for originate a call. This is what I get --- Action: Originate Exten: 555 CallerID: test 6656 Context: local Timeout: 600 Channel: SIP/8782 Priority: 1 Response: Error Message: Originate failed What do I do wrong? Check the errors/messages on your console. I suspect you will see some messages about 'unable to create channel SIP/8782'. The Originate failed message pretty much only comes up when there is a problem creating the Channel. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ STOP MORE SPAM with the MSN Premium and get 2 months FREE* http://join.msn.com/?pgmarket=en-capage=byoa/premxAPID=1994DI=1034SU=http://hotmail.com/encaHL=Market_MSNIS_Taglines ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID
I am having an issue with Callerid (INBOUND). I have a system set up with 4 companies sitting behind the system. On all of the companies except of one of them, it displays callerid withh 'asterisk'. The other company displays the callerid of the person calling. Zapata.conf [channels] musiconhold=default callerid=asreceived threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes jitterbuffers=4 immediate=no context=default-nga signalling=featd group=2 channel = 5-8 context=default-tne signalling=featd group=1 channel = 1-4 context=default-pb signalling=featd group=3 channel = 9-12 context=default=ctm signalling=featd group=3 channel = 13-14 Any thoughts -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAP channels
I have made bri-stuff.0.0.2rc19 to work (I think) but I can not achieve any in-dialing nor I can dial out; this is what I have from pri intense debug span 1 command -- *CLI pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 -- Executing Playback(SIP/201-a862, tt-weasels) in new stack -- Playing 'tt-weasels' (language 'en') -- Executing SetCallerID(SIP/201-a862, 340) in new stack -- Executing Dial(SIP/201-a862, Zap/1/333) in new stack [02 ff 03 08 01 03 05 04 03 80 90 a3 18 01 89 6c 05 21 81 33 34 30 70 03 c1 33 33 a1 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 25 bytes of data Protocol Discriminator: Q.931 (8) len=25 Call Ref: len= 1 (reference 3/0x3) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '340' ] Called Number (len= 5) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '33' ] Sending Complete (len= 0) -- Called 1/333 -- Hungup 'Zap/1-1' == Spawn extension (default, 4000, 3) exited non-zero on 'SIP/201-a862' -- from what I understand here I have initiated a call on ZAP but it dialed number 1/333 instead of 333 this is my zapata.conf file [channels] ; ; Default language ; ;language=en ; ; Default context ; ; switchtype = euroisdn signalling = bri_net_ptmp pridialplan=local echocancel=yes immediate=yes context=default group = 1 stripmsd= 1 channel = 1-2 --- and here is a portion of extensions.conf which deals with outside call (or tries); exten = 4000,1,Playback(tt-weasels) exten = 4000,2,SetCallerID(340) exten = 4000,3,Dial(Zap/1/333) exten = _5xxx,1,Dial,Zap/g1/BYEXTENSION can anybody help me to fix what I am doing wrong? The linuX Files -- The Source is Out There. mailto:[EMAIL PROTECTED] http://printel.hr ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Buzzing on TDM400P FXS?
I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for buzzing on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and IAX to NuFone, so I'm pretty sure that it's happening on the FXS end. Here's that chunk of zapata.conf: context=inside-analog signalling=fxo_ks callwaiting=yes callwaitingcallerid=yes cancallforward=yes callreturn=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=0 immediate=no musiconhold=yes usecallerid=yes callerid=Analog Phone 2201 mailbox=2201 channel = 2 Does anyone have any suggestions on where to start looking? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spring VON Wrap Up
Having just returned from four days at the VON show in Santa Clara, I thought I would submit a highlights message. I hope others who attended the show will take the opportunity to add, as there was far more to see than I can cover on my own. [VoIP IS BIG] First, I have to say that VoIP is BIG. It is the buzz technology of the day. The show was packed, and everybody there was there for a reason. Jeff Pulver, in his introductory remarks told us the walking dead count was zero and he was right. Wall-to-wall VoIP people. [Who Was There] The crowd was a mix of service providers (including CLECs, VoIP pure-plays, ISPs adding VoIP as a service, etc.) and VoIP product vendors looking to sell solutions to the providers. Also sprinkled into the group were regulators from the FCC, advocates for various technologies, representatives from various industry groups, and a fair number of lawyers. Perhaps the most interesting story here was the nearly even split between US citizens and those from other nations. [What Was Hot] 1. SIP. Every presentation I saw mentioned SIP at some point. While it has been obvious for some time that SIP is poised to become _the_ standard for telecom in the this century, the constant repetition is a good indicator that the standards wars are actually over and SIP stands as the survivor. 2. Presence. Everybody wants to know when and where everybody is at all times. Buddy lists are in, dial-pads are out. The message is also clear that presence will go beyond online/away/offline to include actual geographic location. It will also move away from device-centric presence (knowing that a cell phone is on) to user-centric presence (knowing how a user wants to communicate at the time). We need to add presence to Asterisk. Now. 3. Asterisk. While those of us in the Asterisk community have known for some time that Asterisk can do nearly anything, given a bit of time and effort, the word seems to have spread. Asterisk was mentioned in Keynotes, Industry Perspectives, the Town Hall meeting, and in numerous breakout sessions. Hundreds of people came by the Digium/Asterisk booth to either find out more about the system, or to crow about what they are doing with Asterisk. In a feat of irony worthy of mention, Pingtel announced their new SIP Forge organization over an audio conference hosted on an Asterisk system. Asterisk is definitely hot. 4. EoIP (Everything Over IP). The lingo of the trade seems to be changing as things mature. Voice is just one application among many. Robert Pepper of the FCC described that agency's focus as moving to IP communications in general, rather than simply Voice. This makes sense. Voice really _is_ just one of many modes of communication, and a long way away from the original VoIP service. 5. Regulatory Concerns. Several of the presenters brought up social an legal issues related to VoIP, and the associated government regulations that follow. E911 service and CALEA (wiretapping) were both the big concerns, as was inter-carrier compensation and taxation. Dr. Pepper indicated that he was pleased with the direction that the VoIP market is going, in terms of the voluntary compliance with the relevant rules from the existing PSTN regs. He indicated that the FCC was, for the time being, willing to regulate minimally -- following the same model used for the Wireless carriers over the past decade. 6. VoIP Broadband Services. With ATT's announcement that it was moving into the residential and business VoIP market (joining Packet8, Vonage, and countless others), it became clear that the industry has moved beyond how to do VoIP, and into the era of how make money at VoIP. This is a fantastic change for everybody, including the Asterisk community. The gold rush has started, and those of us who understand Asterisk are in a great position to sell shovels to those heading west. Many CLECs and ISPs moving into the business are in need of solutions that work and people who can configure them. Do the math. 7. Session Border Controllers. Everybody seems to want to build walled gardens at this point. Some to keep customers from ENUMing their way to no-cost phone service, others to keep potential bad guys from abusing their resources. Nearly every presentation (at least the technical presentations) mentioned SBCs and the associated positive and negative effects they have on VoIP adoption and scalability. The jury is still out on whether the net result is positive or negative. Thoughts? [Thanks To Digium] Digium's booth became the home-away-from-home for the Asterisk community. At times there were probably 20 to 30 people crowded in and around the display. Many thanks to Mark and Greg who let all of us gather and (I hope) help pitch Asterisk and Digium. [Retraction (Steve Eats Crow)] I would like to retract a statement I made in an earlier report from the show. After sitting through two presentations by ATT, both pitching their new CallVantage
RE: [Asterisk-Users] Buzzing on TDM400P FXS?
