[Asterisk-Users] Multiple outgoing calls through VOIP providers
Trying to get some straight info from the VOIP providers is difficult. Say there's a small Asterisk switch and it's registered with Broadvoice or LiveVOIP or someone. There are a couple of people using the switch, one is on an outgoing call with the VOIP provider. What happens when someone else initiates another outgoing call through that provider on the same SIP registry? Does * know that the SIP account is busy or does it dial out anyway? Does the provider care? Do I establish a call group of SIP accounts like I would of Zap trunks and Dial/g1 ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forwarding to regular numbers?
I'm trying to set up extensions and have them forward to my cell phone or other phones I have and include them in call groups. I tried *72480204 and *7298480204, I get the recording that unconditional forwarding is set to that number.. but when I call that extension I just get silence and eventually it hangs up. Someone throw me a clue stick? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Best Headsets for a Call Center Environment
On Fri, 25 Mar 2005, Jessie Mabanglo wrote: You can look for Plantronics.. we're been using here in our call center for 2 years...they have a variety of models... Our call-center were allowed to test quite a few borrowed headsets. Sennheiser telecom headsets generally came up on top when considering audio quality and comfort. http://www.sennheisercommunications.com/pr-professional.html Their semi-closed design with directed microphones work very well when there are a lot of conversations in the same room. I think we mostly use the SH350. I have not seen any pc-headsets that are nearly as good from any manufacturer. If you must have the call center on softphones then get an adapter and use telecom headsets. Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth Sent: Friday, March 25, 2005 9:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Best Headsets for a Call Center Environment I'm looking for suggestions as to the best multimedia headsets for a call center environment. A few considerations: 1) USB headsets are preferable, because they don't require a soundcard. 2) Omnidirectional microphones are problematic, because they pick up too much background noise. Thanks, Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debia n ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.1 - Release Date: 3/23/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Newbie Instalation
I suggest you strip naked, do a war dance, and sacrifice chickens to the digium gods, and then i'm sure verything will work fine. If that doesn't work, do the same thing while standing on your head. Or, you could post some details of your installation so we have some faint idea of what might possibly be wrong, and then maybe we'll be able to help. -yair On Fri, 25 Mar 2005 13:44:02 +1030, Trevor Tregoweth [EMAIL PROTECTED] wrote: Hi All, This is my first attempt in setting up Asterisk, seems to be installed and running ok. I have installed on local pc x-lite, the phone interface program, but do you think I can get it to work, well no. So has anyone got any ideas on what I might be doing wrong, and helpful tips on getting ti going Thanks Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Newbie Instalation
If we need a dose of Smart Ass, it's always good to know it's available here on this list. The person is new and he is asking a question. You could have emailed him direct, asked for more detail and helped him. Rather than be kind you posted dribble. Daily I speak with people like Trevor. If you do not have something positive to say just do not say it. Brandon Patterson LiveVoip LLC I suggest you strip naked, do a war dance, and sacrifice chickens to the digium gods, and then i'm sure verything will work fine. If that doesn't work, do the same thing while standing on your head. Or, you could post some details of your installation so we have some faint idea of what might possibly be wrong, and then maybe we'll be able to help. -yair On Fri, 25 Mar 2005 13:44:02 +1030, Trevor Tregoweth [EMAIL PROTECTED] wrote: Hi All, This is my first attempt in setting up Asterisk, seems to be installed and running ok. I have installed on local pc x-lite, the phone interface program, but do you think I can get it to work, well no. So has anyone got any ideas on what I might be doing wrong, and helpful tips on getting ti going Thanks Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Converting 7905G to SIP
On Fri, 25 Mar 2005 17:33:17 +1000, Greg [EMAIL PROTECTED] wrote: I am trying to convert my 7905G to be SIP based and seem to be running into a few hassles. Below are all the config files and logs from the server. I have tried to follow the pdf's from cisco and some posts from other mailing lists that google turnedup, but it seems that nothing is working. Am I somehow missing a fundamental step in trying to upgrade from Call Manager to SIP? Any help is greatly appreciated. Regards, Greg That's a configuration file from the 7940/7960 series phones. The 7905G uses a totally different format, has its own firmware, etc. Do you have the SIP firmware for the 7905G? -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Newbie Instalation
Jeez, people...learn to take a joke. I offered to help the man. I didn't make cracks about googling or anything like that. I said explicitly that if he posted details we'd try to help. I didn't insult him, call him a worthness noob, or otherwise offend him. IT WAS JUST A JOKE. Is this different from the post a few days ago about not using enough magic? And, as for emailing people direct and offering to help them i was under the impression that it doesn't work that way...that there is value in the archives, and if he posts his setup and problems and a solution found then that helps everyone. To you, Trevor, if i said something offensive inadvertantly i apologize and i hope we can help you. To Brandon and Randall, i suggest you both try to see more humor and less insult. -yair On Fri, 25 Mar 2005 01:44:54 -0700, Brandon Patterson [EMAIL PROTECTED] wrote: If we need a dose of Smart Ass, it's always good to know it's available here on this list. The person is new and he is asking a question. You could have emailed him direct, asked for more detail and helped him. Rather than be kind you posted dribble. Daily I speak with people like Trevor. If you do not have something positive to say just do not say it. Brandon Patterson LiveVoip LLC I suggest you strip naked, do a war dance, and sacrifice chickens to the digium gods, and then i'm sure verything will work fine. If that doesn't work, do the same thing while standing on your head. Or, you could post some details of your installation so we have some faint idea of what might possibly be wrong, and then maybe we'll be able to help. -yair On Fri, 25 Mar 2005 13:44:02 +1030, Trevor Tregoweth [EMAIL PROTECTED] wrote: Hi All, This is my first attempt in setting up Asterisk, seems to be installed and running ok. I have installed on local pc x-lite, the phone interface program, but do you think I can get it to work, well no. So has anyone got any ideas on what I might be doing wrong, and helpful tips on getting ti going Thanks Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] peering
Our main asterisk box peers with that of a customer. We are trying to assign DID's to their extensions but get this error. What are we doing wrong? Client side Mar 25 18:49:47 NOTICE[1369]: chan_iax2.c:6545 socket_read: Rejected connect attempt from 203.xxx.xxx.16, who was trying to reach 's@' Our side Mar 25 18:56:15 WARNING[705]: chan_iax2.c:5546 socket_read: Call rejected by 203.xxx.xxx.17: No authority found -- Hungup 'IAX2/username/23' == No one is available to answer at this time ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Newbie Instalation
Agreed - So lets help the new Guys! Brandon Patterson LiveVoip LLC Jeez, people...learn to take a joke. I offered to help the man. I didn't make cracks about googling or anything like that. I said explicitly that if he posted details we'd try to help. I didn't insult him, call him a worthness noob, or otherwise offend him. IT WAS JUST A JOKE. Is this different from the post a few days ago about not using enough magic? And, as for emailing people direct and offering to help them i was under the impression that it doesn't work that way...that there is value in the archives, and if he posts his setup and problems and a solution found then that helps everyone. To you, Trevor, if i said something offensive inadvertantly i apologize and i hope we can help you. To Brandon and Randall, i suggest you both try to see more humor and less insult. -yair On Fri, 25 Mar 2005 01:44:54 -0700, Brandon Patterson [EMAIL PROTECTED] wrote: If we need a dose of Smart Ass, it's always good to know it's available here on this list. The person is new and he is asking a question. You could have emailed him direct, asked for more detail and helped him. Rather than be kind you posted dribble. Daily I speak with people like Trevor. If you do not have something positive to say just do not say it. Brandon Patterson LiveVoip LLC I suggest you strip naked, do a war dance, and sacrifice chickens to the digium gods, and then i'm sure verything will work fine. If that doesn't work, do the same thing while standing on your head. Or, you could post some details of your installation so we have some faint idea of what might possibly be wrong, and then maybe we'll be able to help. -yair On Fri, 25 Mar 2005 13:44:02 +1030, Trevor Tregoweth [EMAIL PROTECTED] wrote: Hi All, This is my first attempt in setting up Asterisk, seems to be installed and running ok. I have installed on local pc x-lite, the phone interface program, but do you think I can get it to work, well no. So has anyone got any ideas on what I might be doing wrong, and helpful tips on getting ti going Thanks Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Advanced Cisco 7960 Config
On Thu, 24 Mar 2005 21:26:27 -0800, Max Clark [EMAIL PROTECTED] wrote: Hi all, Good evening I have a working (it was a pain) set of Cisco 7960 phones. In order to dial I have to either pick up the handset or select the line and then dial the extension or outside line. How do I configure the dialplan so I can: - Start dialing via the keypad and have the phone automatically go to speaker on the first line? The 7960 doesn't have a hot keypad (the cheaper and less featured in other ways 7905G/7912G phones do though - go figure). You need to press Speaker first. - Give the user dialtone after they dial '9'? In your dialplan, add a , after 9. eg: TEMPLATE MATCH=9,.* Timeout=3 User=Phone/ A while ago I found a cool asterisk/penguin logo to use on the phone, can anyone point me to a place I can download this again? Wouldn't have a clue, but would also like to know :) -Shaun Thanks in advance, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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/iax routing question
What happens if a SIP call is routed through more than one * server? If canreinvite=yes for all the peers involved, and t or T is not used in the Dial command, then the audio would get routed directly between the endpoints. Also, when setting up an inter asterisk exchange, is all the data routed through the * servers? As long as notransfer=no for all the peers involved, then everything but the endpoints would completely drop out of the call. Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Advanced Cisco 7960 Config
http://www.loligo.com/asterisk/Cisco/79xx/current/asterisk-tux.bmp On 25/03/2005, at 7:01 PM, Shaun Ewing wrote: A while ago I found a cool asterisk/penguin logo to use on the phone, can anyone point me to a place I can download this again? Wouldn't have a clue, but would also like to know :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Newbie Instalation
Yair Hakak wrote: Jeez, people...learn to take a joke. Use your smiley button, then. Lots of people have pretty short fuses on both sides of this issue, and it's well to avoid ambiguity whenever possible. Your post was ambiguous in that respect, where for three extra keystrokes you could have made your humorous intent crystal-clear. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Outlook contacts - Asterisk database(LookupCIDName)
Thanks! That looks like what I need. I just want caller ID to appear on every handset. I have wireless phones too and (fortunately) there is no Outlook on those phones :) but I would like CIDName. Cheers! Remco On Thu, 24 Mar 2005, Jay Milk wrote: Export Outlook to CSV, import name and numbers to mysql, and use cid_rewrite located at http://muware.com/asterisk That's what I did. -Original Message- From: Remco Barende [mailto:[EMAIL PROTECTED] Sent: Thursday, March 24, 2005 1:28 PM To: Asterisk Users List Subject: [Asterisk-Users] Outlook contacts - Asterisk database(LookupCIDName) Is it possible in any way to use an Outlook contacts database as the source for the internal Asterisk database that is used for callerid lookups? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Eicon DIVA PCI ISDN cards (not server) work with asterisk!
