[asterisk-users] Re: Digium makes the list!
In article [EMAIL PROTECTED], Doug Lytle [EMAIL PROTECTED] wrote: There is a link on Groklaw for the following article: Open source companies to watch Digium makes the second entry on the list. Link below: http://www.networkworld.com/news/2006/082806-open-source.html?ts Interesting. Digium's the only one in the list that I've heard of! BTW, don't read anything into being second on the list - the companies are listed in alphabetical order. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and g729 licenses
jurgen ha scritto: Hi, The problem happens when I record a call using MixMonitor. Even though it's recording natively in g729, a single call uses 2 decoders and one encoder! The only explanation I can think of for that is that MixMonitor is transcoding the g729 streams to something else, muxing them, then encoding the muxed stream out to g729. This seems ridiculous - why go through all that work and licenses? Does anyone know for sure what's going on here? I could go back to using Monitor, I suppose, but MixMonitor is somewhat less hacky. I think this is a simple tech problem, to mix 2 compressed source you MUST decode them, mix, and encode after... i think it is impossibile to mix compressed sound! Bye. signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does anyone offer truly unlimited voip in the US
Hi,Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COMRegards,Chandra."Eric \"ManxPower\" Wieling" [EMAIL PROTECTED] wrote: Steven M. Sawczyn wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define "unlimited" as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month?Most providers have unlimited minutes on the plans that are not flat rate. i.e. you can use as many mins as you want at 2/cents/min.If you mean "unlimited for a flat monthly fee" there is nobody out there stupid enough to offer that service, or, if they are, they don't stay in business.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does anyone offer truly unlimited voip in the US
Hi, Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COM Regards, Chandra."Steven M. Sawczyn" [EMAIL PROTECTED] wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define "unlimited" as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Thanks for any suggestions, Steve ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER Dispatcher Load Balance How-To?
Andy Chung (Power-All) wrote: Hi all, I have three Asterisk servers behind a SER. I want to know how to configure the Dispatcher module of SER to achieve load balance for the Asterisk servers. I have visited http://www.openser.org/docs/modules/1.1.x/dispatcher.html, is there any web sites have more detail information on that? Why not just use DNS round robin in various route[x] blocks in SER? This way you can load balance anything based on the URI that was requested. We use this approach and it works very very nicely - In fact I haven't even so much as thought about SER in a very long time, it just works. I prefer SER over OpenSER, but that's me. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GXP-2000 update to betafirmware?
Hi, currently I use version 1.1.0.16 for my GXP-2000 which works really fantastic. The only drawback I see is the addressbook. Is the firmware 1.1.1.9 stable enough to use the phone in normal environment? The webpage http://www.voip-info.org/wiki/view/GXP-2000 says that there it is possible to download the addressbook as a XML-file. The problem is if the version not works it is not possible to downgrade to 1.1.0 Thx for any feedback, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Development and Release Cycle
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Of course we care. Turns out that schedule was unrealistic, and when we start the next cycle we will regroup and decide if we either stretch out the cycle or reduce the amount of new functionality that gets added during the cycle. OK, thank you for info. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.4 I hear other party's voice only when I speack need help - SOLVED -
Giorgio Incantalupo wrote: Hi Tommaso, have you tried to search for noise suppression? I remember some phone has a function to automatically suppress it so the caller does not hear anything and thinks the other party has hung up. Giorgio Incantalupo Tommaso Calosi wrote: I have this problem with Asterisk 1.2.4 I hear other party's voice only when I speack or i make some noise. Otherwise i hear nothing. Futhermore every time i receive a call , this message is displayed : -- Started music on hold, class 'my_class', on SIP/ some random public ip address -08222740 any help? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I've solved right now. The problem was occurring just with musiconhold. Icoming calls are answerd and a message is played back using dial with m option. The problem is that the caller has silence unless it produces noise. I had to modprobe ztdummy. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialplan help
Dear friends,
Does anyone know how do i convert hex to int in the dialplan. I want to do
this:-
Take the sip call-id in hex, use CUT to extract the first part , and convert it
to an int. But the math function ony takes arguments as int. Can anyone suggest
how to do that?
eg:-
exten = _X.,n,Set(sipcid = ${CUT(SIPCALLID,-,1)}) --- evaluates to E305CEC5
I want this hex value in int. But i cant think of a clean solution.
Please help.
Thanks in advance.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest Rutherford
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Development and Release Cycle
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I don't know. Do you use Asterisk? That makes you part of the team. Have you tested the trunk version? Provided assistance testing out patches waiting for completion? Really, once all the new features have been completed, it will be released. If you would prefer it to be released now (I.E. before everything has been tested and possibly fixed), just download SVN trunk. Hi Matt, Yes, I have downloaded SVN trunk. I'm using H264 codec from it. There is one question I need to ask. How can I find out what are new options in SVN trunk? Right now I know only for H264, where can I find the list of others? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk Development and Release Cycle
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I don't know. Do you use Asterisk? That makes you part of the team. Have you tested the trunk version? Provided assistance testing out patches waiting for completion? Really, once all the new features have been completed, it will be released. If you would prefer it to be released now (I.E. before everything has been tested and possibly fixed), just download SVN trunk. Hi Matt, Yes, I have downloaded SVN trunk. I'm using H264 codec from it. There is one question I need to ask. How can I find out what are new options in SVN trunk? Right now I know only for H264, where can I find the list of others? Have a look for UPGRADE.txt. Sorry I can't be more specific. :) Other cool things: make menuconfig Jingle/jabber support IAX2 media transfers new sound files New answer machine detection (AMD) and much much more! :) - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE9Va9S6d5vy0jeVcRAvX4AJ4+z04hRvvVDhYuE4EAy+4cLfh/sQCeKrw8 AwYgI0jqU5skChEJA4QJck0= =TqfH -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Line detection with TDM400P
Hello I just want to have a confirmation, line status detection (with digium TDM400P) is highly not reliable outside of US. With busydetect=yes and callprogress=yes I can experience very strange phenomenons (randomilally occurs) like pick up not detected or hang up not detected. I'm in Israel , somebody knows how to improve this detection (it's very important for me to get call status live). Thank you _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - Comfort Noise - Patch/Release
Hi, Does anybody know if asterisk 1.4 will support comfort noise? Or if there is a patch for it now? If it will be in 1.4 any idea of release date? Thanks, Dean Bath. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom 360 Function Keys
I have a Snom 360 phone and I'm configuring it for use with Asterisk 1.2.9 and Freepbx 2.1.1 On my PBX there are: 1) Some SIP phones 2) One digium quadri primary ISDN interface (TE410P) 3) Two Rhino Channel Banks 4) 25 Analogue Phones on every channel bank How I can configure function keys on my SNOM 360 for monitoring analogue phone status? Configure sip phones is very simple (just put in function keys panel the SIP URI of every phone) but I have same problems with analogue phones! Someone have the same problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail, how to localize date in email notifications?