Haven't seen this, but I do hear a loud click about 5 seconds into any call involving a TDM400P port. Seems like something might not be quite right with the Zap driver. -brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: Monday, April 05, 2004 1:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Buzzing on TDM400P FXS? I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for buzzing on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and IAX to NuFone, so I'm pretty sure that it's happening on the FXS end. Here's that chunk of zapata.conf: context=inside-analog signalling=fxo_ks callwaiting=yes callwaitingcallerid=yes cancallforward=yes callreturn=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=0 immediate=no musiconhold=yes usecallerid=yes callerid=Analog Phone 2201 mailbox=2201 channel = 2 Does anyone have any suggestions on where to start looking? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Steven Sokol wrote: Having just returned from four days at the VON show in Santa Clara, I thought I would submit a highlights message. I hope others who attended the show will take the opportunity to add, as there was far more to see than I can cover on my own. Thank you for a good report! Comments inline: 1. SIP. Every presentation I saw mentioned SIP at some point. While it has been obvious for some time that SIP is poised to become _the_ standard for telecom in the this century, the constant repetition is a good indicator that the standards wars are actually over and SIP stands as the survivor. No one mentioned H.323 any more. It's SIP and only SIP. 2. Presence. Everybody wants to know when and where everybody is at all user wants to communicate at the time). We need to add presence to Asterisk. Now. Right. In SIP and IAX2. Maybe see if we can use Jabber/XMPP for IM integration. 3. Asterisk. While those of us in the Asterisk community have known for SIP Forge organization over an audio conference hosted on an Asterisk system. Asterisk is definitely hot. SIPfoundry.org - no source available yet. And yes, they showed a lot of interest to cooperate with Digium and the asterisk.org community. 4. EoIP (Everything Over IP). The lingo of the trade seems to be changing as things mature. Voice is just one application among many. Robert Pepper of the FCC described that agency's focus as moving to IP communications in general, rather than simply Voice. This makes sense. Voice really _is_ just one of many modes of communication, and a long way away from the original VoIP service. Asterisk SIP supports video now. We're a multimedia platform. 5. Regulatory Concerns. Several of the presenters brought up social an legal issues related to VoIP, and the associated government regulations that follow. E911 service and CALEA (wiretapping) were both the big concerns, as was inter-carrier compensation and taxation. Dr. Pepper indicated that he was pleased with the direction that the VoIP market is going, in terms of the voluntary compliance with the relevant rules from the existing PSTN regs. He indicated that the FCC was, for the time being, willing to regulate minimally -- following the same model used for the Wireless carriers over the past decade. Members of the IETF added information on the to-be-standardized standard, meaning that SIP with TLS over TCP will be mandatory. We need to start working on TCP and TLS support. [Asterisk Get-Together] About 25 of us (I think) gathered at the Mexicali Grill in Santa Clara for a post-show celebration and discussion. It was a BLAST. Even as tired as most of us were (four days of trade show can wear down just about anybody) we all had a great time. It was cool to be able to put faces with names/email addresses. I think Olle Johansson took pictures of the event. They may already be on the WiKi in fact. Not yet, but I'm working on getting them uploaded. Still trying to get accustomed to the cold weather and strange time zone up here in the north. Thank you Steve for organizing this meeting! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Seattle IAX Termination
Packetwest Communcations provides local IAX termination service in Seattle. I use it locally for a small Asterisk setup and they provide me with DID's in the 206 NPA. They also provide outbound long-distance at rates similar to NuFone. I've had a really good experience with service quality and reliability so far. Contact them at (206) 838-4810. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Muiz Motani Sent: Friday, April 02, 2004 2:47 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Seattle IAX Termination Does anybody know of any commercial providers of IAX termination with DIDs in the Seattle, WA area? I believe the area codes are: 425, 206, 253 Failing any commercial providers, is there anybody in the seattle area running Asterisk with a PRI coming in who might be willing to sell me an IAX trunk with a DID in Seattle? -- Muiz Motani Intelligent Distribution 72-6800 Lynas Lane, Richmond, B.C. V7C 5E2 email: [EMAIL PROTECTED] phone: +1 604 448 9293 fax: +1 604 448 9296 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Call Queuing App?