On Thu, Mar 24, 2005 at 05:36:35PM +0200, Mark Elkins wrote: I am still curious. Which Driver do you use for the HFC card? I manage my own Debian package repository for Debian stable (woody) backports of asterisk and related stuff (based somewhat on backports.org). I currently use: Version: 1:1.0.6-0.bristuff_0.2.0_RC7k.cril.0 so it's about the same (I just have a few additional patches). It could be: bristuff-0.2.0-RC7k stuff from http://www.junghanns.net/ - but this locks you into using a particular - non-HEAD version of Asterisk.. (and missing all the new goodies) not really, AFAIK the last time I tried, the BRI patches apply cleanly also to more recent versions. And 1.0.6 works quite well for me. The only problem I still have with 1.0.6 is that for some reason, IAXcomm user tell me that when they get a call, and answer it, then after 30 seconds or 1 minute a new call comes in (which is fake) and you have to cancel it in IAXcomm to get the first call correctly. I haven't debugged it yet. I have SIP phones, IAX2 connections to remote Asterisk, ISDN bidirectionnal gateway, analog TDM board with el cheapo analog tel, DECT CLIP-compatible phone (works), and also ISDN local phones (using HFC NT mode). I wish there were single, four and eight port ISDN BRI cards that Digium sold and supported - so I could run whichever version of Asterisk I wanted... ISDN was never popular in the US for BRI lines. In Europe we even do stupid things such as multiple-BRI operated in cascade (e.g. 4 BRIs, giving you the equivalent of 8 communication channels), where it would be more intelligent to use (partial) E1 for that purpose. I think Germany has those partial E1 available to the public. In the US, people usually do analog upto 10 lines and then get a T1. As analog lines include caller ID (however AFAIK no easy ability to *set* outgoing caller ID nor real calle*d* ID, without distinctive ringing), most benefits of BRI ISDN are unneeded. That's why most BRI ISDN development is done in Europe -- or more precizely looks like it's Germany, really. An alternative to zaptel is to use the m_isdn implementation of the Linux kernel. As I use 2.4 and it works very well with zaptel/zaphfc, I didn't bother to try the 2.6 (crappy) kernels or the 2.4-m_isdn backport yet. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Backup for linux/asterisk
What about imaging? We use acronis true image 8.0. You can create an image of your asterisk box within 20 minutes (120 GB HD !) and deploy it to another server in the same time. Even if changing your hardware from VIA to SIS and back to INTEL wasn't a problem for us. Btw we use Fedora Core 2 for our * servers. Regards, Guido Hecken after getting my feet wet with [EMAIL PROTECTED], I want to set up a second asterisk box to add a call shop billing and other add-ons such as LCR. My question is as follows. Is there a backup program that will save to a tape drive or a USB CD Writer so if I mess up an install I don't have to go through a complete reinstall? I saw a few programs out there but they required X windows and from what I read it is suggested that X windows not be installed on an Asterisk box. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Newbie Instalation
Hi All, Thanks for the wonderful advice, and comments, and anything I might of missed, and no offence taken. Yes I am new to this program, and Linux too, so this is a big learning curve. I installed the software Asterisk which I believed it did straight from the cd, rebooted the computers, and it installed more stuff. I am then lead to believe that I can use x-lite a phone interface, I guess, to interact with the new pabx-asterisk system I now have. I can see from the gui interface that I am trying to make calls, but that's about it, not much else is happening Well if you want to know much more, please ask, as I have no idea what I am doing :) You help and direction would be much appreciated. Cheers Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie questions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, I've some questions about asterisk, and in general about voip, please help me :) 1. I've SIP accounts on external servers, and I would that my local server will connect with those and redirect all calls from those to an internal SIP account (just one). It's possible to do that? In this case, I think asterisk will work as UA for external accounts, and as sip server for internal. I've to use SER with asterisk? 2. the internal account it's important that will be SIP, or I could forward calls from my external sip account to an h323 account? 3. I could configure a voicemail account (with an internal number) for all calls that I would redirect from all internal phones? 4. I could use a welcome message on an internal account, and/or auto attendant? I hope this is clear. Any advice to put me in the right direction will be appreciated. Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCQ+AHMakHrsrHP9wRAmbDAJ428+4F+R/RSv0CGMVZVwo73z1OAwCgoiDe F6eXRsp/JX4QD78tDE9Jiro= =Zx81 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Emailed voicemail
Richard, Yep, got that config'd in there: 1001 = 1001,Andy Stewart,[EMAIL PROTECTED] 1002 = 1002,Lorri Barnett,[EMAIL PROTECTED] 1003 = 1003,Andy Stewart - Home,[EMAIL PROTECTED] 1004 = 1004,Andy Stewart - HardPhone,[EMAIL PROTECTED] 1005 = 1005,Lorri Barnett - HardPhone,[EMAIL PROTECTED] Or it this maybe the problem? Your example is ext = ext,emailMine above (and the example in voicemail.conf) is ext = ext,name,email ?? Thanx A From: Richard J. Sears [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Emailed voicemail To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Hi Andy, did you configure voicemail.conf with the users e-mail address...? 1234 = 1234,[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Dynamically limiting the number of outbound calls
Jim Singh wrote: In our setup, outbound call volume frequently exceeds the line capacity of the DSL line. We do not want to move to another codec to better utilize the line, but instead wish to automatically divert overflow to the Long Distance T1 when the DSL is full. Ideally the system would also be able to adjust automatically to network conditions such as network outage, high latency, jitter and/or packet loss. If the LD T1 was also full or if there was no other path, Dial would return busy/congestion instead of connecting a call of low quality. I realize that one solution is to manage variables using macros in the dialplan and keep a count of VOIP calls. I believe that this a) difficult to maintain b) can be difficult to dynamically adjust based on parameters from the jitter buffer, round trip time, and/or packet loss c) couldn't be the best way to do it. Before I go slinging code, does anyone know of a clean solution? Do other people need / desire this functionality? Our Setup: Software: Suse 9.2 + Asterisk 1.0.7 (built from CVS) Network: DSL measured to be 2 mbps up / 430 kbps down Termination: IAX2 / G711 / nufone and voipjet Zaptel: 2 digium 100 cards one connected to a Siemens PBX and the other to a (long distance) provider,signaling is EM Wink I'm from an SNMP background, so thats the way I'd be looking. Something like: Enable SNMP on your DSL router Select your scripting language with SNMP support (scotty, shell with net-snmp, python,perl whatever) Write a script that queries the router, checking 1) outbound queue lenght, 2) outbound packets/sec 3) interface status 4) dropped udp packets also perhaps the ping your VOIP provider check to see if all of the above are within acceptable limits (tweaking required) Get the script to set a value in the asterisk db based on the result of the check. get cron to call the script every 60 secs (or whatever) In your dialplan check the value of the variable and dial outbound calls accordingly. Tim. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] * - SMS w/out PSTN
Mark Charlton wrote: Hi all I want to send an SMS message whenever I get a voicemail left on my [EMAIL PROTECTED] 0.6 box. I don't have any pstn attached the the box, and I am running FWD, voipuser, and alg as providers for various routes and redundancy. I can find a number of providers for sending SMS via pstn to BT, but nothing to reliably use online SMS services. I saw a few for outbound that were in beta. Basically can anyone recommend an SMS client for internet only usage in the UK. We have been using simplewire (www.simplewire.com), they may not be the cheapest, but the api is ok, and I've never had a problem with them. They are US based, but that seems to make little or no difference. Many thanks Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Newbie Instalation
Hi, a few pointers: 1. the wiki is your friend: http://www.voip-info.org/wiki-Asterisk lots of good stuff and good documents for getting started. If i were you i might reinstall asterisk from CVS just to make sure you have the latest version, and because this way you can learn about installing. In general asterisk usually runs on a machine without x-windows so doing everything from a command-line window might also help with the learning, rather than being dependent on prepackaging. 2. did you install asterisk with the example configs (make samples)? asterisk is dependent on a directory of config files (usually /etc/asterisk) and make samples populates the dir with some basic config files. 3.make sure that the x-lite that you want to register is defined in sip.conf and extensions.conf 4. configure x-lite using the following: http://www.astmasters.net/stuff/X-Lite-and-Asterisk.pdf 5. run asterisk in verbose mode (asterisk -r vgc) so you can see what's happening You're in for quite a learning experience...hope this has been helpful. good luck and good hunting. -yair On Fri, 25 Mar 2005 20:21:57 +1030, Trevor Tregoweth [EMAIL PROTECTED] wrote: Hi All, Thanks for the wonderful advice, and comments, and anything I might of missed, and no offence taken. Yes I am new to this program, and Linux too, so this is a big learning curve. I installed the software Asterisk which I believed it did straight from the cd, rebooted the computers, and it installed more stuff. I am then lead to believe that I can use x-lite a phone interface, I guess, to interact with the new pabx-asterisk system I now have. I can see from the gui interface that I am trying to make calls, but that's about it, not much else is happening Well if you want to know much more, please ask, as I have no idea what I am doing :) You help and direction would be much appreciated. Cheers Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Error cannot record voicemail
On Thu, Mar 24, 2005 at 01:30:25PM -0500, Joel Duffield wrote: I tried to share my spool directory so I could get monitored calls, and now this error comes up when I try to leave a message in any of my voicemail boxes. Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v oicemail/default/300/INBOX/msg.WAV: No such file or directory When you get the error No such file or directory when opening a file for writing it generally means that one of the preceeding directories doesn't exist. So check if the directory /var/spool/asterisk/voicemail/default/300/INBOX/ exists. I beleive the Voicemail app creates the directories itself so if the directory doesn't exist, it can't create them. Make sure you're the same user as asterisk is running as and try: mkdir -p /var/spool/asterisk/voicemail/default/300/INBOX/ Maybe one of the components has been replaced by a file or directory with bad permissions Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] atxfer
Hi list, This wll be my first post, so I want to thank all the developers for the great product they have created. Now, the question, I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) This all works fine, exept for som echo on the ISDN channel, but I'll replace the I4L card with an AVM-C4 card next week I am trying to get supervised/ attended tranfer working, blind transfer by pressing the # key works fine this is my features.conf [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 600 ; Number of seconds a call can be parked for ; (default is 45 seconds) transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call adsipark = yes ; if you want ADSI parking announcements pickupexten = *8; Configure the pickup extension. Default is *8 [featuremap] blindxfer = # atxfer = * I have tried atxfer=# and atxfer=*2 but nothing works, also, when I disable both blindxfer and atxfer, blind transfer still works. I have turned sip debug on, and I can see that asterisk does receieve the * keys parking calls does work btw. Kind regards, Joop ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does IAX supports silence suppression?