Hi all Two questions. We have a multi language voicemail setup. Unfortunately I did not find a way to localize the email notification sent to the customer. How can one do this? For the moment messages are hard-coded in german. The System Locale is 'C'. emaildateformat=%A, %d %B %Y um %H:%M:%S produces English Day and Month Names within our email sent in german. Can this be changed without altering the System Locale? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ pgpfvVHtTQjlN.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk Development and Release Cycle
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I don't know. Do you use Asterisk? That makes you part of the team. Have you tested the trunk version? Provided assistance testing out patches waiting for completion? Really, once all the new features have been completed, it will be released. If you would prefer it to be released now (I.E. before everything has been tested and possibly fixed), just download SVN trunk. Hi Matt, Yes, I have downloaded SVN trunk. I'm using H264 codec from it. There is one question I need to ask. How can I find out what are new options in SVN trunk? Right now I know only for H264, where can I find the list of others? Err, wasn't the patch for H.264 just changing one digit for another? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compile problems with app_rxfax.c and asterisk 1.2.11
Hello! On 8/29/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Trying to add faxing to asterisk but get a compile error. Any ideas? Is it broken for Asterisk 1.2.11 or was it me again :-) app_rxfax.c:105: error: structure has no member named `column_resolution' app_rxfax.c:105: error: structure has no member named `row_resolution' app_rxfax.c:116: error: structure has no member named `row_resolution' app_rxfax.c:122: error: structure has no member named `row_resolution' This is happen when you compile with spandsp-0.0.3. Remove and use spandsp-0.0.2 instead. bye, a ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] caller display problem
Hi all, I have 2 agents (1234, 4321) and a PSTN phone (9876). When 1234 makes a call to 4321, 4321 will have a callerid 1234 on his screen. Now, 4321 has forwarded his call to PSTN phone. When 1234 makes a call to 4321, it will forward to PSTN phone. However, caller display can't show correctly in the PSTN phone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX questions
Hello! Thanks for your answers! Everything works fine now there was some problem at my provider. I compiled and use rxfax successfully. bye, Zsolt On 8/15/06, Marco Mouta [EMAIL PROTECTED] wrote: Hi, I didn't try that way, only tx fax in call file. But my experience is when u r working with FAX you MUST disable echocanceller! On 8/15/06, Andy Kuo [EMAIL PROTECTED] wrote: Hi Marco, I'm using T406P(with hardware EC) with a T1-PRI, and I'm having trouble sending fax out though SIP ATA in the same LAN subnet with the Asterisk box. I can send fax out using txfax in call file, but I did have to lower the rxgain and txgain. This is what I'm trying to do: Fax machine --- SIP ATA --LAN-- Asterisk --PRI-- PSTN Have you tried this? Do you have to disable Echo canneler? Thanks. Andy On 8/15/06, Marco Mouta [EMAIL PROTECTED] wrote: Hi, Another question. With latest version of asterisk softwares am I able using rxfax? I had read some remarks about incompatibility between TDM card and rxfax. Is it still exist? I've been using rx for fax reception with TE110P as well as X100P (this only for tests and learning) with very success. As far as i know what could be a problem is that SpanDSP doesn't implements ECM (error correction mode) For Fax reception, only with X100P i've had to play with rxgains, nothing else. I've had some problems only for tx fax lots of errors transmiting faxs, but i think that could be because my * is behind a legacy pbx and i could be facing time sinchronization problems. bye, Zsolt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7960G SIP firmware 8.4
Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Any info? SIP Flash Image for 7940/7960 IP Phone v8.4(0) - Non CallManager ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] personal address progress pri
the configuration is this : NT PRI TD405P TE A -- B (Asterisk) A make a call to B. A can display the ID (caller ID , example John) of B ? these information are exchanged in the call progress ? B can change the called number and communicate this change to A whene the call is hangup ? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SER Dispatcher Load Balance How-To?
SER/OpenSER can get around call forwarding/transfer problems. You just need to account for those SIP dialogue's that can be problematic, and bypass using the dispatcher module for those situations. One thing to remember is to replicate usr-loc info that is cached in memory, otherwise load-balancing for INVITE's will fail if passed to a proxy that did not receive the REGISTER request for sending party. The standard t_replicate() function only supports replication between two SER/OpenSER proxies, so if you are looking to add in any more, you will need a more manual approach. OpenSER 1.1.0 actually offers the ability to operate completely out of a database, even for usr-loc, which could work for you. Other approaches for load balancing include DNS round-robin techniques, and separate software or hardware appliances dedicated to load distribution. I have successfully worked with Foundry ServerIron's, and Vovida's Open Source load-balancing proxy, in the past. Cheers, Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: 30 August 2006 04:05 To: Asterisk Users Mailing List - Non-Commercial Discussion; asterisk-users@lists.digium.com Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To? That might not be a good idea. If you transfer or forward calls on your phones, you MUST make sure the transferred or forwarded call goes back to the same Asterisk box it was handled on. If you use the dispatcher, and load balance, there is no guarantee of that, and transfers and forwarding will break. Doug. -Original Message- From: Andy Chung (Power-All) [mailto:[EMAIL PROTECTED] Sent: Tue 8/29/2006 7:49 PM To: asterisk-users@lists.digium.com Cc: Subject: [asterisk-users] SER Dispatcher Load Balance How-To? Hi all, I have three Asterisk servers behind a SER. I want to know how to configure the Dispatcher module of SER to achieve load balance for the Asterisk servers. I have visited http://www.openser.org/docs/modules/1.1.x/dispatcher.html, is there any web sites have more detail information on that? Thanks! Andy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analyze core file prodeced after safe_asterisk crashh
Thanks a lot!Muchas gracias amigo!EstebanOn 8/29/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:-BEGIN PGP SIGNED MESSAGE-Hash: SHA1Steve Edwards wrote: It's not clear if the OP wanted 1) information on how to analyse the core file or 2) provide information to the bug tracker for others to analyse. Matt's answer addresses #2. How about #1? Anybody care to share their techniques for analysing a core dump?Doing the bt full as described in the document I posted is how youanalyse the core file.- --Cheers,Matt Riddell ___http://www.sineapps.com/news.php (Daily Asterisk News - html)http://freevoip.gedameurope.com (Free Asterisk Voip Community)http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.orgiD8DBQFE9J+OS6d5vy0jeVcRAlSYAJ4rt5j9UPkiMqsjumHAdgWCrZhcOgCfWy1QtlXc8iRplvZp3IE/TvWroZ8==p+IX-END PGP SIGNATURE- ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Prompts playback changing tempo in SMP kernel
Hi all, this is probably a weird question and something I'm not doing right but I got this bizarre thing going on here. When I boot the system with the SMP kernel and compile (*) with the smp kernel source (actually even if I don't compile, but as long as I boot into the SMP kernel), I get this problem where calling into the system, say to check my voicemail, the prompt playback continously changes tempo. The prompts are played in slow-motion, and then it speeds up to its normal speed, then goes back in slow-mo and so on. It happens (I think) at constant periods. Only the tempo changes, not the pitch of the prompt. Does anyone have any idea what could be happening? I have watched topconstantly but haven't noticed anything bizarre in terms of CPU or Mem usage. This is on a 100mbps LAN with nothing much else on it. And it only happens when it's booted into the smp kernel. So it's something to do with smp, thread scheduling, or some buffer BUT I don't know what exactly. All you champs out there, esp. the asterisk-dev people, any light you can shed on this? Thanks much \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] does anyone offer truly unlimited voip in the US