X-Analitica - MD-MailScanner-OpenProtect-Information: Please contact the ISP for more information X-Analitica - MD-MailScanner-OpenProtect: Found to be clean X-MailScanner-MCPCheck: is it already inside * 0.7.2? El lun, 05 de 04 de 2004 a las 03:21, Senad Jordanovic escribi: here you go :) http://bugs.digium.com/bug_view_page.php?bug_id=214 Ta SJ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Kirkland Sent: 05 April 2004 04:25 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New Call Queuing App? I thought I saw online in a list somewhere that theres an improved Call Queuing app out; supposedly it has the capability to tell the caller how far down the queue they are, etc? I saw one post about it somewhere but then no mention of it anywhere else Andy, [EMAIL PROTECTED] --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.648 / Virus Database: 415 - Release Date: 3/31/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.648 / Virus Database: 415 - Release Date: 3/31/2004 -- Este mensaje ha sido analizado por OpenProtect en busca de virus y otros contenidos peligrosos, y se considera que est limpio. -- Este mensaje ha sido analizado por OpenProtect en busca de virus y otros contenidos peligrosos, y se considera que est limpio. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote: Members of the IETF added information on the to-be-standardized standard, meaning that SIP with TLS over TCP will be mandatory. We need to start working on TCP and TLS support. Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADPCM 4-bit, 6 kHz
I found some posts regarding this issue dating of September 2003, but no real answer. The ADPCM format supported by Asterisk (the .vox files) is 4-bit, 8 kHz. I need 4-bit, 6 kHz, which is also a widespread Dialogic format, to help migration. Is there an existing format/codec for this? If not, can I make myself a shared object in /usr/lib/asterisk/modules? Is this easy??? :-( Thanks, Yves Chouinard Vox-Tel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seattle IAX Termination
On Apr 5, 2004, at 12:18 PM, Mark Hagler wrote: Packetwest Communcations provides local IAX termination service in Seattle. I use it locally for a small Asterisk setup and they provide me with DID's in the 206 NPA. They also provide outbound long-distance at rates similar to NuFone. I've had a really good experience with service quality and reliability so far. Contact them at (206) 838-4810. Wow, a VoIP company that's even harder to contact then NuFone. Impressive :-). What kind of rates do they charge for DID numbers? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
On Mon, 5 Apr 2004, Scott Laird wrote: Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? The RTP would still be UDP. Just the SIP part (call signaling) would be TCP. SIP can be TCP or UDP, many implementations (including asterisk) support only UDP. TCP for SIP (especially with TLS) will reduce the risk of a mitm attack. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Spring VON Wrap Up
On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote: Members of the IETF added information on the to-be-standardized standard, meaning that SIP with TLS over TCP will be mandatory. We need to start working on TCP and TLS support. Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? TCP/TLS would be used for the SIP messaging which handles call setup, teardown, and other non-Realtime functions. The voice stream will still be handled via RTP which is a UDP-based protocol. The reason for doing the call setup as TCP is to allow for TLS encryption. The SIP messages themselves are simply bits of ASCII text (much like SMTP messages). Currently Asterisk does SIP over UDP only (I think...). In order to support SIPS (Secure SIP, like HTTPS) we need to build a version of chan_sip (or chan_sip2 ;-) that supports SIP over TCP. The voice stream will remain UDP an therefore not succumb to enormous delay. -S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
On Apr 5, 2004, at 12:34 PM, James Golovich wrote: On Mon, 5 Apr 2004, Scott Laird wrote: Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? The RTP would still be UDP. Just the SIP part (call signaling) would be TCP. SIP can be TCP or UDP, many implementations (including asterisk) support only UDP. TCP for SIP (especially with TLS) will reduce the risk of a mitm attack. Ah, okay. That makes sense. Thanks. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Call Queuing App?
Asterisk - MD wrote: X-Analitica - MD-MailScanner-OpenProtect-Information: Please contact the ISP for more information X-Analitica - MD-MailScanner-OpenProtect: Found to be clean X-MailScanner-MCPCheck: is it already inside * 0.7.2? Yap... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Spring VON Wrap Up
Scott Laird [EMAIL PROTECTED] wrote: On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote: Members of the IETF added information on the to-be-standardized standard, meaning that SIP with TLS over TCP will be mandatory. We need to start working on TCP and TLS support. Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? It is only SIP that would be on TCP. RTP (media stream) would still be UDP. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Scott Laird wrote: On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote: Members of the IETF added information on the to-be-standardized standard, meaning that SIP with TLS over TCP will be mandatory. We need to start working on TCP and TLS support. Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? SIP over TCP means signalling over TCP. Media is still usually RTP/UDP. SIP over TCP and TLS authenticates both ends and may also protect the signalling with encryption. SRTP protects RTP/UDP media with encryption. There are concerns that sending positioning within SIP/UDP will reveal private detailes, like position. Hence the encryption requirement. The position data needs to be given by the ISP in DHCP configuration. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
James Golovich wrote: On Mon, 5 Apr 2004, Scott Laird wrote: Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? The RTP would still be UDP. Just the SIP part (call signaling) would be TCP. SIP can be TCP or UDP, many implementations (including asterisk) support only UDP. TCP for SIP (especially with TLS) will reduce the risk of a mitm attack. ...and SIP over TCP is a requirement in the SIP RFC... /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Steven Sokol wrote: I think Olle Johansson took pictures of the event. They may already be on the WiKi in fact. I've uploaded the pictures without editing at http://www.voip-forum.com/asterisk/von2004/index.htm Enjoy! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco QoS Howto