Hi, Does IAX supports silence suppression? If yes, is there any way to detect that the other party has turned on silence suppression and there is no packet loss? Is (Halt|Reasume) audio/video transmission control messages used for this reason? Regards, Marcin Okraszewski ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hello Everyone
would like to test this e-mail list. anyway, have anybody here install and run [EMAIL PROTECTED] how was it? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Does IAX supports silence suppression?
Short answer is no. You should always turn it off on any client you have. Longer answer is that is is being worked on and should be available any day now (although that has been the case for some months). Also someone is working on porting it to SIP as well as IAX2. No idea if the new work will tell your if the client is using silence suppression. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marcin Okraszewski Sent: 25 March 2005 12:14 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Does IAX supports silence suppression? Hi, Does IAX supports silence suppression? If yes, is there any way to detect that the other party has turned on silence suppression and there is no packet loss? Is (Halt|Reasume) audio/video transmission control messages used for this reason? Regards, Marcin Okraszewski ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP issue
Hello List, I'm trying to setup MGCP channel with a Centile Media Hub box. My Centile box has 4 ports and I got no dial tone. Can somebody help with this isuue? This is my mgcp.conf and extensions.conf Thanks Daniel. ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.11.20 disallow=all allow=g729 allow=alaw allow=ulaw [192.168.11.200] context=MGCP host=192.168.11.200 wcardep=aaln/* callerid = test 8000100 callwaiting=no transfer=no cancallforward=no dtmfmode=rfc2833 canreinvite=no singlepath=no slowsequence=yes line = aaln/1 callerid= test 8000101 callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line = aaln/2 callerid= test 8000102 callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line = aaln/3 callerid= test 8000104 callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line = aaln/4 extensions.conf [MGCP] include = Toll Free include = CreoLink exten = 8000100,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000100,2,Hangup exten = 8000101,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000101,2,Hangup exten = 8000102,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000102,2,Hangup exten = 8000103,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000103,2,Hangup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Optional URL in App. Queue
+++ Dan [20/03/05 09:17 +0200]: Hi James, - Original Message - From: James Coberly [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 19, 2005 11:41 PM Subject: Re: [Asterisk-Users] Re: Optional URL in App. Queue There appears to be a bug in 0.9.10f. was great in 0.9.10e and is working great in the 0.9.10g I have been testing for Dan. I'll push him a message and see if he can update it to the web site. James- A newer version (0.9.11a) is under testing now. I hope to be able to post it on my site later today. Best regards, Dan I am using 9.10g and yes the Url option with the Asterisk DIAL command works great but not the URL option with the asterisk QUEUE command any help with that ? -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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-write statement
I have some numbers, which should be treated equally. To avoid double coding, I would like that this extension could be re-written. E.g., some users are used to dial 002 ~ 009 as international prefix, while I have choosen to use the USA way (011). It would be nice if the user can dial 9-002-43-456-3456-7890 and it would be re-written to: 9-011-43-456-3456-7890 and 9-02-2345-6789 as 9-011-886-2-2345-6789 and than with a Goto statement to the right place. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Newbie Instalation
Hi, Thanks for the pointers, I will do a reinstall and see how I go. I am presuming that this program will work with just the x-lite and no others phone related hardware. I am just wanting to get it working from pc to pc to start with, then will attach the next step of going to a live phone line Cheers Trevor -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak Sent: Friday, 25 March 2005 8:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Instalation Hi, a few pointers: 1. the wiki is your friend: http://www.voip-info.org/wiki-Asterisk lots of good stuff and good documents for getting started. If i were you i might reinstall asterisk from CVS just to make sure you have the latest version, and because this way you can learn about installing. In general asterisk usually runs on a machine without x-windows so doing everything from a command-line window might also help with the learning, rather than being dependent on prepackaging. 2. did you install asterisk with the example configs (make samples)? asterisk is dependent on a directory of config files (usually /etc/asterisk) and make samples populates the dir with some basic config files. 3.make sure that the x-lite that you want to register is defined in sip.conf and extensions.conf 4. configure x-lite using the following: http://www.astmasters.net/stuff/X-Lite-and-Asterisk.pdf 5. run asterisk in verbose mode (asterisk -r vgc) so you can see what's happening You're in for quite a learning experience...hope this has been helpful. good luck and good hunting. -yair On Fri, 25 Mar 2005 20:21:57 +1030, Trevor Tregoweth [EMAIL PROTECTED] wrote: Hi All, Thanks for the wonderful advice, and comments, and anything I might of missed, and no offence taken. Yes I am new to this program, and Linux too, so this is a big learning curve. I installed the software Asterisk which I believed it did straight from the cd, rebooted the computers, and it installed more stuff. I am then lead to believe that I can use x-lite a phone interface, I guess, to interact with the new pabx-asterisk system I now have. I can see from the gui interface that I am trying to make calls, but that's about it, not much else is happening Well if you want to know much more, please ask, as I have no idea what I am doing :) You help and direction would be much appreciated. Cheers Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Hello Everyone
Welcome, Yes I have used it. It's great to get started. Give it a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bagan Jermal Sent: Friday, March 25, 2005 6:34 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hello Everyone would like to test this e-mail list. anyway, have anybody here install and run [EMAIL PROTECTED] how was it? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream firmware update 1.0.5.23
Version 1.0.5.23 is now available from http://gs-firmware.gratissip.dk/ Or directly from Grandstream at http://www.grandstream.com/BETATEST/Release-b21p1.0.5.23.zip Release notes doc here http://www.grandstream.com/BETATEST/Release_Note_1.0.5.23.doc while on the matter I just want to extend a note of thanks to Grandstream, I had 2 early handsets of theirs fail recently (about 9 months old) when I was unable to return them to the dealer I bought them from they organized for me to rma them directly. 2 brand new grandstreams now sitting on my desk. Has anyone else noticed that they have changed not only the plastic composition of the handsets but also the design of the handpiece itself (slightly thinner and slightly heavier) I could be wrong but to me it sounds like the voice quality has improved between the older model and the newer one so slightly. And for $50 or there abouts you cant complain. Just my 0.02c worth. Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 parsing unusual SIP/SDP
Stewart Nelson wrote: How should I proceed? IMO, this provider offers an excellent combination of price, reliability, quality, and support, and I believe that many in Asterisk community would want to use them. AFAICT, their SIP/SDP does not actually violate any RFCs. The next step would to be turn pedantic=yes back on, then generate a failing call with 'sip debug', 'set verbose 255' and 'set debug 255' in place. Capture all the output (there will be a lot) and then post a bug in Mantis describing the situation and attaching the output file. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] atxfer
Look at the options for the dial command on the wiki, you have to use t or T or calls are not eligible to be transferred. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Friday, March 25, 2005 3:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] atxfer Hi list, This wll be my first post, so I want to thank all the developers for the great product they have created. Now, the question, I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) This all works fine, exept for som echo on the ISDN channel, but I'll replace the I4L card with an AVM-C4 card next week I am trying to get supervised/ attended tranfer working, blind transfer by pressing the # key works fine this is my features.conf [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 600 ; Number of seconds a call can be parked for ; (default is 45 seconds) transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call adsipark = yes ; if you want ADSI parking announcements pickupexten = *8; Configure the pickup extension. Default is *8 [featuremap] blindxfer = # atxfer = * I have tried atxfer=# and atxfer=*2 but nothing works, also, when I disable both blindxfer and atxfer, blind transfer still works. I have turned sip debug on, and I can see that asterisk does receieve the * keys parking calls does work btw. Kind regards, Joop ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] peering
Looks like you do not have the context set correctly in your iax.conf on both sides. Make sure that it exists and it is going to do what you want it to do. On Fri, 25 Mar 2005 18:57:06 +1000 AS [EMAIL PROTECTED] wrote: Our main asterisk box peers with that of a customer. We are trying to assign DID's to their extensions but get this error. What are we doing wrong? Client side Mar 25 18:49:47 NOTICE[1369]: chan_iax2.c:6545 socket_read: Rejected connect attempt from 203.xxx.xxx.16, who was trying to reach 's@' Our side Mar 25 18:56:15 WARNING[705]: chan_iax2.c:5546 socket_read: Call rejected by 203.xxx.xxx.17: No authority found -- Hungup 'IAX2/username/23' == No one is available to answer at this time ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re: Emailed voicemail
Yes Andy - that was my mistake. I have my system hacked up to do some other things. It should be: 1234 = 1234,Bob Jones,[EMAIL PROTECTED] do your mail logs have any errors at all in them in regards to mail bouncing or anything like that..? Do you have your servermail settings configured in voicemail.conf and did you (maybe) compile asterisk to use asterisk_vm mysql db instead of the voicemail.conf..? On Fri, 25 Mar 2005 04:55:48 -0500 Andy Stewart [EMAIL PROTECTED] wrote: Richard, Yep, got that config'd in there: 1001 = 1001,Andy Stewart,[EMAIL PROTECTED] 1002 = 1002,Lorri Barnett,[EMAIL PROTECTED] 1003 = 1003,Andy Stewart - Home,[EMAIL PROTECTED] 1004 = 1004,Andy Stewart - HardPhone,[EMAIL PROTECTED] 1005 = 1005,Lorri Barnett - HardPhone,[EMAIL PROTECTED] Or it this maybe the problem? Your example is ext = ext,emailMine above (and the example in voicemail.conf) is ext = ext,name,email ?? Thanx A From: Richard J. Sears [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Emailed voicemail To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Hi Andy, did you configure voicemail.conf with the users e-mail address...? 1234 = 1234,[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Major problems with TDM400 and specific telephones: suggestions?