Packet8 is unlimited usa, or a more expensive plan for unlimited global. You have the use an ata however. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, 30 August 2006 12:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] does anyone offer truly unlimited voip in the US Steven M. Sawczyn wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define unlimited as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Most providers have unlimited minutes on the plans that are not flat rate. i.e. you can use as many mins as you want at 2/cents/min. If you mean unlimited for a flat monthly fee there is nobody out there stupid enough to offer that service, or, if they are, they don't stay in business. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Ring on Multiple Phones
Sorry I was not clear William. In the actual code, the exten marked 'old' is commented out and only 'new' is active. Then I reload. But only the single 120 instrument rings. Larry William Piper wrote: The whole thing. Both (old and new) have the same exten and the same priority, you can't do that and expect it to work properly. The new exten will call all 3 phones at the same time, whoever answers first gets the call. If you want it to call SIP/120 first and if they don't answer then ring to all 3, you'd want to do this: exten =_879677[67],1,Dial(SIP/120|20) ;this will ring for 20 seconds then go to priority 2. exten =_879677[67],2,Dial(SIP/120SIP/122SIP/124) bp On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote: Color me puzzled. What part of: exten = _879677[67],1,Dial(SIP/120) should be deleted? Larry William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote: This is a reply to a fairly old thread. My EXTEN string is meant to ring 3 phones (will increase to 12) thus: old: exten =_879677[67],1,Dial(SIP/120); works fine new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124) I edit extensions.conf to the new line above, type 'reload' into the CLI, see the new line with 'show dialplan' and actually see the new line above, but when I dial the DID 879-6777 it rings on extension 120 only. Have I missed a step? Larry Jonathan k. Creasy wrote: EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow Sent: Tuesday, November 08, 2005 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extension Ring on Multiple Phones Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Ring on Multiple Phones
Sorry I was not clear Rushowr. In the actual extensions.conf as used, the 'old' line is commented out so only 'new' is active. Then I reload. However, only the single 120 line rings instead of all. Larry Rushowr wrote: Then entire OLD extension must be removed so the new one will match -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Larry Alkoff Sent: Tuesday, August 29, 2006 6:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones Color me puzzled. What part of: exten = _879677[67],1,Dial(SIP/120) should be deleted? Larry William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote: This is a reply to a fairly old thread. My EXTEN string is meant to ring 3 phones (will increase to 12) thus: old: exten =_879677[67],1,Dial(SIP/120); works fine new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124) I edit extensions.conf to the new line above, type 'reload' into the CLI, see the new line with 'show dialplan' and actually see the new line above, but when I dial the DID 879-6777 it rings on extension 120 only. Have I missed a step? Larry Jonathan k. Creasy wrote: EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow Sent: Tuesday, November 08, 2005 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extension Ring on Multiple Phones Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does anyone offer truly unlimited voip in the US
Teliax is not unlimited but has a cap of 2500 minutes per month.*** Softcap of 2500 Minutes (including 1000 minutes of toll-free inbound, if applicable).On 8/30/06, Crazy Boy [EMAIL PROTECTED] wrote:Hi, Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COM Regards, Chandra.Steven M. Sawczyn [EMAIL PROTECTED] wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define unlimited as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Thanks for any suggestions, Steve ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PrivacyManager
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all, I mentioned this back in February, and there wasn't much response (John Novack was the only one who responded.) I assumed it was due to the fact that nobody was really sure. :) So, I dropped the idea and haven't re-visited it until today -- and I still have the same issue. I set up PrivacyManager for calls coming into my DID. Calls coming into my DID will ring my SIP phone first (Polycom 501) and then my cordless phone (Panasonic 2.4GHz set attached to a Digium TDM card.) When I call from my cell phone, I am prompted for my number by PrivacyManager. I enter the number, and the call proceeds. However, when it rings on my SIP phone, it displays as Unknown. When it finally rings over to my analog phone, it will display Privacy Manager along with my cell number. Obviously, it *should* display that same Privacy Manager and number on the SIP phone. It used to work. (Although, I don't think it has worked since 1.2.) We're using 1.2.11 now (just upgraded -- but again, this problem seems to have gone through all the 1.2 versions.) I did a SIP debug on my peer (the Polycom 501) and the packets being sent to the phone *do* say Unknown -- and NOT the PrivacyManager info. Anyone have any ideas on this? Is PrivacyManager busticated (and has been all this time), or am I overlooking some obvious config option or flag to Dial? (I've played with a few flags -- no luck.) If anyone can shed some light on this, I'd appreciate it. Jeremy - -- - -- Jeremy G. Gault, KD4NED | Network/Telecom Admin +1 (423) 303-2562 voice | WinWorld Corporation +1 (423) 472-9265 fax| http://www.winworld.com/ ICQ: 54084581| AIM: WinWorldJG - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE9Y3XXRhQkRPmcc8RApB+AKDM4HuLWda+kHTgW7hlCmHHPatuSwCfRaDb gw6AbmpKqw4zGfDVu1ASKc8= =5fRp -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] does anyone offer truly unlimited voip in the US
What cost do you pay per month for the 2500 minutes? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Tom VileSent: 30 August 2006 13:54To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] does anyone offer truly unlimited voip in the USTeliax is not unlimited but has a cap of 2500 minutes per month."*** Softcap of 2500 Minutes (including 1000 minutes of toll-free inbound, if applicable)." On 8/30/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COMRegards,Chandra. "Steven M. Sawczyn" [EMAIL PROTECTED] wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define "unlimited" as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Thanks for any suggestions, Steve ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Everyone is raving about the all-new Yahoo! Mail. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX call drops, recent instability
Hi all I've had a number of servers, all generally running Asterisk 1.0.9-1.0.11.1, with TDM cards for analog lines. They have been in production use for many months, handling incoming calls, and also allowing daily inter-server calls over IAX (transfers, extension calls etc) All of a sudden, in the last 3-4 weeks, with little to no changes to any config or setup on the servers -- a large number of IAX-IAX calls are dropping. It is driving me nuts because I can't pinpoint any change in the system that might be a catalyst ... nor rectify with any modifications to iax.conf, zapata.conf etc All servers are iax.conf 'friend' entries.. standardized with disallow=all, allow=gsm, and allow=ulaw jitterbuffer=off trunk=yes I have explored a number of theories and none seem to be really helping the situation. The call drops are not consistent, so it is hard to say. One thing I have considered is shared IRQs on some of the servers --- BUT while I know this can affect TDM installations -- these machines have been in production with no drops for months! So does anyone have any suggestions as to why all of a sudden IAX calls and my asterisk network would become so unstable? Any suggestions appreciated -- Chris Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does anyone offer truly unlimited voip in the US