You can also take a look at the following URL: http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_command_ref erence_chapter09186a0080087f26.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Troy Settle Sent: Monday, April 05, 2004 11:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco QoS Howto Can anyone point me to some sample Cisco QoS configurations suitable for IAX2? I've looked through Cisco's site, and get overwhelmed with the level of documentation (too much of a good thing). My PSTN gateway and PBX (both *) are connected via 2xT1 (per-packet load balancing) between a Cisco 7206 and a 3640. When the total bandwidth pushes much past 50%, I start getting some crazy distrotion (jitter?), making it impossible for one or both parties to understand the other. TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk IAX gatewway
Hi, We are using Nufone as our voip provider and it is working fine except for the problems i mentioned in my email. Thanks Owais Bin Zuber"James H. Thompson" [EMAIL PROTECTED] wrote: Just curious - was wondering who you are using as your VOIP provider and how its working out? Thanks Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Owais Zuber To: [EMAIL PROTECTED] Sent: Friday, April 02, 2004 8:00 PM Subject: [Asterisk-Users] * server acting as SIP/IAX gateway problem Hi, My company is a call center and we are using * server for voip calls to america. * server is installed on a dual CPU machine and it is acting as a SIP/IAX2 gateway. SIP protocol is used for agents to connect to * server and * server used IAX2 protocol to connect to our VoIP service provider. There are around 25 agents currently working and making maximum 25 calls simentaneously. Agents are using estara softphone (www.estara.com) as SIP client. We are making around 4000 calls to america in one day and we are working 7 hours per day. Do you Yahoo!? Yahoo! Small Business $15K Web Design Giveaway - Enter today
Re: [Asterisk-Users] Spring VON Wrap Up
On Apr 5, 2004, at 12:49 PM, Olle E. Johansson wrote: SRTP protects RTP/UDP media with encryption. There are concerns that sending positioning within SIP/UDP will reveal private detailes, like position. Hence the encryption requirement. The position data needs to be given by the ISP in DHCP configuration. This brings up two more questions: 1. What does 'positioning' mean in a SIP context--Google isn't helpful. Is this basically just physical location? 2. Is anyone working on SRTP for Asterisk? Are there any SRTP clients out there? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Spring VON Wrap Up
K...maybe this was stated earlier in the conversation...but what's the deal with the phone? Or was this phone just being carried around by everyone and ripped apart? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Monday, April 05, 2004 3:59 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Spring VON Wrap Up Steven Sokol wrote: I think Olle Johansson took pictures of the event. They may already be on the WiKi in fact. I've uploaded the pictures without editing at http://www.voip-forum.com/asterisk/von2004/index.htm Enjoy! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP dataflow directly from a SIP phone to a H323 phone
Hi there, Is there anyway to make the RTP data flow directly a SIP phone and a H323 phone through the oh323 or chan_h323 modules? Something like waht the canreinvite = yes option inside the sip.conf does for SIP to SIP calls. thanks, Pablo Salinas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
At 01:44 PM 4/5/2004, you wrote: Having just returned from four days at the VON show in Santa Clara, I thought I would submit a highlights message. I hope others who attended the show will take the opportunity to add, as there was far more to see than I can cover on my own. Was there any aggressive pricing given for nationwide voip LD? I just lost an Internet customer today who has 6 voice and 2 fax business lines. He is moving to McLeod (regional bankrupt CLEC) for both voice and data. They are putting in a T1 and giving him 2.2 cents a minute for nationwide LD. He had to sign a 3 year contract. The CLEC battle is heating up here. I can't compete when I have to pay more than that for VOIP LD calls that terminate on POTS. Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Scott Laird wrote: On Apr 5, 2004, at 12:49 PM, Olle E. Johansson wrote: SRTP protects RTP/UDP media with encryption. There are concerns that sending positioning within SIP/UDP will reveal private detailes, like position. Hence the encryption requirement. The position data needs to be given by the ISP in DHCP configuration. This brings up two more questions: 1. What does 'positioning' mean in a SIP context--Google isn't helpful. Is this basically just physical location? If I understand Brian correctly, it will be a global system that can look up the closes 911 service - any where. Possibly latitude and longitude. Drafts out there somewhere, RFCs on it's way before new year. 2. Is anyone working on SRTP for Asterisk? Are there any SRTP clients out there? SIPfoundry got one, another one on SourceForge - maybe they're the same. More information about SRTP and pointers: http://www.voip-info.org/tiki-index.php?page=srtp /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Mark Messmore, Technical Support, University Telcom Inc. wrote: K...maybe this was stated earlier in the conversation...but what's the deal with the phone? Or was this phone just being carried around by everyone and ripped apart? Old Bell Phone + IAXy + 802.11b card + Batteries = Homemade WiSIP, IIRC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Scott Laird wrote: 2. Is anyone working on SRTP for Asterisk? Are there any SRTP clients out there? Checked again, the vovida.org and the sourceforge one are the same. And here's the good news: THey're using a BSD license. That means we can incorporate this library into Asterisk without a licensing problem. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Change IP info.