Hello. Sorry for my bad english. I'm a french guy. I have the same problems with siemens dect phones S100 The caller id don't work on tdm... In France, the CID is differant than other country. Then standard ring cadence is: 1500 3000 1500 3 and so... The etsi standard (used in France) say: The short ring is called Ring Pulse Alert Signal (RP-AS) . From ETSI EN 300 659-1 V1.3.1 (2001) : 200ms RP-AS duration 300ms 500ms RP-AS to FSK 800ms 200ms FSK to FSK 500ms Try adding cadence=250,1500,1500,3000,1500,3000 In zapata.conf And use in extension.conf exten = 200,1,Dial(Zap/1r1,20,tr) In my case, it's ok with siemens dect Send me your feedback. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hardware question
Hello I want to to know if the motherboards VIA are fully supporte by asterisk. And also, some of those motherboars say that with 1 pci slot , using a special riser card you can connect 2 pci cards. Will that work to have 2 pci cards (FXS or FXO ) on asterisk? thank you Fabian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Dial Out??
Hi Noah - I've managed to get my asterisk server up and running with a single POTS line and a polycom IP500. It will happily answer the phone line, tranfer calls, voicemail, etc. The problem comes when I pick up the polycom phone and want to place an outside call. If I dial 913237773456 it just gives me a fast busy before I can enter the last digig If I just dial 97773456 it waits a second or two and then gives me the same fast busy. I have a Digium card running the Zaptel inerface. I know it works because * will answer incomming calls fine. I am using the sample extensions.cfg file that came with asterisk. I only changed the following line: TRUNK=Zap/1 The sample config changes from time to time, and is not always particularly useful for a working asterisk install. You're going to need to write it yourself. 1. You'll need to see what's going on when you dial. To do so, run asterisk like this: asterisk -vvrgc That will give you enough detail on the console that you can see everything that asterisk is doing. 2. What do your current outgoing exten statements look like? 3. Take a look on the WIKI at how to configure extensions.conf: http://www.voip-info.org/wiki-Asterisk+config+extensions.conf If that doesn't answer your questions, do a thorough search of the list with google: search terms site:lists.digium.com Thanks, Noah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web based Asterisk management tool
I do, like I do with my IAX2 softphone. It's just that I haven't tookthe time to make a webpage that explains what it does and provide alink to download it.I already send it to peoples on this list that asked for it.Anybody want it, just email (privately, since this list is already pretty busy)As soon as I have some free time, I'll do a page for it, I promiseSharing is caring ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web based Asterisk management tool
Sorry about the previous post. Is this still available? The main thing is I need a management tool I can use in commercial sales. Regards, Chris [EMAIL PROTECTED] Original Message I do, like I do with my IAX2 softphone. It's just that I haven't took the time to make a webpage that explains what it does and provide a link to download it. I already send it to peoples on this list that asked for it. Anybody want it, just email (privately, since this list is already pretty busy) As soon as I have some free time, I'll do a page for it, I promise Sharing is caring ;)___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Backup for linux/asterisk
I gave up on tape as being a nightmare to maintain, I now back all my servers and workstaions using backuppc. One linux server with a 5 device RAID can easily backup 100 workstatons and several servers beacuase of the pooling system used. For a smaller situation I would use 2 disks in RAID1 (mirror). Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Backup for linux/asterisk
On 25 Mar 2005, at 14:35, Chris Mason wrote: I gave up on tape as being a nightmare to maintain, I now back all my servers and workstaions using backuppc. One linux server with a 5 device RAID can easily backup 100 workstatons and several servers beacuase of the pooling system used. For a smaller situation I would use 2 disks in RAID1 (mirror). I don't know about you, but our business insurance requires us to have 'an up to date backup stored off site'. They won't cover us for disaster (fire,theft etc) losses unless I do. Check your policy before you go for any 'non-removable' backup. Tim. http://www.westhawk.co.uk/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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-addons and 64bit make
Hi all, I have an Athlon 64 server with Fedora Core 2 x86_64. When I try to make asterisk-addons-1.0.7 (and olders) it can't and it say me [EMAIL PROTECTED] asterisk-addons-1.0.7]# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons-1.0.7/format_mp3' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o common.o common.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o dct64_i386.o dct64_i386.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o decode_ntom.o decode_ntom.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o layer3.o layer3.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o tabinit.o tabinit.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o interface.o interface.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o format_mp3.o format_mp3.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o /usr/bin/ld: common.o: relocation R_X86_64_32 can not be used when making a shared object; recompile with -fPIC common.o: could not read symbols: Bad value collect2: ld returned 1 exit status make[1]: *** [format_mp3.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk-addons-1.0.7/format_mp3' make: *** [format_mp3/format_mp3.so] Error 2 What can i do? Thanks all, Daniele Gallina -- Daniele Gallina 3P System S.r.l. - Software Developer Web: http://www.3psystem.net E-Mail: [EMAIL PROTECTED] Tel: 041.8626401 Scelta 2 Fax: 041.5161655 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Forwarding to regular numbers?
Message: 16 Date: Fri, 25 Mar 2005 01:06:21 -0700 From: JD [EMAIL PROTECTED] Subject: [Asterisk-Users] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed I'm trying to set up extensions and have them forward to my cell phone or other phones I have and include them in call groups. I tried *72480204 and *7298480204, I get the recording that unconditional forwarding is set to that number.. but when I call that extension I just get silence and eventually it hangs up. Someone throw me a clue stick? JD -- To call forward from an extension you call the server from the extension: *72 You'll be prompted to enter the extension, 257# You'll be asked where to send the incoming call. your outbound zap*your cell number So, if you dial a 9 for your outbound ZAP calls and your cell number is 123-456-7890 it would look like this 9*123-456-7890. And anytime anyone calls your extension it will forward to your cell phone. Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What is web login password for Asteirsk@Home
Hello I have setup [EMAIL PROTECTED] and can login to the system via the asterisk box. But if I try same username and password to login using the Asterisk Management Portal I try the same username and password and cannot login. says authorization failure. I have tried from a Windows 2000 and a Windows XP machine running Internet Explorer v6. What am I doing wrong?Angus ComberItel Office Software Ltd5 Enmore GardensLondon, SW14 8RFTel: 020 8878 7367Fax: 020 8876 7257Em: [EMAIL PROTECTED]web: www.iteloffice.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] RE: Forwarding to regular numbers?