$24 per monthOn 8/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: What cost do you pay per month for the 2500 minutes? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Tom VileSent: 30 August 2006 13:54To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] does anyone offer truly unlimited voip in the USTeliax is not unlimited but has a cap of 2500 minutes per month.*** Softcap of 2500 Minutes (including 1000 minutes of toll-free inbound, if applicable). On 8/30/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COMRegards,Chandra. Steven M. Sawczyn [EMAIL PROTECTED] wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define unlimited as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Thanks for any suggestions, Steve ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Everyone is raving about the all-new Yahoo! Mail. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] does anyone offer truly unlimited voip in the US
How many simultaneous calls? Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: Wednesday, 30 August 2006 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] does anyone offer truly unlimited voip in the US $24 per month On 8/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: What cost do you pay per month for the 2500 minutes? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Tom Vile Sent: 30 August 2006 13:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] does anyone offer truly unlimited voip in the US Teliax is not unlimited but has a cap of 2500 minutes per month. *** Softcap of 2500 Minutes (including 1000 minutes of toll-free inbound, if applicable). On 8/30/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi, Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COM Regards, Chandra. Steven M. Sawczyn [EMAIL PROTECTED] wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define unlimited as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Thanks for any suggestions, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] does anyone offer truly unlimited voip in the US
Doesnt matter I just checked, only 2. Also the soft-cap for residential is 1500 mins for $24.99 2500 soft-cap is for corporate with $44 a month (but has 4 lines) Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: Wednesday, 30 August 2006 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] does anyone offer truly unlimited voip in the US $24 per month On 8/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: What cost do you pay per month for the 2500 minutes? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Tom Vile Sent: 30 August 2006 13:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] does anyone offer truly unlimited voip in the US Teliax is not unlimited but has a cap of 2500 minutes per month. *** Softcap of 2500 Minutes (including 1000 minutes of toll-free inbound, if applicable). On 8/30/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi, Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COM Regards, Chandra. Steven M. Sawczyn [EMAIL PROTECTED] wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define unlimited as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Thanks for any suggestions, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: IAX call drops, recent instability
In article [EMAIL PROTECTED], Chris Earle [EMAIL PROTECTED] wrote: Hi all I've had a number of servers, all generally running Asterisk 1.0.9-1.0.11.1, with TDM cards for analog lines. They have been in production use for many months, handling incoming calls, and also allowing daily inter-server calls over IAX (transfers, extension calls etc) All of a sudden, in the last 3-4 weeks, with little to no changes to any config or setup on the servers -- a large number of IAX-IAX calls are dropping. It is driving me nuts because I can't pinpoint any change in the system that might be a catalyst ... nor rectify with any modifications to iax.conf, zapata.conf etc All servers are iax.conf 'friend' entries.. standardized with disallow=all, allow=gsm, and allow=ulaw jitterbuffer=off trunk=yes I have explored a number of theories and none seem to be really helping the situation. The call drops are not consistent, so it is hard to say. One thing I have considered is shared IRQs on some of the servers --- BUT while I know this can affect TDM installations -- these machines have been in production with no drops for months! So does anyone have any suggestions as to why all of a sudden IAX calls and my asterisk network would become so unstable? Is it a private network, or used by other machines and traffic? Could it be that the network has started getting overloaded with broadcast or multicast traffic or something? Just a thought. Do the asterisk logs give any clues about why they are dropping the calls? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma Problems - A104d not detected
Hi! I'm trying to install a A104d. 1. LSPCI detects the card: # lspci ... 00:1f.2 IDE interface: Intel Corporation 82801FB/FW (ICH6/ICH6W) SATA Controller (rev 03) 05:04.0 Class affe: Sirrix AG security technologies Sirrix.PCI4S0 4-port ISDN S0 interface (rev 02) 05:09.0 Network controller: Sangoma Technologies Corp. A104d QUAD T1/E1 AFT card 40:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit Ethernet PCI Express (rev 01) 2. I build the wanpipe drivers - all without problems 3. The wanrouter utility can not find the card (thus wancfg also does not detect the card) # wanrouter hwprobe --- | Wanpipe Hardware Probe Info | --- Card Cnt: S508=0 S514X=0 S518=0 A101-2=0 A104=0 A300=0 A200=0 A108=0 Any hints how to solve the problem? I am using: # cat /etc/redhat-release Red Hat Enterprise Linux AS release 4 (Nahant Update 4) # uname -a Linux salxvoip01 2.6.9-42.0.2.ELsmp #1 SMP Thu Aug 17 18:00:32 EDT 2006 i686 i686 i386 GNU/Linux I tried with the newest wanpipe drivers wanpipe-beta7-2.3.4.tgz thanks klaus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Problems - A104d not detected - solved
Hi! Problem solved. I just removed the wanrouter modules and tried again. This thime there were some more modules loaded and the card is found: ]# wanrouter hwprobe --- | Wanpipe Hardware Probe Info | --- 1 . AFT-A104-SH : SLOT=9 : BUS=5 : IRQ=193 : CPU=A : PORT=1 : HWEC=128 : V=20 2 . AFT-A104-SH : SLOT=9 : BUS=5 : IRQ=193 : CPU=A : PORT=2 : HWEC=128 : V=20 3 . AFT-A104-SH : SLOT=9 : BUS=5 : IRQ=193 : CPU=A : PORT=3 : HWEC=128 : V=20 4 . AFT-A104-SH : SLOT=9 : BUS=5 : IRQ=193 : CPU=A : PORT=4 : HWEC=128 : V=20 Card Cnt: S508=0 S514X=0 S518=0 A101-2=0 A104=1 A300=0 A200=0 A108=0 regards klaus Klaus Darilion wrote: Hi! I'm trying to install a A104d. 1. LSPCI detects the card: # lspci ... 00:1f.2 IDE interface: Intel Corporation 82801FB/FW (ICH6/ICH6W) SATA Controller (rev 03) 05:04.0 Class affe: Sirrix AG security technologies Sirrix.PCI4S0 4-port ISDN S0 interface (rev 02) 05:09.0 Network controller: Sangoma Technologies Corp. A104d QUAD T1/E1 AFT card 40:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit Ethernet PCI Express (rev 01) 2. I build the wanpipe drivers - all without problems 3. The wanrouter utility can not find the card (thus wancfg also does not detect the card) # wanrouter hwprobe --- | Wanpipe Hardware Probe Info | --- Card Cnt: S508=0 S514X=0 S518=0 A101-2=0 A104=0 A300=0 A200=0 A108=0 Any hints how to solve the problem? I am using: # cat /etc/redhat-release Red Hat Enterprise Linux AS release 4 (Nahant Update 4) # uname -a Linux salxvoip01 2.6.9-42.0.2.ELsmp #1 SMP Thu Aug 17 18:00:32 EDT 2006 i686 i686 i386 GNU/Linux I tried with the newest wanpipe drivers wanpipe-beta7-2.3.4.tgz thanks klaus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Problems - A104d not detected
Sangoma provides EXCELLENT support. I would try them I just installed a A101, and had some problems, but the hwprobe found the card OK. You MIGHT want to try different PCI slots before contacting them My problem was somewhat different, and was fixed by a reboot of the machine between installation steps Also, I am using the latest stable drivers John Novack Klaus Darilion wrote: Hi! I'm trying to install a A104d. 1. LSPCI detects the card: # lspci ... 00:1f.2 IDE interface: Intel Corporation 82801FB/FW (ICH6/ICH6W) SATA Controller (rev 03) 05:04.0 Class affe: Sirrix AG security technologies Sirrix.PCI4S0 4-port ISDN S0 interface (rev 02) 05:09.0 Network controller: Sangoma Technologies Corp. A104d QUAD T1/E1 AFT card 40:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit Ethernet PCI Express (rev 01) 2. I build the wanpipe drivers - all without problems 3. The wanrouter utility can not find the card (thus wancfg also does not detect the card) # wanrouter hwprobe --- | Wanpipe Hardware Probe Info | --- Card Cnt: S508=0 S514X=0 S518=0 A101-2=0 A104=0 A300=0 A200=0 A108=0 Any hints how to solve the problem? I am using: # cat /etc/redhat-release Red Hat Enterprise Linux AS release 4 (Nahant Update 4) # uname -a Linux salxvoip01 2.6.9-42.0.2.ELsmp #1 SMP Thu Aug 17 18:00:32 EDT 2006 i686 i686 i386 GNU/Linux I tried with the newest wanpipe drivers wanpipe-beta7-2.3.4.tgz thanks klaus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New to Asterisk...