Hello i was wondering how i can change the IP address information for my Asterisk box, IP addy, Gateway, DNS. I have a smoothwall router that i am using and i am tring to put the Asterisk box on the orange interface so if anyone can help me please i can use it. Thanks alot William Ray
Re: [Asterisk-Users] Spring VON Wrap Up
Was there any aggressive pricing given for nationwide voip LD? Level3 had several products, one they called Enhanced which was supposed to also include E911 service. They quoted me about $.01 per minute inbound or outbound nation wide. They said they support the top 300 cities in the US and, of course, have plans to serve every rate center in the US. I also went and talked with ITXC, but the rather bad sales person said they were only really interested in international calling and not domestic LD. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Passing
I am trying to get dtmf digits to pass from a SNOM 200 through * to a Cisco AS5300. I have setup the cisco gateway and the sip.conf file to use rfc2833 and I have disabled inband dtmf on the snom 200. Unfortunately, the digits are still not being passed. Something tells me that I am missing something in the extensions.conf file, but I am at a loss. I would greatly appreciate any help you can give me. Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disambiguating incoming IAXTel calls
Title: Disambiguating incoming IAXTel calls I have two 1-700 numbers from IAXTel. Both get registered from the same Asterisk server. I can make and receive calls on each without any difficulty. What I can't figure out how to do is route the incoming calls differently based on which 1-700 number is dialed. I must be missing something obvious. Thanks -brian
[Asterisk-Users] Auto connect to voicemail
I have the voicemail setup working in that I get the MWI and it emails the message correctly. When I pressed the MWI button on my SNOM 200, it dials into the voicemail system and prompts me for a mailbox and password. I know there is a way to automatically connect directly into the mailbox via the extension.conf file, but I can not find the documentation I am looking for in reference to variables and macros for the extensions file. Can someone please help me with this issue? Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Seattle IAX Termination
Their base rate is $35/mo per peer (single call transit at any given moment) and this provides unlimited local and inbound calling. If you are connecting a PBX and need 1 voice path at any given moment you can discuss different pricing arrangements for your needs. DID numbers are 15 cents/number/month. Long distance is something around 3 or 4 cents/min, I don't recall exactly. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: Monday, April 05, 2004 12:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Seattle IAX Termination On Apr 5, 2004, at 12:18 PM, Mark Hagler wrote: Packetwest Communcations provides local IAX termination service in Seattle. I use it locally for a small Asterisk setup and they provide me with DID's in the 206 NPA. They also provide outbound long-distance at rates similar to NuFone. I've had a really good experience with service quality and reliability so far. Contact them at (206) 838-4810. Wow, a VoIP company that's even harder to contact then NuFone. Impressive :-). What kind of rates do they charge for DID numbers? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extensions.conf sending calls to Cisco AS5300
I have my server configured to send to send all PSTN traffic to my Cisco AS5300 gateway via SIP. I use the following line in the extensions.conf file to accomplish this: exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],240,T) Unfortunately, when I removed the T from the end of the statement, the calls still complete, but they drop as soon as the called party answers the phone. I thought that the T had something to do with a timeout, but I have also seen documentation referencing that it allows * to stay in the middle of the call to determine if the customer use the # key, etc. I have not been able to find the detailed documentation that I was looking for on this subject. Can someone please direct me to this? Also it is my understanding, that if * stays in the middle of the call, I can not use the g729 codec without licensing from Digium. If this is the case, is there a way that I can use g729 in pass thru and still complete calls to the gateway? Any help would be greatly appreciated. Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto connect to voicemail
On Apr 5, 2004, at 1:57 PM, Brian Rathman wrote: I have the voicemail setup working in that I get the MWI and it emails the message correctly. When I pressed the MWI button on my SNOM 200, it dials into the voicemail system and prompts me for a mailbox and password. I know there is a way to automatically connect directly into the mailbox via the extension.conf file, but I can not find the documentation I am looking for in reference to variables and macros for the extensions file. Can someone please help me with this issue? exten = ..., VoiceMailMain(s1234) will connect to box 1234 without prompting for a password or mailbox number. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto connect to voicemail
I use something like this:
exten = 8500,1,Ringing
exten = 8500,2,Wait,1
exten = 8500,3,VoicemailMain(s${CALLERIDNUM})
Basically, this rings the phone for once second (thus setting up the audio
path), then goes to voicemail without requiring the password. Leave out the
's' to have VM prompt for the password.
-brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brian Rathman
Sent: Monday, April 05, 2004 3:58 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Auto connect to voicemail
I have the voicemail setup working in that I get the MWI and
it emails the message correctly. When I pressed the MWI
button on my SNOM 200, it dials into the voicemail system and
prompts me for a mailbox and password. I know there is a way
to automatically connect directly into the mailbox via the
extension.conf file, but I can not find the documentation I
am looking for in reference to variables and macros for the
extensions file. Can someone please help me with this issue?
Thanks,
Brian
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Re: [Asterisk-Users] Spring VON Wrap Up
Bob Klepfer wrote: Mark Messmore, Technical Support, University Telcom Inc. wrote: K...maybe this was stated earlier in the conversation...but what's the deal with the phone? Or was this phone just being carried around by everyone and ripped apart? Old Bell Phone + IAXy + 802.11b card + Batteries = Homemade WiSIP, IIRC After a peek under the hood, I would guess we could have these manufactured over seas for around $1000 USD per unit. It would not be the same to modify the design in any way. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropped calls, 5-10 seconds of silence
Hello, We have an * installation that is causing us fits. The problems we are seeing: 1) In the middle of a call the call gets dumped and the caller hears a dial tone. 2) While talking on a call the caller hears nothing for 5 to 10 seconds. The person on the other end of the call hears everything just fine. Then the call returns to normal and both parties can hear. our network: PSTN -- Asterisk /w T100p -IAX2- Asterisk2 /w zapdummy -sip snom 200's Any insight would be much appreciated. lach ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seattle IAX Termination