On Friday 25 March 2005 10:09, Jeff R Glassman wrote: Message: 16 Date: Fri, 25 Mar 2005 01:06:21 -0700 From: JD [EMAIL PROTECTED] Subject: [Asterisk-Users] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed I'm trying to set up extensions and have them forward to my cell phone or other phones I have and include them in call groups. I tried *72480204 and *7298480204, I get the recording that unconditional forwarding is set to that number.. but when I call that extension I just get silence and eventually it hangs up. Someone throw me a clue stick? Hmm, I'm not sure if I get what you are asking for correctly but this is what I do. One line dials my extension with a timout value like 20 sec. One could put a message saying trying next extension, please wait. Next line dials my cell with a timeout value like 20 sec. Last line (timeout) goes to VM. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Multiple outgoing calls through VOIP providers
David Josephson wrote: Trying to get some straight info from the VOIP providers is difficult. Say there's a small Asterisk switch and it's registered with Broadvoice or LiveVOIP or someone. There are a couple of people using the switch, one is on an outgoing call with the VOIP provider. What happens when someone else initiates another outgoing call through that provider on the same SIP registry? Does * know that the SIP account is busy or does it dial out anyway? Does the provider care? Do I establish a call group of SIP accounts like I would of Zap trunks and Dial/g1 ? Depends on the provider. voicepulse allows up to 4 outgoing and 4 incoming connections on their connect service, which is iax2 by the way. Highly recommended for multiple calls. I imagine if I hit that limit, the call will fail and look for the t or i extension and run that. I can only advise that you try it and watch the console. It's usually pretty clear what's happening. You can then script the extension that it's trying to jump to. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] What is web login password for Asteirsk@Home
Title: Message i believe the login for AMP is username:maint and password:password. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angus ComberSent: Friday, March 25, 2005 8:08 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] What is web login password for [EMAIL PROTECTED] Hello I have setup [EMAIL PROTECTED] and can login to the system via the asterisk box. But if I try same username and password to login using the Asterisk Management Portal I try the same username and password and cannot login. says authorization failure. I have tried from a Windows 2000 and a Windows XP machine running Internet Explorer v6. What am I doing wrong? Angus ComberItel Office Software Ltd5 Enmore GardensLondon, SW14 8RFTel: 020 8878 7367Fax: 020 8876 7257Em: [EMAIL PROTECTED]web: www.iteloffice.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] What is web login password for Asteirsk@Home
Type help-ahh from the console and you will be able to change logins etc. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angus Comber Sent: Friday, March 25, 2005 10:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] What is web login password for [EMAIL PROTECTED] Hello I have setup [EMAIL PROTECTED] and can login to the system via the asterisk box. But if I try same username and password to login using the Asterisk Management Portal I try the same username and password and cannot login. says authorization failure. I have tried from a Windows 2000 and a Windows XP machine running Internet Explorer v6. What am I doing wrong? Angus Comber Itel Office Software Ltd 5 Enmore Gardens London, SW14 8RF Tel: 020 8878 7367 Fax: 020 8876 7257 Em: [EMAIL PROTECTED] web: www.iteloffice.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] wrapuptime + agents.conf
voip technocrat [EMAIL PROTECTED] wrote: my aim is every call needs have wrapuptime of 5000 ms but when ever a call comes its directly connecting not wating any more. your views will be highly regarded with regards I'm using AddQueueMember.. for me, wrapuptime only seems to work from the beginning of the call, not the end. However, I'm defining wrapuptime in queues.conf, not agents.conf, since my agents are dynamic. I know in agents.conf it says that wrapuptime is in MS, but from what I can tell, if you define it in queues.conf, it's seconds. If the call exceeds the length of the wrapuptime value, there is no wrapuptime. So if you set it to 120, and the call only lasts a minute, the next caller will wait another minute before being connected. Likewise, if you set it to 120, and the call runs 135 seconds, there's no wrapuptime. I'd say that's probably a bug.. At this point it's strictly an annoyance, but I'd love to hear suggestions from the list. --Sean -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- sean a. newton [EMAIL PROTECTED] louisville, ky, usa http://wewt.net Another day, another convertible and another hotel full of cops.-- Hunter S. Thompson -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial command problem(VOIP+*+TDM400P+Legacy PBX)
Hello,
I just setup the Asterisk to integrate with Panasonic legacy PBX. Config as followings,
PSTN -- PanasonicPBX--TDM400P(FXO)--AsteriskPC-- Internet
* is for AA / Voicemail and VOIP in/out
Currently the AA / Voicemail function for incoming PSTN callsare working well.
My problem is for the incoming VOIP call. It can ring my internal extensions and talk without problem.
But it's all :-( I can not monitor the calling progress and handle it with * because of something wrong with
the Dial command. For example,ring forever if nobody answer the call, the call just be disconnected if
called extension busy, etc.
The code is as followings. Incoming VOIP call get into [via-net], then go to [test] to dial the extension.
From the console, I can see the Dial command is excuted and always stay there if nobody answer the call.
1.Why no dial timeout? I doubt if it's because the TDM400P FXO is connected to the extension port of
Panasonic PBX, so that it's recognized as answeredjust while DTMF tone is sent. Is it true?
2.MusicOn Hold can be heard but stop right away while extension is ringing.Seems answered even still
nobody pickup the phone.
3.My Panasonic can sent the DTMF tone to indicate the status(called extension is ringback, busy, etc)
I have used this good function to finishAAfor incoming PSTN call. But here I cando nothingsince it stay
on the Dial application andnot continue (if in Background, I can detect the tone).
I even use the option M(macro) try to catch the tone send from Panasonic,but failed also. Anybody can
give me your comment? Thanks!
BR, Dominic
[via-net]
exten = _1XX,1,Answer
exten = _1XX,2,SetVar(called_ext=${EXTEN})
exten = _1XX,3,Goto(test,s,1)
[test]
exten = s,1,Background(transfer)
exten = s,2,Dial(Zap/g1/${called_ext}|10|m)
exten = s,3,NoOp
--
Message on console while FOREVER ringing,
-- Executing Dial(IAX2/[EMAIL PROTECTED]/1, Zap/g1/102|10|m) in new stack
-- Called g1/102
-- Started music on hold, class 'default', on IAX2/[EMAIL PROTECTED]/1
-- Zap/1-1 answered IAX2/[EMAIL PROTECTED]/1
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/1
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Re: [Asterisk-Users] atxfer
On Fri, 25 Mar 2005 11:54:21 +0100, asterisk [EMAIL PROTECTED] wrote: I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) I am trying to get supervised/ attended tranfer working, blind transfer by pressing the # key works fine atxfer = * Attended transfers are only supported in CVS, not 1.0.X Julian J. M. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] What is web login password for Asteirsk@Home
Type help-aah and you will get list of commands to reset your passwords. Neel From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angus Comber Sent: Friday, March 25, 2005 7:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] What is web login password for [EMAIL PROTECTED] Hello I have setup [EMAIL PROTECTED] and can login to the system via the asterisk box. But if I try same username and password to login using the Asterisk Management Portal I try the same username and password and cannot login. says authorization failure. I have tried from a Windows 2000 and a Windows XP machine running Internet Explorer v6. What am I doing wrong? Angus Comber Itel Office Software Ltd 5 Enmore Gardens London, SW14 8RF Tel: 020 8878 7367 Fax: 020 8876 7257 Em: [EMAIL PROTECTED] web: www.iteloffice.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] debugging trunks between two asterisk boxes at two different locations
objective: users connected to box A can dial the extension number of users connected to box B boxA at location 1: works fine for internal lan users using the firefly softphone boxB at location 2: works fine for internal lan users using the firefly softphone Both the boxes have a IAX trunk defined following the instructions on: https://sourceforge.net/docman/display_doc.php?docid=26418group_id=121515 and guess what .. it doesnt work how do i go about debugging this thing ? t ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] What is web login password for Asteirsk@Home
Login: maint Password: password -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angus ComberSent: Friday, March 25, 2005 7:08 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] What is web login password for [EMAIL PROTECTED] Hello I have setup [EMAIL PROTECTED] and can login to the system via the asterisk box. But if I try same username and password to login using the Asterisk Management Portal I try the same username and password and cannot login. says authorization failure. I have tried from a Windows 2000 and a Windows XP machine running Internet Explorer v6. What am I doing wrong? Angus ComberItel Office Software Ltd5 Enmore GardensLondon, SW14 8RFTel: 020 8878 7367Fax: 020 8876 7257Em: [EMAIL PROTECTED]web: www.iteloffice.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Newbie Instalation
We have some good walkthrus at http://www.geekgazette.com. These should answer most of your questions all in one shot. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trevor Tregoweth Sent: Friday, March 25, 2005 1:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Newbie Instalation Hi All, Thanks for the wonderful advice, and comments, and anything I might of missed, and no offence taken. Yes I am new to this program, and Linux too, so this is a big learning curve. I installed the software Asterisk which I believed it did straight from the cd, rebooted the computers, and it installed more stuff. I am then lead to believe that I can use x-lite a phone interface, I guess, to interact with the new pabx-asterisk system I now have. I can see from the gui interface that I am trying to make calls, but that's about it, not much else is happening Well if you want to know much more, please ask, as I have no idea what I am doing :) You help and direction would be much appreciated. Cheers Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Square Key system
I have searched both the wiki and googled looking for a solution to a square key configuration. I need to have C.O. lines to appear on the buttons to facilitate a small office. All of the users can see each other and calls are put on hold and picked up by the other users instead of transferred. Has anyone done this? Can it be accomplished and how is it accomplished? Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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-write callerid?
Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk 208) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re-write callerid?
On Fri, 25 Mar 2005, Remco Barende wrote:
Is it possible to rewrite caller id's?
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I would like their
full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but surely there
must be a better solution?
Thanks!!
Remco
I set the Caller*ID before I place the outgoing call, like so:
exten = _91NXXNXX,1,SetCallerID(IgLou Internet 5029663848)
exten = _91NXXNXX,2,Dial(Zap/r1/${EXTEN:1})
Hope that helps,
--Sean
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
sean a. newton [EMAIL PROTECTED]
louisville, ky, usa http://wewt.net
Another day, another convertible and another hotel
full of cops.-- Hunter S. Thompson
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
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Re: [Asterisk-Users] Re: Optional URL in App. Queue
Vikram Rangnekar wrote: +++ Dan [20/03/05 09:17 +0200]: Hi James, - Original Message - From: James Coberly [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 19, 2005 11:41 PM Subject: Re: [Asterisk-Users] Re: Optional URL in App. Queue There appears to be a bug in 0.9.10f. was great in 0.9.10e and is working great in the 0.9.10g I have been testing for Dan. I'll push him a message and see if he can update it to the web site. James- A newer version (0.9.11a) is under testing now. I hope to be able to post it on my site later today. Best regards, Dan I am using 9.10g and yes the Url option with the Asterisk DIAL command works great but not the URL option with the asterisk QUEUE command any help with that ? You need to be running CVS later than Mar 1 to have the Queue URL function proper, there were some issues in chan_agent that were not passing the data. James- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with *72
I have the following config:
[app-callforward]
; dialed call forward app - forwards calling extension
exten = _*72.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:3})
exten = _*72.,2,Answer
exten = _*72.,3,Wait(1)
exten = _*72.,4,Playback(loligo/call-fwd-unconditional)
exten = _*72.,5,Playback(loligo/for)
exten = _*72.,6,Playback(loligo/extension)
exten = _*72.,7,SayDigits(${CALLERIDNUM})
exten = _*72.,8,Playback(loligo/is-set-to)
exten = _*72.,9,SayDigits(${EXTEN:3})
exten = _*72.,10,Macro(hangupcall)
However, when I do something like
*723372806
The system says it takes it but I am unable to make it work. Instead
of forwarding to an outside line, it sends the caller direct to
voicemail. The way I have things setup, I do not need to dial a 9 to
get an outside line. Any thoughts?
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Re: [Asterisk-Users] Re-write callerid?
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I would like their
full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but surely there
must be a better solution?
Thanks!!