Hi UsersI'm new to Asterisk PBX.Mainly i'm using the openser for call routing and Asterisk as PBX and Voicemail generating.let see my secnario ---UAC -- ser Asterisk(for voice mail only and extension and PBX Purposes SER system ip is 192.168.2.75:5060Asterisk is in 192.168.2.76:5060 When i start the asterisk server by typing asterisk -c [chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': FoundAug 30 19:41:05 WARNING[6694]: acl.c:244 ast_get_ip_or_srv: Unable to lookup ' 192.168.2.75:5060'Aug 30 19:41:11 WARNING[6694]: acl.c:244 ast_get_ip_or_srv: Unable to lookup ' 192.168.2.75:5060' == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0please help in this. -- Thanks and Regards with cheersSunkara Ravi Prakash (Voip Developer) Hyperion TechnologyKondapur, Hi-tech city,Hyderabad.www.hyperion-tech.com+91-9985077535 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SER Dispatcher Load Balance How-To?
Anything is possible. The biggest challenge with OpenSER is getting past the horrible documentation and the cryptic, one line responses to questions asked in the mailing list. -Original Message- From: Adam Linford [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 30, 2006 5:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To? SER/OpenSER can get around call forwarding/transfer problems. You just need to account for those SIP dialogue's that can be problematic, and bypass using the dispatcher module for those situations. One thing to remember is to replicate usr-loc info that is cached in memory, otherwise load-balancing for INVITE's will fail if passed to a proxy that did not receive the REGISTER request for sending party. The standard t_replicate() function only supports replication between two SER/OpenSER proxies, so if you are looking to add in any more, you will need a more manual approach. OpenSER 1.1.0 actually offers the ability to operate completely out of a database, even for usr-loc, which could work for you. Other approaches for load balancing include DNS round-robin techniques, and separate software or hardware appliances dedicated to load distribution. I have successfully worked with Foundry ServerIron's, and Vovida's Open Source load-balancing proxy, in the past. Cheers, Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: 30 August 2006 04:05 To: Asterisk Users Mailing List - Non-Commercial Discussion; asterisk-users@lists.digium.com Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To? That might not be a good idea. If you transfer or forward calls on your phones, you MUST make sure the transferred or forwarded call goes back to the same Asterisk box it was handled on. If you use the dispatcher, and load balance, there is no guarantee of that, and transfers and forwarding will break. Doug. -Original Message- From: Andy Chung (Power-All) [mailto:[EMAIL PROTECTED] Sent: Tue 8/29/2006 7:49 PM To: asterisk-users@lists.digium.com Cc: Subject: [asterisk-users] SER Dispatcher Load Balance How-To? Hi all, I have three Asterisk servers behind a SER. I want to know how to configure the Dispatcher module of SER to achieve load balance for the Asterisk servers. I have visited http://www.openser.org/docs/modules/1.1.x/dispatcher.html, is there any web sites have more detail information on that? Thanks! Andy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SER Dispatcher Load Balance How-To?
-Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 30, 2006 1:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SER Dispatcher Load Balance How-To? Andy Chung (Power-All) wrote: Hi all, I have three Asterisk servers behind a SER. I want to know how to configure the Dispatcher module of SER to achieve load balance for the Asterisk servers. I have visited http://www.openser.org/docs/modules/1.1.x/dispatcher.html, is there any web sites have more detail information on that? Why not just use DNS round robin in various route[x] blocks in SER? This way you can load balance anything based on the URI that was requested. What about transfers and forwards? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SER Dispatcher Load Balance How-To?
-Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 29, 2006 11:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To? Well, it really depends on what he's using the asterisk servers for. If it's for voicemail or apps, he'll just have to make sure that certain apps land on certain servers, and voicemail can be distributed for various things. If ser can do what I've heard/read it can do, it can handle all the basic call functions (i.e. forwarding) for plenty of calls. Also, if the asterisk servers are just acting as gateways (i.e. t1, e1, etc), then they will have no problem handling a load balanced configuration. To do that, you'd need to use the avpops module in OpenSER. You think Asterisk documentation is bad? Wait until you try and get that stuff to work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SER Dispatcher Load Balance How-To?
On Wed, 2006-08-30 at 08:34 -0600, Douglas Garstang wrote: -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 29, 2006 11:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To? Well, it really depends on what he's using the asterisk servers for. If it's for voicemail or apps, he'll just have to make sure that certain apps land on certain servers, and voicemail can be distributed for various things. If ser can do what I've heard/read it can do, it can handle all the basic call functions (i.e. forwarding) for plenty of calls. Also, if the asterisk servers are just acting as gateways (i.e. t1, e1, etc), then they will have no problem handling a load balanced configuration. To do that, you'd need to use the avpops module in OpenSER. You think Asterisk documentation is bad? Wait until you try and get that stuff to work. LOL, I've never gotten further than installing SER, so yeah, I understand ;) -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SER Dispatcher Load Balance How-To?