On Apr 5, 2004, at 2:01 PM, Mark Hagler wrote: Their base rate is $35/mo per peer (single call transit at any given moment) and this provides unlimited local and inbound calling. If you are connecting a PBX and need 1 voice path at any given moment you can discuss different pricing arrangements for your needs. DID numbers are 15 cents/number/month. Long distance is something around 3 or 4 cents/min, I don't recall exactly. Okay, thanks. So they're on the Vonage-style all-you-can-eat model instead of the VoicePulse Connect/NuFone pay-as-you go model. Right now, I'm better off paying $0.029/minute for an 800-number through NuFone then $35/month for a free local number that doesn't get all that much use. It's nice to know that they're available, though. I'll try to throw some business their way if anything presents itself. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
On Mon, Apr 05, 2004 at 11:16:39AM -0500, Bob Klepfer wrote: Olle E. Johansson wrote: Richard Airlie wrote: On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote: And now leads me to ask... why should my SIP softphones be unable to register? They are on the same subnet as asterisk. If i have sip debug turned on, shouldn't I at least be seeing some action on the Asterisk console when they try to register? Turn on SIP debug and you'll be able to see what happens. Check with sockstat -l if Asterisk is listening to port 5060. Also, make sure you start asterisk with a lot of -v to get debug output. Since you see no SIP traffic with SIP debug on, is ipfw blocking SIP? I'm actually running IPFilter, but I've checked the logs and it definitely isn't blocking any SIP traffic. And I've also confirmed that Asterisk is listening on port 5060 with netstat. So.. Asterisk is running, listening on UDP port 5060, the firewall hasn't logging any blocked packets, and yet my IP softphones still cant register. This leads me to believe I must be doing something really stupid. My Asterisk server is 192.168.100.3. Kphone is running on 192.168.100.13, and SJPhone is on 192.168.100.11. I'm configuring the softphones so that they register with the (outbound) proxy at 192.168.100.3. I've set their IDs to be sip:[EMAIL PROTECTED], and created the appropriate username and password in sip.conf on Asterisk. I turn sip debug on at the Asterisk console, then restart the phones. They log lots of attempts to register in the softphone windows, but Asterisk doesn't see anything at all. (I can also get the softphones to talk directly to one another and they seem to be working fine). I guess my next step will be tcpdump.. but any other suggestions most welcomed! best, Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto connect to voicemail
On Mon, 2004-04-05 at 15:57, Brian Rathman wrote: I have the voicemail setup working in that I get the MWI and it emails the message correctly. When I pressed the MWI button on my SNOM 200, it dials into the voicemail system and prompts me for a mailbox and password. I know there is a way to automatically connect directly into the mailbox via the extension.conf file, but I can not find the documentation I am looking for in reference to variables and macros for the extensions file. Can someone please help me with this issue? Thanks, Brian Brian, At the CLI, type 'show application VoiceMailMain'. You can use the CLI 'show applications' command to list all available apps. If you hit tab, it acts just like BASH's auto complete. Wonderful feature! Mitch Sharp Innovative Solutions
Re: [Asterisk-Users] Extensions.conf sending calls to Cisco AS5300
On Mon, 2004-04-05 at 22:02, Brian Rathman wrote: I have my server configured to send to send all PSTN traffic to my Cisco AS5300 gateway via SIP. I use the following line in the extensions.conf file to accomplish this: exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],240,T) Unfortunately, when I removed the T from the end of the statement, the calls still complete, but they drop as soon as the called party answers the phone. I thought that the T had something to do with a timeout, but I have also seen documentation referencing that it allows * to stay in the middle of the call to determine if the customer use the # key, etc. I have not been able to find the detailed documentation that I was looking for on this subject. Can someone please direct me to this? Also it is my understanding, that if * stays in the middle of the call, I can not use the g729 codec without licensing from Digium. If this is the case, is there a way that I can use g729 in pass thru and still complete calls to the gateway? Any help would be greatly appreciated. Sorry, 'T' prevents pass-thru: http://voip-info.org/wiki-Asterisk+G.729+pass-thru F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Capacity
And could anybody say the concurrent calls limit for one Asterisk Box? Let's say it is a Pentium IV 1.6GHz, 256 MB RAM, RedHat 9 thanks, Pablo Salinas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto connect to voicemail
I think this is what you are looking for Exten = 1000,1,Answer,1Exten = 1000,2,Wait,1Exten = 1000,3,Voicemailmain([EMAIL PROTECTED]) - Original Message - From: Mitchell S. Sharp To: [EMAIL PROTECTED] Sent: Monday, April 05, 2004 5:27 PM Subject: Re: [Asterisk-Users] Auto connect to voicemail On Mon, 2004-04-05 at 15:57, Brian Rathman wrote: I have the voicemail setup working in that I get the MWI and it emails the message correctly. When I pressed the MWI button on my SNOM 200, it dials into the voicemail system and prompts me for a mailbox and password. I know there is a way to automatically connect directly into the mailbox via the extension.conf file, but I can not find the documentation I am looking for in reference to variables and macros for the extensions file. Can someone please help me with this issue? Thanks, BrianBrian,At the CLI, type 'show application VoiceMailMain'. You can use the CLI 'show applications' command to list all available apps. If you hit tab, it acts just like BASH's auto complete. Wonderful feature!Mitch SharpInnovative Solutions
Re: [Asterisk-Users] Auto connect to voicemail
The snom dials into an account caled 'asterisk'
Exten = asterisk,1,Answer,1
Exten = asterisk,2,Wait,1
Exten = asterisk,3,Voicemailmain(${CALLERIDNUM})
- Original Message Follows -
I think this is what you are looking for
Exten = 1000,1,Answer,1
Exten = 1000,2,Wait,1
Exten = 1000,3,Voicemailmain([EMAIL PROTECTED])
- Original Message -
From: Mitchell S. Sharp
To: [EMAIL PROTECTED]
Sent: Monday, April 05, 2004 5:27 PM
Subject: Re: [Asterisk-Users] Auto connect to voicemail
On Mon, 2004-04-05 at 15:57, Brian Rathman wrote:
I have the voicemail setup working in that I get the MWI
and it emails the message correctly. When I pressed the
MWI button on my SNOM 200, it dials into the voicemail
system and prompts me for a mailbox and password. I know
there is a way to automatically connect directly into the
mailbox via the extension.conf file, but I can not find
the documentation I am looking for in reference to
variables and macros for the extensions file. Can someone
please help me with this issue?
Thanks,
Brian
Brian,
At the CLI, type 'show application VoiceMailMain'. You
can use the CLI 'show applications' command to list all
available apps. If you hit tab, it acts just like BASH's
auto complete. Wonderful feature!
Mitch Sharp
Innovative Solutions
Willy Wouters
ypOne Publishing
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[Asterisk-Users] Stable Relase Broken ?
All, I upgraded to the [*] stable release branch. When I call into the box (confirmed on 2 installations) the caller no longer hears the ringing. The CLI confirms that extensions are being 'rung'. Whassup? Willy Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stable Relase Broken ?