Remco
I set the Caller*ID before I place the outgoing call, like so:
exten = _91NXXNXX,1,SetCallerID(IgLou Internet 5029663848)
exten = _91NXXNXX,2,Dial(Zap/r1/${EXTEN:1})
Yes, but this way you can only display one single phone number, and not
the MSN number for each SIP phone?
For example Henk has SIP/208 and MSN 0031201208
I would like to display Henk 208 for any call that stays in the company
but 0031201208 to the outside.
Thanks!
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[Asterisk-Users] ways to get more accuracy from ztdummy
Greetings! I have a FreeBSD 5.3 running on Intel SR1300 (dual xeon 2.6, scsi) server, with ztdummy.ko driver as timing source for asterisk. The typical output from zttest is: $ zttest Opened pseudo zap interface, measuring accuracy... [..skip..] --- Results after 192 passes --- Best: 99.987793 -- Worst: 98.266602 But when i listening MOH, it's quality is not very good. It is slightly choppy. (Asterisk _uses_ ztdummy, without ztdummy MOH sound is more awful) Is there any techniques for getting more accuracy? May be some software or hardware tricks? Thanks for any help. -- Aleksey Skripka .masterhost ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Outlook contacts-Asteriskdatabase(LookupCIDName)
http://www.identafone.com/cidpop.html Show right on the product page that it uses asttapi and integrates with Asterisk Henry . - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 24, 2005 8:44 PM Subject: RE: [Asterisk-Users] Outlook contacts-Asteriskdatabase(LookupCIDName) Which one? Didn't see it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Jueves, 24 de Marzo de 2005 04:39 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Outlook contacts-Asteriskdatabase(LookupCIDName) There is a separate component that you can purchase that will allow popups from outlook db. - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 24, 2005 4:36 PM Subject: RE: [Asterisk-Users] Outlook contacts -Asteriskdatabase(LookupCIDName) I tried using that. Works for outbound calls thru outlooks but didn't find a way to make it do the cidlookup on incoming calls, also, doesn't have any help that worked for this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Jueves, 24 de Marzo de 2005 02:56 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Outlook contacts - Asteriskdatabase(LookupCIDName) Search wiki for ASTTAPI - Original Message - From: Remco Barende [EMAIL PROTECTED] To: Asterisk Users List asterisk-users@lists.digium.com Sent: Thursday, March 24, 2005 1:28 PM Subject: [Asterisk-Users] Outlook contacts - Asterisk database(LookupCIDName) Is it possible in any way to use an Outlook contacts database as the source for the internal Asterisk database that is used for callerid lookups? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Re-write callerid?
On Fri, 25 Mar 2005, Remco Barende wrote:
exten = _91NXXNXX,1,SetCallerID(IgLou Internet 5029663848)
exten = _91NXXNXX,2,Dial(Zap/r1/${EXTEN:1})
Yes, but this way you can only display one single phone number, and not
the MSN number for each SIP phone?
For example Henk has SIP/208 and MSN 0031201208
I would like to display Henk 208 for any call that stays in the company
but 0031201208 to the outside.
I see.. I can't think of an easy way to do that, short of an AGI script
that checked a flat file or database. You'd pass the local extension to
the AGI and it'd return the MSN number.
--Sean
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
sean a. newton [EMAIL PROTECTED]
louisville, ky, usa http://wewt.net
Another day, another convertible and another hotel
full of cops.-- Hunter S. Thompson
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
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Re: [Asterisk-Users] Re-write callerid?
On Friday 25 March 2005 16:30, Remco Barende wrote:
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I would like
their full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but surely there
must be a better solution?
Thanks!!
Remco
I set the Caller*ID before I place the outgoing call, like so:
exten = _91NXXNXX,1,SetCallerID(IgLou Internet 5029663848)
exten = _91NXXNXX,2,Dial(Zap/r1/${EXTEN:1})
Yes, but this way you can only display one single phone number, and not
the MSN number for each SIP phone?
For example Henk has SIP/208 and MSN 0031201208
I would like to display Henk 208 for any call that stays in the company
but 0031201208 to the outside.
What's wrong with SetCallerID(${CallerIDName} 0031201${CallerIDNum}).
B
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[Asterisk-Users] Can I get a sip doorbell?
My home office is away from my house - so if anyone rings door I cannot hear it. How would I rig up a doorbell which would ring an extension on my Asterisk box? Angus Comber [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Polycom phones-buggy SIP firmware or am I missingsomething in the XML configs?
Jason Brown wrote: | Anyone have experiece with polycom phones? | | I am experiencing a really weird problem. In an office where I have | the following extensions: | On the Polycom phones, when I want to dial from extension 100 to any | extension 120 or above, or dial out, it dials just fine. If I want to | dial from extension 100 to extension 101,or 102 or 103 or 104, after | you dial 10 then it flashes connecting (really fast flash) but doesn't | connect to anything. Then you can dial the last digit of the extension. | Otherwise, if you dial 101 you are forced to dial the last 1 twice | because it wont send it. | | I have ruled out asterisk completely. Nothing wrong in the dialplan. I | have also ruled out DTMF. So it can either be buggy firmware or | something I am missing in the XML configs. Phone dialplan rules seem to be the culprit for something like that. Take a look at what the phone has set as the rules, and set as appropriate. Instructions are available, a link was posted earlier on the list... Kris | | Any ideas? I second the dialplan as the culprit...I know you said that nothing is wrong in the dialplanI could have sworn mine was good too, I was experiencing the exact same symptoms as you. Went over the dialplan with a fine tooth comb, corrected some logic mistakes that I had been overlooking every time I checked it, and finally the problem is gone. This cannot be the dialplan, since the dialplan does not see which digits you dial until after the entire number gets sent to asterisk (in other words, the digits don't get sent to asterisk as you press them). The digits get sent when you press the send button, or when the phone's digit map says the dial string is complete. I'm pretty certain this would be a result of your digit map matching some dial pattern. I think the default digit map will match '00', '11', and '10' and send them immediately to asterisk. You can manipulate the digit map on the web interface, or in the XML files (sip.cfg). As a stopgap measure, you can tell your users to just dial when the phone is on the hook and press the dial softbutton. This will bypass the digitmap check. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re-write callerid?
Hi,
-Original Message-
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I
would like their
full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but
surely there
must be a better solution?
Sure
SetCallerID(0031201234${CALLERIDNUM})
This assumes there is a direct link between the MSN and the internal CID
ofcourse. If that is not the case, you would need to create some form of
translation table.
Best regards,
Florian Overkamp
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RE: [Asterisk-Users] Re-write callerid?
Hi,
-Original Message-
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I
would like their
full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but
surely there
must be a better solution?
Sure
SetCallerID(0031201234${CALLERIDNUM})
This assumes there is a direct link between the MSN and the internal CID
ofcourse. If that is not the case, you would need to create some form of
translation table.
Best regards,
Florian Overkamp
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Re: [Asterisk-Users] Square Key system
Search Google. This is not a key system it is a pbx. I don't think you can accomplish what you want with this. - Original Message - From: Mark W Wood To: asterisk-users@lists.digium.com Sent: Friday, March 25, 2005 10:00 AM Subject: [Asterisk-Users] Square Key system I have searched both the wiki and googled looking for a solution to a square key configuration. I need to have C.O. lines to appear on the buttons to facilitate a small office. All of the users can see each other and calls are put on hold and picked up by the other users instead of transferred. Has anyone done this? Can it be accomplished and how is it accomplished? Thanks in advance. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re-write callerid?
This can be accomplished if the last 3 of the number you want to send to the
outside match the extension by using variables.
- Original Message -
From: Sean A. Newton [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, March 25, 2005 10:49 AM
Subject: Re: [Asterisk-Users] Re-write callerid?
On Fri, 25 Mar 2005, Remco Barende wrote:
exten = _91NXXNXX,1,SetCallerID(IgLou Internet 5029663848)
exten = _91NXXNXX,2,Dial(Zap/r1/${EXTEN:1})
Yes, but this way you can only display one single phone number, and not
the MSN number for each SIP phone?
For example Henk has SIP/208 and MSN 0031201208
I would like to display Henk 208 for any call that stays in the company
but 0031201208 to the outside.
I see.. I can't think of an easy way to do that, short of an AGI script
that checked a flat file or database. You'd pass the local extension to
the AGI and it'd return the MSN number.
--Sean
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
sean a. newton [EMAIL PROTECTED]
louisville, ky, usa http://wewt.net
Another day, another convertible and another hotel
full of cops.-- Hunter S. Thompson
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
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Re: [Asterisk-Users] Can I get a sip doorbell?