If you want a good explaination of SER and how to use it start here. http://siprouter.onsip.org/doc/gettingstarted/ They have GREAT pre-written configs and walk you through ever part of SER. I was about scrap SER before I found these tutorials. Natambu Obleton Network Engineer FastTrack Communications [EMAIL PROTECTED] (970) 247-3366 office (970) 247-2426 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, August 30, 2006 8:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To? -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 29, 2006 11:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To? Well, it really depends on what he's using the asterisk servers for. If it's for voicemail or apps, he'll just have to make sure that certain apps land on certain servers, and voicemail can be distributed for various things. If ser can do what I've heard/read it can do, it can handle all the basic call functions (i.e. forwarding) for plenty of calls. Also, if the asterisk servers are just acting as gateways (i.e. t1, e1, etc), then they will have no problem handling a load balanced configuration. To do that, you'd need to use the avpops module in OpenSER. You think Asterisk documentation is bad? Wait until you try and get that stuff to work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan help
On 14:23, Wed 30 Aug 06, [EMAIL PROTECTED] wrote:
Dear friends,
Does anyone know how do i convert hex to int in the dialplan. I want to do
this:-
Take the sip call-id in hex, use CUT to extract the first part , and convert
it to an int. But the math function ony takes arguments as int. Can anyone
suggest how to do that?
eg:-
exten = _X.,n,Set(sipcid = ${CUT(SIPCALLID,-,1)}) --- evaluates to E305CEC5
I want this hex value in int. But i cant think of a clean solution.
Please help.
Use a simple agi script that does this for you.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help please == Wrong password
Hi i have a small problems with my asterisk connected to phonesystems : Now i have this message: -- SIP read from 62.39.136.151:5060: SIP/2.0 403 Cant accept register from myself Via: SIP/2.0/UDP 84.14.xx.xx:5060;branch=z9hG4bK38f74bd7;rport=5060 From: sip:[EMAIL PROTECTED];tag=as42b95c05 To: sip:[EMAIL PROTECTED];tag=e3fe971527b049ab0c1e91db33fcbf5f.cf8c Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER Server: PSN Sip Proxy (1.1.3 (PRX3-EXTERNAL)) Content-Length: 0 Warning: 392 62.39.136.151:5060 Noisy feedback tells: pid=11434 req_src_ip=62.39.136.151 req_src_port=5060 in_uri=sip:sip3.phonesystems.net out_uri=sip:sip3.phonesystems.net via_cnt==2 --- (9 headers 0 lines)--- Aug 30 17:12:50 WARNING[15568]: chan_sip.c:10010 handle_response: Forbidden - wrong password on authentication for REGISTER but my login/password are correct into sip.conf the configuration have changed in asterisk 1.2.11 ? thanks for your help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using HINT with Cisco 7940/SIP
On Sat, 2006-06-17 at 15:49 -0500, Lacy Moore - Aspendora wrote: Can't be done using the 7960 with SIP, unless you are talking about just monitoring that phone. You can monitor a 7960, but you can't show the status of other phones on a 7960 with SIP. Do you know wether it can be done with a 7940(SIP)? Can it display status of (for example) 4205,hint,SIP/phone1 ? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] does anyone offer truly unlimited voip in the US
You can get as many minutes and channels as you require from TelIAX. You just have to call them to customize the account. Start by setting up the Corporate Account, then call them to customize it to your needs. Erv Bauman NISCOMM +1-412-567-0343 ext. 150 11 Aldred Lane Pittsburgh, PA 15227 USA Please visit our website at www.niscomm.net Where there is an open mind, there will always be a frontier Charles F. Kettering From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 30, 2006 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] does anyone offer truly unlimited voip in the US Doesnt matter I just checked, only 2. Also the soft-cap for residential is 1500 mins for $24.99 2500 soft-cap is for corporate with $44 a month (but has 4 lines) Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: Wednesday, 30 August 2006 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] does anyone offer truly unlimited voip in the US $24 per month On 8/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: What cost do you pay per month for the 2500 minutes? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tom Vile Sent: 30 August 2006 13:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] does anyone offer truly unlimited voip in the US Teliax is not unlimited but has a cap of 2500 minutes per month. *** Softcap of 2500 Minutes (including 1000 minutes of toll-free inbound, if applicable). On 8/30/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi, Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COM Regards, Chandra. Steven M. Sawczyn [EMAIL PROTECTED] wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define unlimited as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Thanks for any suggestions, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk presence (from manager API)
Hello, I would like to somehow get the presence of IAX2 and SIP users from Asterisk Manager API or using any other means. I tried watching for PeerStatus event, but it seems unrealiable (http://bugs.digium.com/view.php?id=7833). I tried defining hint for user and sending ExtensionState event, which is also unreliable (once I had qualify OK status in iax2 show peers, I could receive calls and I got status of 4, which is unavailable). How to get reliable information about peer status? I have qualify=yes in all iax friends, I am using realtime and I can receive calls or dial without any problems. Thanks, Juraj. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sipura 3000 and Asterisk
Hi pals, im trying to save some money in
cellphones calls, so i bought a GSM gateway and a Sipura SPA3000
gateway.
The GSM gw is currently working, and now im trying to configure the
SPA, but every call i send, i get a 503 service unavailable.
Im using this extension to match cell calls and sendthem to the spa:
exten = _15.,1,Dial(SIP/300/${EXTEN}) ; cellphones are 15
and 300 is the spa3000 extension, registered OK
exten = _15.,2,Hangup
In the dialplan section of the sipura, i ve tried many different
options like xx.:@gw0, (xx.) and many others.
I cannot find a formal configuration doc for this device, so if you
giveme a hand to configure it or tellme where to start, or where
is the problem i would be very pleased.