More Info: And I went back to CVS-03/26/04 and can hear the 'ringing' again when I call in to the box ... BTW: This behavior exists on the production system (T1 PRI interface to PSTN only) and on the Developent system (FXO/FXS and IAX2 interfaces) Cheers, Willy - Original Message Follows - All, I upgraded to the [*] stable release branch. When I call into the box (confirmed on 2 installations) the caller no longer hears the ringing. The CLI confirms that extensions are being 'rung'. Whassup? Willy Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto connect to voicemail
What do you do when $CALLERIDNUM of the caller isnt the 4-digit extension? I set all of my users Caller ID entries to their 10-digit phone # so that Caller ID appears correctly when I send their call out the PRI to the public network. The side effect of this is breaking convenient access to voicemail using this method, and I havent found a way to fix it yet. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Glenn Dalgliesh Sent: Monday, April 05, 2004 3:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Auto connect to voicemail I think this is what you are looking for Exten = 1000,1,Answer,1 Exten = 1000,2,Wait,1 Exten = 1000,3,Voicemailmain([EMAIL PROTECTED]) - Original Message - From: Mitchell S. Sharp To: [EMAIL PROTECTED] Sent: Monday, April 05, 2004 5:27 PM Subject: Re: [Asterisk-Users] Auto connect to voicemail On Mon, 2004-04-05 at 15:57, Brian Rathman wrote: I have the voicemail setup working in that I get the MWI and it emails themessage correctly. When I pressed the MWI button on my SNOM 200, it dialsinto the voicemail system and prompts me for a mailbox and password. I knowthere is a way to automatically connect directly into the mailbox via theextension.conf file, but I can not find the documentation I am looking forin reference to variables and macros for the extensions file. Can someoneplease help me with this issue?Thanks,Brian Brian, At the CLI, type 'show application VoiceMailMain'. You can use the CLI 'show applications' command to list all available apps. If you hit tab, it acts just like BASH's auto complete. Wonderful feature! Mitch Sharp Innovative Solutions
RE: [Asterisk-Users] Stable Relase Broken ?
I ran into the same problem. It seems to be fixed in later builds. -brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, April 05, 2004 5:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Stable Relase Broken ? All, I upgraded to the [*] stable release branch. When I call into the box (confirmed on 2 installations) the caller no longer hears the ringing. The CLI confirms that extensions are being 'rung'. Whassup? Willy Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto connect to voicemail
On Apr 5, 2004, at 3:53 PM, Mark Hagler wrote:
What do you do when $CALLERIDNUM of the caller isnt the 4-digit
extension? I set all of my users Caller ID entries to their
10-digit phone # so that Caller ID appears correctly when I send their
call out the PRI to the public network. The side effect of this is
breaking convenient access to voicemail using this method, and I
havent found a way to fix it yet.
Use the database.
exten = 1000,1,Answer,1
exten = 1000,2,Wait,1
exten = 1000,3,DBGet(MYEXTEN=extension/#{CALLERIDNUM})
exten = 1000,4,Voicemailmain([EMAIL PROTECTED])
Then populate the DB via the asterisk CLI:
database put extension 1234567890 5432
This maps the phone with caller ID 1234567890 to extension 5432.
Scott
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RE: [Asterisk-Users] Auto connect to voicemail
I use an AGI script I wrote. It's specific to my setup, but you can get a copy at http://www.fnords.org/~eric/asterisk/downloads/ You'd have to adapt it to your own needs, of course. Basically it does this: When called with no options it strips off the first 6 digits of the CallerID if the Caller*ID is more than 4 digits. When called with a phone number as an option it replaces the Caller*ID number with the option, but only if the Caller*ID is 4 digits long. On Mon, 2004-04-05 at 17:53, Mark Hagler wrote: What do you do when $CALLERIDNUM of the caller isnt the 4-digit extension?I set all of my users Caller ID entries to their 10-digit phone # so that Caller ID appears correctly when I send their call out the PRI to the public network.The side effect of this is breaking convenient access to voicemail using this method, and I havent found a way to fix it yet. __ From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Glenn Dalgliesh Sent: Monday, April 05, 2004 3:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Auto connect to voicemail I think this is what you are looking for Exten = 1000,1,Answer,1 Exten = 1000,2,Wait,1 Exten = 1000,3,Voicemailmain([EMAIL PROTECTED]) - Original Message - From: Mitchell S. Sharp To: [EMAIL PROTECTED] Sent: Monday, April 05, 2004 5:27 PM Subject: Re: [Asterisk-Users] Auto connect to voicemail On Mon, 2004-04-05 at 15:57, Brian Rathman wrote: I have the voicemail setup working in that I get the MWI and it emails the message correctly. When I pressed the MWI button on my SNOM 200, it dials into the voicemail system and prompts me for a mailbox and password. I know there is a way to automatically connect directly into the mailbox via the extension.conf file, but I can not find the documentation I am looking for in reference to variables and macros for the extensions file. Can someone please help me with this issue? Thanks, Brian Brian, At the CLI, type 'show application VoiceMailMain'. You can use the CLI 'show applications' command to list all available apps. If you hit tab, it acts just like BASH's auto complete. Wonderful feature! Mitch Sharp Innovative Solutions -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SingTel ready to break into web telephony
http://www.smh.com.au/articles/2004/04/05/1081017104255.html SingTel ready to break into web telephony April 6, 2004 Singapore Telecommunications is teaming up with US internet phone start-up SIPphone to offer low cost, and in some cases free, phone services over the web. The deal, expected to be announced today, will allow SIPphone - started by MP3.com founder Michael Robertson - to route calls anywhere in the world over SingTel's global phone network. SingTel and SIPphone, based in San Diego, initially will market the service in Asia and try to strike deals with other regional internet service providers to sell packages of phone services. But the SingTel-SIPphone deal's reach will be global: it will allow anyone using special SIPphone gear, such as one of the company's phones or adaptors, to use a new type of SingTel calling card to place calls cheaply from anywhere, to anywhere. Customers could buy the calling cards online, not just in Asia, when the new service starts later this month, said Richard Tan, SingTel's vice-president of international carrier services. SingTel owns Optus and has sizeable stakes in telecom firms in other countries including India, Indonesia, Thailand and the Philippines. Until SIPphone linked up with SingTel, SIPphone users could use the service only by calling other people with one of the company's phones or adaptors, which plug into a high-speed internet outlet. Those calls are free, not counting the cost of the SIPphone hardware. SIP stands for session initiation protocol, the internet standard the technology uses. The partnership is another sign of the power of voice over internet protocol, or VOIP, technology, which is disrupting the business models of major phone companies and threatening to slash their profits. The technology transforms a voice on the phone into digital packets, which then travel over the web and are reassembled at their destination. Because they are carried over the internet, and not a traditional phone line, calls are either free or heavily discounted because they avoid many regulatory fees. By using VOIP, the difference between local and long-distance [calling] evaporates, said Mr Robertson.