I would you an ATA and something like a Viking door box. Then if they ring the door bell it can call your phone and you could speak to the person to tell them you are on your way, leave the package or whatever. - Original Message - From: Angus Comber To: asterisk-users@lists.digium.com Sent: Friday, March 25, 2005 10:57 AM Subject: [Asterisk-Users] Can I get a sip doorbell? My home office is away from my house - so if anyone rings door I cannot hear it. How would I rig up a doorbell which would ring an extension on my Asterisk box? Angus Comber [EMAIL PROTECTED] ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound audio fades out with IAX Provider
I have an account with a IAX service provider that I'm happy with but recently I started getting rather strange reports from users. They're saying that occasionally, they'll be on a call via the provider and the outbound audio appears to slowly fade out to nothing with a bit of static during the fade. Does this ring any bells with someone? I'm going to update * on the server sometime soon and hopefully that'll fix this but I have no idea where the cause may be. Could this be a provider problem? A network issue? Any suggestions for further investigation would be appreciated. Paul Config: * CVS Nov-14, Dell SC420, FC3, 1xFXO Digium card (x100p?), SPA-841 desksets, SPA-3k for second FXO line and a cordles, behind a NAT'ing router with IAX port forwarded. IAX VoIP provider. -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] small qos switch
I have multiple locations running * where all the phone are on their own lan and all the data is on a separate lan. The problem is they are sharing the same dsl connection. The locations are IAX2 trunked together, but it only takes one data down/up load to just kill the voice. What I am looking for is a small switch with QoS that I can stick in ahead of the dsl modem. Plug in one connection from the voice lan and one from the data lan. I have found quite a few 24 or 48 port switches that will do this, but I really do not need anything that big. There are already switches in place. Any recommendations please? thanks, bk... -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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@Home Usage
hi all; my first post/question is a bit vague. i'll be more specific on [EMAIL PROTECTED] usage (eg: how was it?): 1. am able to make it running at home using a braodband connection of dynamic IP with no-ip, i able to SSH to the box and access to the web pages. the problem is with the maint password. how could i change it? in which mechanism that maint reside? is it on the mysql or the linux system user? 2. it look like [EMAIL PROTECTED] lacking documentation. but with the PHP and source is all there, it should not make a big of a problem (for a large number of asterisk user/developer). still, it will be nicer for a documentation. i have try to googled for a documentation still could not find any in depth documentation (such as the flash use). have you guys have any on the web. i may contribute on the documentation if necessary. well, maybe not necassary. but i'll try. 3. have anybody have been hacked by installing [EMAIL PROTECTED] looking at the main website, it says somebody does get attack buy not changing the root password. anything else? 4. also, i do not see any DUNDI config. maybe i should instal the main Asterisk on the system. i would like to test out this DUNDI thingie. 5. also, i would like to know, on the user of [EMAIL PROTECTED], on what instances that [EMAIL PROTECTED] usable for you? do you use it daily? does it hang on you? does it recompile your kernel without you knowing? i think that about it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio codec MP108
hi all, can any 1 pls tell me the context i shld add on sip.conf for Audiocodec MP108 8 fxs please. can`t get a dialtone only busy signal. Thnx ppls Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Detect called party pickup
I have been playing with getting the sample.call file to work by dropping it into /var/spool/asterisk/outgoing. The process works to the point of calling the desired number and plays the message. The problem is that the message starts playing almost immediately, so if the called person takes 2 or 3 rings to pick up the phone, half the message has already been played.Here's the relevant portion of my extensions.conf file:[outmsg1]exten = s,1,DigitTimeout,5 ; Set Digit Timeout to 5 secondsexten = s,2,ResponseTimeout,10 ; Set Response Timeout to 10 secondsexten = s,3,Wait(4)exten = s,4,Answerexten = s,5,Background(demo-congrats) ; "play outbound msg"exten = s,6,Background(demo-instruct) ; "Press 1 to replay or 2 to acknowledge receiving this message"exten = 1,1,Goto(s,5) ; replay messageexten = 2,1,Goto(msgack,s,1) ; acknowledge messageexten = t,1,Playback(vm-goodbye)exten = t,2,Hangup[msgack]exten = s,1,Playback(auth-thankyou)exten = s,2,Playback(vm-goodbye)exten = s,3,HangupThanks!PatPatrickHealyU.S.DistrictCourt,NYWD[EMAIL PROTECTED]304U.S.CourthouseVoice:716-332-177068CourtStreetFAX:716-551-4850Buffalo,NY14202___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re-write callerid?
Remco Barende wrote:
For example Henk has SIP/208 and MSN 0031201208
I would like to display Henk 208 for any call that stays in the
company but 0031201208 to the outside.
If your internal numbers always match your outside numbers just prefix it
SetCallerID(Company Name 0031201${CALLERIDNUM})
If outside numbers do not always match internal numbers you can use the
astdb to help
First populate the database
CLI database put outcid 208 0031201208
CLI database put outcid 209 0031201209
etc.
Second create a macro (not tested)
[macro-setclid]
exten = s,1,DBGet(clid=outcid/${CALLERIDNUM})
exten = s,2,SetCallerID(Company Name ${clid})
exten = s,102,SetCallerID(Company Name -- main company number here
-- )
Then before you dial it's as easy as
Macro(setclid)
This is similar to what I am currently using to tackel this exact problem.
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[Asterisk-Users] We just released our new Asterisk Installation CD set. with 24/7 monitoring
Here's our recent announcement of our new Asterisk Installation CD set: Signate has announced its new Asterisk Installation 2005 CD Set. It's, a complete software PBX (private branch exchange) telephony appliance in a single package. The CD set installs Linux pre-configured for telephony, a stable 1.0x distribution of the open source Asterisk PBX, and Signate's optional, free PBX monitoring. When Signate's Asterisk Installation 2005 CD set is loaded onto a PC with an internet or PSTN telephone connection, it creates a running VoIP PBX ready for configuration in about twenty minutes. SigMON, Signate's included PBX monitoring software, helps keep the PBX running. SigMON monitors about 20 different conditions on the PBX and sends alerts if a condition needs to be attended to. Monitored conditions range from hardware conditions such as available disk space and CPU utilization, software conditions such as whether the PBX is running, and telephony conditions such the state of connections to telecommunications providers. One instance of Signates PBX monitoring service is free for the PBX created by a Signate Asterisk Installation 2005 CD set. Signates VoIP Telephony with Asterisk Book and CD Set is $89.95 and Signate's Asterisk Installation 2005 CD is $49.95. They are available at Amazon, Signate or Ebay. Paul Mahler www.signate.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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/iax routing question
On Fri, 25 Mar 2005 04:07:13 -0500, Nabeel Jafferali [EMAIL PROTECTED] wrote: What happens if a SIP call is routed through more than one * server? If canreinvite=yes for all the peers involved, and t or T is not used in the Dial command, then the audio would get routed directly between the endpoints. Also, when setting up an inter asterisk exchange, is all the data routed through the * servers? As long as notransfer=no for all the peers involved, then everything but the endpoints would completely drop out of the call. Nabeel Thanks Nabeel, that's what I needed to know. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re-write callerid?
What about Callerid on call forwarding? I.e. an external call comes in and is forwarded to a cell phone, how do I make the callerid that is displayed on the cell phone the same as the inbound call? Thanks, Max Max Clark max [at] clarksys.com http://www.clarksys.com Remco Barende wrote: Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk 208) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] small qos switch
Linksys makes a VPN router with Dual WAN interfaces and QoS http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-31672629504.htm On Fri, 2005-03-25 at 11:13, Bob Knight wrote: I have multiple locations running * where all the phone are on their own lan and all the data is on a separate lan. The problem is they are sharing the same dsl connection. The locations are IAX2 trunked together, but it only takes one data down/up load to just kill the voice. What I am looking for is a small switch with QoS that I can stick in ahead of the dsl modem. Plug in one connection from the voice lan and one from the data lan. I have found quite a few 24 or 48 port switches that will do this, but I really do not need anything that big. There are already switches in place. Any recommendations please? thanks, bk... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] hardware question
[EMAIL PROTECTED] wrote on 03/25/2005 09:14:42 AM: Hello I want to to know if the motherboards VIA are fully supporte by asterisk. This is a complex question. The *software* is fully supported. Depending on the CPU you use, you may have to modify the makefiles (some VIA CPU's do not implement the CMOV instruction), but with that change the software will work just fine. However, Digium *hardware* is a different story. The TDM and X100P boards require that the card be placed on its own interrupt. Interrupts are scarce on a VIA platform: there's no IO-APIC, and there's a lot of integrated hardware. It is doable, however. I'm using a TDM board with a VIA EPIA-MII board with zero problems. No clicks, no static, nothing. I'm even sharing an interrupt (the TDM board and an unused (and AFAICT not-diableable) Cardbus controller), and still no problems. However, YMMV... And also, some of those motherboars say that with 1 pci slot , using a special riser card you can connect 2 pci cards. Will that work to have 2 pci cards (FXS or FXO ) on asterisk? Again, a complex question. The short answer is yes, the dual riser in and of itself will not cause a problem. The long answer is that it is highly unlikely that you'll find an interrupt for it. I have the dual riser and the second port wants to use an interrupt that already has a couple of devices on it, including the Ethernet interface. So, that's probably not possible on my system. The really annoying part is that my system has *SIX* unused interrupts: 3,4,6,10,11 and 13. Now I know that two of those are traditionally used by legacy devices (math coprocessor and floppy controller), but what about 3,4,10 and 11?!? I can find no way to get the computer to use those IRQ's. Everything's onboard, so changing PCI slots is not possible. It's frustrating. 15 interrupts is not exactly a lot, but when you ignore nearly half of them, it's real hard to use your motherboard... Tim Massey thank you Fabian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] hardware question
[EMAIL PROTECTED] wrote on 03/25/2005 09:14:42 AM: Hello I want to to know if the motherboards VIA are fully supporte by asterisk. This is a complex question. The *software* runs on Mini-ITX (what I assume you're asking about) just fine. The *hardware* *may* have issues however. These devices do not support IO-APIC, so you can have interrupt issues with the X100P and TDM400 devices. I am running a TDM400 on a Via EPIA-MII board, so far, without problems. No static, no clicks, no buzzing, no erros, nothing. So far... And also, some of those motherboars say that with 1 pci slot , using a special riser card you can connect 2 pci cards. Will that work to have 2 pci cards (FXS or FXO ) on asterisk? Again, a complex question. The short answer is yes. The PCI riser cards will work just fine in and of themselves. However, the odds of you being able to get two interrupts completely free and clear for the use of two TDM boards is slim. This whole IRQ routing issue is a drag. On my system, there are three interrupts completely unused (3, 4, 6, 10, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re-write callerid?
On Fri, 25 Mar 2005, Bob Goddard wrote:
On Friday 25 March 2005 16:30, Remco Barende wrote:
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I would like
their full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but surely there
must be a better solution?
Thanks!!
Remco
I set the Caller*ID before I place the outgoing call, like so:
exten = _91NXXNXX,1,SetCallerID(IgLou Internet 5029663848)
exten = _91NXXNXX,2,Dial(Zap/r1/${EXTEN:1})
Yes, but this way you can only display one single phone number, and not
the MSN number for each SIP phone?
For example Henk has SIP/208 and MSN 0031201208
I would like to display Henk 208 for any call that stays in the company
but 0031201208 to the outside.