Thanks in advance
--
Francisco Seratti
Sunesys Telecomunicaciones
Bouchard 644. 5to A. Puerto Madero
[EMAIL PROTECTED]
Tel: (54) 011- 4311-9009 (Rotativas)
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] does anyone offer truly unlimited voip in the US
At 05:02 AM 8/30/2006, you wrote: Packet8 is unlimited usa, or a more expensive plan for unlimited global. You have the use an ata however. I think you'll find they're only unlimited until you abuse them! Most seem to have a 2000-3000 minutes/month limit written somewhere in the fine print of their contract. Most home users will never reach that so it appears unlimited. An unlimited business plan likely still has a limit of 5000 or 6000 minutes/month which is about all 1 person making outbound calls can reasonably do. Personally, one I figured out Asterisk and SIP it became more cost efficient to pay 1.4 cents/minute for outgoing, 2 cents/minute for incoming and $1 to $5/month for each incoming number, at least it does having 5 numbers for 2 people. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Ring on Multiple Phones
I don't know then, I do the same exact thing: exten = _352688,3,Dial,SIP/202SIP/214|20 Perhaps try sending everything in that context exactly as it is typed let us look at it. I'mpretty sure you have something configured incorrectly. Thanks, bp On 8/30/06, Larry Alkoff [EMAIL PROTECTED] wrote: Sorry I was not clear Rushowr.In the actual extensions.conf as used, the 'old' line is commented out so only 'new' is active.Then I reload.However, only the single 120line rings instead of all.LarryRushowr wrote: Then entire OLD extension must be removed so the new one will match -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Larry Alkoff Sent: Tuesday, August 29, 2006 6:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones Color me puzzled.What part of: exten = _879677[67],1,Dial(SIP/120) should be deleted? Larry William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agent solution w/o id/password
Hello, I'm looking for an agent managing dialplan/software/agi/whatever that independent from asterisk queue management. I already tried this http://www.voip-info.org/wiki/view/Agents+without+agent+channel with no success but a lot of warning. I'm using asterisk 1.2.10 and the dialplan above made for 1.0 might that cause the trouble. So I'm looking for an agent management that not need agents.conf like id and password for login. Instead if someone dial an extension from his phone that agent (extension actually) login. If dial an another extension he logout. If a logged in agent don't answer for a duration automatically logoff. If no free agent on incoming call just play a sound and hangup. This time I don't need queues just 'plain' agents whos dynamically login/logout. For example: I dial 8301 and I log in with my phone (Zap, SIP, whatever). If I dial 8302 then I log off. If I don't answer for an incoming within 15 secs asterisk automatically log me out. If asterisk's queue managent can do this by default that would be much better but as I see that only know the id/password solution. bye, Zsolt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk presence (from manager API)
Google is your friend: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState bp On 8/30/06, Juraj Bednar [EMAIL PROTECTED] wrote: Hello,I would like to somehow get the presence of IAX2 and SIP users fromAsterisk Manager API or using any other means. I tried watching for PeerStatus event, but it seems unrealiable(http://bugs.digium.com/view.php?id=7833).I tried defining hint for user and sending ExtensionState event, which is also unreliable (once I had qualify OK status in iax2 showpeers, I could receive calls and I got status of 4, which isunavailable).How to get reliable information about peer status? I have qualify=yes in all iax friends, I am using realtime and I can receive calls ordial without any problems.Thanks, Juraj.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] does anyone offer truly unlimited voip in the US
The one we use here works out at $0.0286 cents per min, but has unlimited amount of lines,we use one account for our call centre and we have had up to 40 calls in the call queue, and it works fine. Not sure if they do USA numbers but could find out if needed. We also use one account for all outbound calls, 20 people here use the one account fine. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Dean CollinsSent: 30 August 2006 14:31To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] does anyone offer truly unlimited voip in the US Doesnt matter I just checked, only 2. Also the soft-cap for residential is 1500 mins for $24.99 2500 soft-cap is for corporate with $44 a month (but has 4 lines) Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom VileSent: Wednesday, 30 August 2006 9:16 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] does anyone offer truly unlimited voip in the US $24 per month On 8/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: What cost do you pay per month for the 2500 minutes? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Tom VileSent: 30 August 2006 13:54To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] does anyone offer truly unlimited voip in the US Teliax is not unlimited but has a cap of 2500 minutes per month."*** Softcap of 2500 Minutes (including 1000 minutes of toll-free inbound, if applicable)." On 8/30/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COMRegards,Chandra. "Steven M. Sawczyn" [EMAIL PROTECTED] wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define "unlimited" as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Thanks for any suggestions, Steve ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Everyone is raving about the all-new Yahoo! Mail. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Any thoughts on the new Xserve?
I found this, which looked interesting: http://wiki.onmac.net/index.php/Triple_Boot_via_BootCamp Also, Apple released a new version of BootCamp that supports the Xserve on Aug 16. If it'd work, and you could shoehorn a PRI card into it, man wouldn't that make a nice Asterisk box? And at $2999, quite competitive with a DL380G4. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] upgrade problem on IP phone 9133i
hi everybody,I bought few units for evaluation but we were not able to upgrade the firmware to 1.4 , it's currently set at 1.2, when we go to the webadmin page, whether we try to change the IP of the tftp server or the firmware name and set values, the reply is always Invalid IP address Please try again if we change the value via the phone itself , the tftp ip is changed but the firmware does not come up we are sure of our tftp server since it's used to upgrade other phones from other brands any idea ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960G SIP firmware 8.4
Like what? I haven't tried the non-Call Manager version yet. The Call Manager version seems to work fine with Asterisk. Haven't run into any issues yet. I wish there was a softkey for DND, but that hasn't seemed to be in any SIP version. I thought maybe the CallManager version would have this. On 8/30/06, Hermann Wecke [EMAIL PROTECTED] wrote: Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Anyinfo?SIP Flash Image for 7940/7960 IP Phone v8.4(0) - Non CallManager___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does anyone offer truly unlimited voip in the US
Steve, VoiceEclipse has a US unlimited plan for $20/month. Two inbound numbers that can be in different area codes. I have not figured out how to recognize which number the inbound call came in on, but, right now, that is not that important to me. Others have had other problems. Research is recommended. Bob... Steven M. Sawczyn wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define unlimited as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Thanks for any suggestions, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using HINT with Cisco 7940/SIP
On Wed, 2006-08-30 at 16:25 +0100, Conrad Wood wrote: On Sat, 2006-06-17 at 15:49 -0500, Lacy Moore - Aspendora wrote: Can't be done using the 7960 with SIP, unless you are talking about just monitoring that phone. You can monitor a 7960, but you can't show the status of other phones on a 7960 with SIP. Do you know wether it can be done with a 7940(SIP)? Can it display status of (for example) 4205,hint,SIP/phone1 ? Conrad No, not running SIP. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Ring on Multiple Phones
Make sure all of the lines you are ringing are registered up and running. I noticed this when I did a paging extension. I rang about 40 phones and the second it saw one offline it failed only ringing one phone. William Piper wrote: I don't know then, I do the same exact thing: exten = _352688,3,Dial,SIP/202SIP/214|20 Perhaps try sending everything in that context exactly as it is typed let us look at it. I'm pretty sure you have something configured incorrectly. Thanks, bp On 8/30/06, *Larry Alkoff* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sorry I was not clear Rushowr. In the actual extensions.conf as used, the 'old' line is commented out so only 'new' is active. Then I reload. However, only the single 120 line rings instead of all. Larry Rushowr wrote: Then entire OLD extension must be removed so the new one will match -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Larry Alkoff Sent: Tuesday, August 29, 2006 6:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones Color me puzzled. What part of: exten = _879677[67],1,Dial(SIP/120) should be deleted? Larry William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk = Master and Slave ?