Re: [Asterisk-Users] Asterisk Capacity
Hola Pablo, on the box you describe the maximum would be ZERO. You haven't mentioned what CT hardware you would like to use. Salud! On Tue, 2004-04-06 at 07:27, pesb wrote: And could anybody say the concurrent calls limit for one Asterisk Box? Let's say it is a Pentium IV 1.6GHz, 256 MB RAM, RedHat 9 thanks, Pablo Salinas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ben Kramer [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto connect to voicemail
Title: Message
Try
placing the following in your extensions.conf file:
exten =
1000,1,VoicemailMain(${EXTEN:6})
That
strips the first six number off of the Caller ID leaving the last four digits to
correspond with the voicemailbox. I've got it working on one of my
servers.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
HaglerSent: Monday, April 05, 2004 5:53 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Auto
connect to voicemail
What do you do when $CALLERIDNUM
of the caller isnt the 4-digit extension? I set all of my
users Caller ID entries to their 10-digit phone # so that Caller ID appears
correctly when I send their call out the PRI to the public network.
The side effect of this is breaking convenient access to
voicemail using this method, and I havent found a way to fix it yet.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Glenn DalglieshSent: Monday, April 05, 2004 3:26
PMTo:
[EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Auto
connect to voicemail
I think this is what you are
looking for
Exten =
1000,1,Answer,1Exten = 1000,2,Wait,1Exten =
1000,3,Voicemailmain([EMAIL PROTECTED])
- Original Message -
From: Mitchell S. Sharp
To: [EMAIL PROTECTED]
Sent: Monday,
April 05, 2004 5:27 PM
Subject: Re:
[Asterisk-Users] Auto connect to voicemail
On Mon, 2004-04-05 at 15:57, Brian Rathman wrote:
I have the voicemail setup working in that I get the MWI and it emails themessage correctly. When I pressed the MWI button on my SNOM 200, it dialsinto the voicemail system and prompts me for a mailbox and password. I knowthere is a way to automatically connect directly into the mailbox via theextension.conf file, but I can not find the documentation I am looking forin reference to variables and macros for the extensions file. Can someoneplease help me with this issue?Thanks,Brian
Brian,At the CLI, type 'show application
VoiceMailMain'. You can use the CLI 'show applications' command to
list all available apps. If you hit tab, it acts just like BASH's auto
complete. Wonderful feature!Mitch SharpInnovative
Solutions
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Re: [Asterisk-Users] ADPCM 4-bit, 6 kHz
Yves Chouinard wrote: I found some posts regarding this issue dating of September 2003, but no real answer. The ADPCM format supported by Asterisk (the .vox files) is 4-bit, 8 kHz. I need 4-bit, 6 kHz, which is also a widespread Dialogic format, to help migration. Is there an existing format/codec for this? If not, can I make myself a shared object in /usr/lib/asterisk/modules? Is this easy??? :-( Thanks, Yves Chouinard Vox-Tel There is no support for this right now. However, as you say, the OKI/Dialogic 24kbps format is very widely used in the IVR business. It might be worth having a 6k - 8K rate converter so the existing Dialogic 32kbps code can also work with 24kbps Dialogic files. P.S. Just saying 4-bit, 8 kHz or 4-bit, 6 kHz is meaningless. There are *many* ADPCM formats which fit those descriptions. The format is OKI ADPCM. Dialogic originally used OKI's ADPCM chips, although all recent cards implement the codec in a programmable DSP. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Buzzing on TDM400P FXS?
I had slightly different problems - but the resolutions might help you: I had a problem with an intermittent loud buzzing on my X100P (heard when accessing PSTN from my SIP and Zap clients). The problem went away when I physically moved the card down a PCI slot (further away from my TDM400P and my power supply). I also had an additional problem with jittery/choppy sound when receiving calls from PSTN on SIP devices. This was due to my X100P sharing interrupts with other devices (e.g. NIC). Cheers Ryan On 5-Apr-04, at 12:41 PM, Scott Laird wrote: I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for buzzing on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and IAX to NuFone, so I'm pretty sure that it's happening on the FXS end. Here's that chunk of zapata.conf: context=inside-analog signalling=fxo_ks callwaiting=yes callwaitingcallerid=yes cancallforward=yes callreturn=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=0 immediate=no musiconhold=yes usecallerid=yes callerid=Analog Phone 2201 mailbox=2201 channel = 2 Does anyone have any suggestions on where to start looking? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto connect to voicemail
I'm running into a similar situation. We have 3-digit extensions and a 4-digit DID numbers that get used for for outbound CID. Therefore, no $CALLERIDNUM direct access to voicemail. Suggestions? What do you do when $CALLERIDNUM of the caller isnt the 4-digit extension? I set all of my users Caller ID entries to their 10-digit phone # so that Caller ID appears correctly when I send their call out the PRI to the public network. The side effect of this is breaking convenient access to voicemail using this method, and I havent found a way to fix it yet. I think this is what you are looking for Exten = 1000,1,Answer,1 Exten = 1000,2,Wait,1 Exten = 1000,3,Voicemailmain([EMAIL PROTECTED]) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