What's wrong with SetCallerID(${CallerIDName} 0031201${CallerIDNum}).
Nothing other than that if you have a number block of 100 MSN's it would
benice to actually use them instead of just showing the first number for
every call.
There must be companies that have a similar problem?
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Re: [Asterisk-Users] small qos switch
On 09:13, Fri 25 Mar 05, Bob Knight wrote: I have multiple locations running * where all the phone are on their own lan and all the data is on a separate lan. The problem is they are sharing the same dsl connection. The locations are IAX2 trunked together, but it only takes one data down/up load to just kill the voice. What I am looking for is a small switch with QoS that I can stick in ahead of the dsl modem. Plug in one connection from the voice lan and one from the data lan. I have found quite a few 24 or 48 port switches that will do this, but I really do not need anything that big. There are already switches in place. Any recommendations please? You can install a bridging firewall loaded with your fav OS for firewalling that supports qos (Linux, *BSD, Solaris) -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Can I get a sip doorbell?
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20door gee that took a lot of effort. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angus Comber Sent: Friday, March 25, 2005 11:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can I get a sip doorbell? My home office is away from my house - so if anyone rings door I cannot hear it. How would I rig up a doorbell which would ring an extension on my Asterisk box? Angus Comber [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re-write callerid?
On Fri, 25 Mar 2005, Trevor Peirce wrote:
Remco Barende wrote:
For example Henk has SIP/208 and MSN 0031201208
I would like to display Henk 208 for any call that stays in the company
but 0031201208 to the outside.
If your internal numbers always match your outside numbers just prefix it
SetCallerID(Company Name 0031201${CALLERIDNUM})
If outside numbers do not always match internal numbers you can use the astdb
to help
First populate the database
CLI database put outcid 208 0031201208
CLI database put outcid 209 0031201209
etc.
Second create a macro (not tested)
[macro-setclid]
exten = s,1,DBGet(clid=outcid/${CALLERIDNUM})
exten = s,2,SetCallerID(Company Name ${clid})
exten = s,102,SetCallerID(Company Name -- main company number here --
)
Then before you dial it's as easy as
Macro(setclid)
This is similar to what I am currently using to tackel this exact problem.
Great this and Florian Overkamp's suggestion will solve the problem
indeed.
And in extensions.conf I have separate extensions for national and
international calls so I can just create a set for each.
Thanks both!
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Re: [Asterisk-Users] Zap Detect called party pickup
On Fri, 25 Mar 2005 [EMAIL PROTECTED] wrote: I have been playing with getting the sample.call file to work by dropping it into /var/spool/asterisk/outgoing. The process works to the point of calling the desired number and plays the message. The problem is that the message starts playing almost immediately, so if the called person takes 2 or 3 rings to pick up the phone, half the message has already been played. You need answer supervision on your line. It is available on isdn lines and some analogue lines. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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@Home Usage
Jermal, Your second round of questions are just as basic, do some research on wiki. 1/ did you not see that when you log onto the console it says type help-aah to change passwords? 2/ [EMAIL PROTECTED] doesn't need any more documentation - all of the documentation for [EMAIL PROTECTED] is on the voip-info wiki or the asterisk docs. 3/ - pass 4/ of course you can add dundi by modifying the dial plan. 5/ - pass but it works fine for me since Andrew relased [EMAIL PROTECTED] a few months ago. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bagan jermal Sent: Friday, March 25, 2005 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [EMAIL PROTECTED] Usage hi all; my first post/question is a bit vague. i'll be more specific on [EMAIL PROTECTED] usage (eg: how was it?): 1. am able to make it running at home using a braodband connection of dynamic IP with no-ip, i able to SSH to the box and access to the web pages. the problem is with the maint password. how could i change it? in which mechanism that maint reside? is it on the mysql or the linux system user? 2. it look like [EMAIL PROTECTED] lacking documentation. but with the PHP and source is all there, it should not make a big of a problem (for a large number of asterisk user/developer). still, it will be nicer for a documentation. i have try to googled for a documentation still could not find any in depth documentation (such as the flash use). have you guys have any on the web. i may contribute on the documentation if necessary. well, maybe not necassary. but i'll try. 3. have anybody have been hacked by installing [EMAIL PROTECTED] looking at the main website, it says somebody does get attack buy not changing the root password. anything else? 4. also, i do not see any DUNDI config. maybe i should instal the main Asterisk on the system. i would like to test out this DUNDI thingie. 5. also, i would like to know, on the user of [EMAIL PROTECTED], on what instances that [EMAIL PROTECTED] usable for you? do you use it daily? does it hang on you? does it recompile your kernel without you knowing? i think that about it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Help With Adit 600 Configuration
Andrew Kohlsmith wrote: Also, carrier access has an incredible support site that they do not charge for. You do need to register with them but that's free. Just for a follow up to this statement, I recently purchased an Adit 600 via eBay, tried to gain access to Carrier Access's website. They no longer allow for account creation without calling them. When I called, and gave them the serial number, they told me that access to their website will cost me $50. After I paid the $50, I found that all their firmware was $185 to download, each package. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Any 24 (or 30) way FXS PCI cards?
Just found a 12 port single card with opensource drivers 12 user configurable FX0/FXS analogue ports for $1,680 at asterisk mall ( http://www.asteriskmall.com ). I am not sure how well this card works with asterisk. Has anyone used these cards? Voip supply has a few 24 port gateways that are FXS based. The biggest one for FXO is 10 ports. They are not cheap the both cost about $2000 USD. a Channel bank with a T1 card will cost you about the same at least with a FXS ports. FXO costs more usually because that is typically the Office station side that has much lager power requirements. Where FXS is the phone/customer side of the Communications. . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] We just released our new Asterisk Installation CD set. with 24/7 monitoring
SNIP SigMON, Signate's included PBX monitoring software, helps keep the PBX running. SigMON monitors about 20 different conditions on the PBX and sends alerts if a condition needs to be attended to. Monitored conditions range from hardware conditions such as available disk space and CPU utilization, software conditions such as whether the PBX is running, and telephony conditions such the state of connections to telecommunications providers. One instance of Signates PBX monitoring service is free for the PBX created by a Signate Asterisk Installation 2005 CD set. SNIP Am I reading correctly that the only way to use the monitoring feature is to allow you to monitor the PBX for me?? You do not say anything about this in your email but is listed on your web site. There is no way I am sending unknown data to a third party about MY PBX... I am very capapble of fixing it myself. Was about ready to order the CD set until I read this!!! Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Major problems with TDM400 and specific telephones: suggestions?
Sorry for my bad english. I'm a french guy. Absolument rien à critiquer de ton anglais I have the same problems with siemens dect phones S100 The caller id don't work on tdm... snip Try adding cadence=250,1500,1500,3000,1500,3000 In zapata.conf And use in extension.conf exten = 200,1,Dial(Zap/1r1,20,tr) Excellent idea, thanks for posting it! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
CAUTION: Re: [Asterisk-Users] grandstream firmware update 1.0.5.23
CAUTION: voicemail screwed up for me (garbled) with upgrade to 23, went back to .22 and all is well. Don't know why, I'll look at it later. dean collins wrote: Version 1.0.5.23 is now available from http://gs-firmware.gratissip.dk/ Or directly from Grandstream at http://www.grandstream.com/BETATEST/Release-b21p1.0.5.23.zip Release notes doc here http://www.grandstream.com/BETATEST/Release_Note_1.0.5.23.doc while on the matter I just want to extend a note of thanks to Grandstream, I had 2 early handsets of theirs fail recently (about 9 months old) when I was unable to return them to the dealer I bought them from they organized for me to rma them directly. 2 brand new grandstreams now sitting on my desk. Has anyone else noticed that they have changed not only the plastic composition of the handsets but also the design of the handpiece itself (slightly thinner and slightly heavier) I could be wrong but to me it sounds like the voice quality has improved between the older model and the newer one so slightly. And for $50 or there abouts you cant complain. Just my 0.02c worth. Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] small qos switch
Hi,The lan is probably not the problem, but the dsl connection is.There are some things you can do that can help to a certain degree.First, set tos=lowdelay in your iax.conf.Most routers obey the ToS field.second, try to find out if your firewall (I assume you use one) support QoS.then there is a possibility you can set a QoS or ToS in the rulebase.Maybe you can even set a guaranteed and maximum bandwidth for this rule(at least this is the way I work, but then again I use Juniper/Netscreen firewalls)The third choice is to use a switch that can set port based QoS or ToS,but make sure it is not reset further down the line. But having said that, if your provider does'nt do anything with QoS and/or ToS it might al be useless.Try and find out.RegardsAndre- Oorspronkelijk Bericht -Onderwerp:[Asterisk-Users] small qos switchAfzender: Bob Knight [EMAIL PROTECTED]Aan:asterisk-users@lists.digium.comDatum:25-03-2005 18:22I have multiple locations running * where all the phone areon their own lan and all the data is on a separate lan.The problem is they are sharing the same dsl connection.The locations are IAX2 trunked together, but it only takesone data down/up load to just kill the voice.What I am looking for is a small switch with QoS that Ican stick in ahead of the dsl modem. Plug in one connectionfrom the voice lan and one from the data lan.I have found quite a few 24 or 48 port switches that will do
this, but I really do not need anything that big. There arealready switches in place.Any recommendations please?thanks, bk...-- Bob Knight[-w] the work option[EMAIL PROTECTED]925-449-9163___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Detect called party pickup
Date: Fri, 25 Mar 2005 18:39:26 +0100 (CET) From: Peter Svensson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zap Detect called party pickup On Fri, 25 Mar 2005 [EMAIL PROTECTED] wrote: I have been playing with getting the sample.call file to work by dropping it into /var/spool/asterisk/outgoing. The process works to the point of calling the desired number and plays the message. The problem is that the message starts playing almost immediately, so if the called person takes 2 or 3 rings to pick up the phone, half the message has already been played. You need answer supervision on your line. It is available on isdn lines and some analogue lines. We have a module that will do this detection. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