Hi a small question: I have one Asterisk Server with: VoIP Provider gateway for incomming/outgoing call 5 VoIP Phone (i name it Master) i want add a another Asterisk server but only connected to: 5 new VoIP Phone To the master for incoming/outgoing call (in g729) It's possible ? anyone have a sample of config ? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan help
Hi Michael,
Thanks a lot. I am working on an agi script and it does it. Thanks a lot again.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest Rutherford
Michiel van Baak wrote:
On 14:23, Wed 30 Aug 06, [EMAIL PROTECTED] wrote:
Dear friends,
Does anyone know how do i convert hex to int in the dialplan. I want to do
this:-
Take the sip call-id in hex, use CUT to extract the first part , and convert
it to an int. But the math function ony takes arguments as int. Can anyone
suggest how to do that?
eg:-
exten = _X.,n,Set(sipcid = ${CUT(SIPCALLID,-,1)}) --- evaluates to
E305CEC5
I want this hex value in int. But i cant think of a clean solution.
Please help.
Use a simple agi script that does this for you.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960G SIP firmware 8.4
Seems to be working ok on my handset for the past couple of weeks. No major bugs, registration, xml services and MWI works etc..etc.. Have not given it a thorough testing though. Regards, Nathan. On 30/08/2006, at 6:51 PM, Hermann Wecke wrote: Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Any info? SIP Flash Image for 7940/7960 IP Phone v8.4(0) - Non CallManager ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco 7960G SIP firmware 8.4
For DND press Call Forward All (CFwdAll softkey) then Messages button on the SCCP version. I havent seen the SIP version of 7961G. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Wednesday, August 30, 2006 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960G SIP firmware 8.4 Like what? I haven't tried the non-Call Manager version yet. The Call Manager version seems to work fine with Asterisk. Haven't run into any issues yet. I wish there was a softkey for DND, but that hasn't seemed to be in any SIP version. I thought maybe the CallManager version would have this. On 8/30/06, Hermann Wecke [EMAIL PROTECTED] wrote: Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Any info? SIP Flash Image for 7940/7960 IP Phone v8.4(0) - Non CallManager ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Aspendora, Inc. Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER Dispatcher Load Balance How-To?
Douglas Garstang wrote: What about transfers and forwards? if your system is designed properly, it doesn't matter which Asterisk box actually processes the call. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk speaks Russian!
Westany, the Asterisk voice experts, announce their first Russian voice for the Asterisk PBX. Tamara, a Russian female voice, is the latest addition to Westany¹s growing catalogue of proven, meticulously-crafted voice prompt¹ suites for Asterisk, Freepbx, trixbox, Bicomsystems and Amp. Produced by our qualified, experienced sound staff to our own exacting standards of excellence, Tamara is voiced by a native-speaking, professionally trained voice artist. There¹s simply no substitute for knowledge and experience. Tamara includes all the standard voice prompts you¹ll need to run your Asterisk PBX. Including Voice Menus, Call Queues, Call transfers, Call Parking, Voice Mail, Error messages, Numbers (digits), letters and Phonetics. Westany voice prompt suites provide everything you need in one box. Call us on +44 (0) 800 066 4864, or go to http://www.westany.com and find out more. First-class voices at no-frills prices from Westany, the Asterisk voice experts. Regards Stuart -- Westany- Voices that bring asterisk to life [EMAIL PROTECTED] Sales +44 (0)800 066 4864 Direct +44 (0)207 043 8814 Mobile +44 (0)79 7045 9548 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Ring on Multiple Phones
William I found and fixed the problem. Your comment gave me the kick to persevere. Thank you very much. My exten line had a comment at the end that contained a close paren. That apparently screwed up the context line - although it shouldn't have. Now all three extensions ring. Note my mail program wrapped the line but it's not wrapped in the file: [telasip-in] ;=== exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124) ; to be all extensions) exten =_512879677[67],1,Dial(SIP/120SIP/122SIP/124) This leads to another problem. I have 13 sip phones for [telasip-in] and other contexts to call ring groups for extension intercomming. Is there some kind of macro I could have to replace the instances of: (SIP/120SIP/122SIP/124) I have not yet written or read up on macros. Larry William Piper wrote: I don't know then, I do the same exact thing: exten = _352688,3,Dial,SIP/202SIP/214|20 Perhaps try sending everything in that context exactly as it is typed let us look at it. I'm pretty sure you have something configured incorrectly. Thanks, bp On 8/30/06, Larry Alkoff [EMAIL PROTECTED] wrote: Sorry I was not clear Rushowr. In the actual extensions.conf as used, the 'old' line is commented out so only 'new' is active. Then I reload. However, only the single 120 line rings instead of all. Larry Rushowr wrote: Then entire OLD extension must be removed so the new one will match -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Larry Alkoff Sent: Tuesday, August 29, 2006 6:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones Color me puzzled. What part of: exten = _879677[67],1,Dial(SIP/120) should be deleted? Larry William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Intertex IX68 GW2 AIR 802.11G ADSL2+ ?
Does anyone have any experience with this device? Does it interface nicely as a FXS / FXO for use with Asterisk? smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Speex Problemz
Hi all, I've compiled/installed both * and Speex but I'm getting an error upon * startup: ...Aug 30 11:23:34 WARNING[27652]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/codec_speex.so: undefined symbol: speex_preprocess_ctl Aug 30 11:23:34 WARNING[27652]: loader.c:554 load_modules: Loading module codec_speex.so failed! Just wondering if anybody else has come across this error. Thanks! Tj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk with PABX
How about creating some documentation? -Original Message- From: Ira [mailto:[EMAIL PROTECTED] Sent: Monday, August 28, 2006 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk with PABX At 11:16 AM 8/28/2006, you wrote: Actually Eric I disagree with you. Through the use of config edit it allows you to look into each of the conf folders to understand the layout of a multi channel in/out asterisk server. IMHO: I started with AAH pre TrixBox and soon thereafter moved to a clean * install and learned to create my own dial plans. Not to say that AAH is bad, it's just so advanced that I personally don't think I learned much if anything from using it. If you intend to use and stay with that solution then by all means use it, but in the end for me it didn't do anything other than prove * would work for me and teach me that it was more trouble to learn AAH than it was going to be to just start using *. And again, I'm not trying to knock TrixBox, but it's a box and if you fit in it, it's probably the best thing there is. Personally, when I started, it seemed too small and limiting so I learned to program my own dial plans. It's not trivial as the language we've been given is not really well thought out and poorly documented at best, but in the end it's done everything I've asked of it. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agent solution w/o id/password
Here's what we do:
[agent-login]
exten = s,1,NoOp(${AgentUser})
exten =
s,2,AddQueueMember(${AgentContext}|${AgentChannel}|${AgentPenalty})
exten = s,3,Wait(1)
exten = s,4,Playback(agent-loginok)
exten = s,5,Hangup
exten = s,103,RemoveQueueMember(${AgentContext}|${AgentChannel})
exten = s,104,Wait(1)
exten = s,105,Playback(agent-loggedoff)
exten = s,106,Hangup
[tax-line]
exten = s,1,Macro(dnv-messagebox-setup)
exten = s,n,Set(AgentContext=${CONTEXT})
exten = s,n,Set(AgentChannel=${CHANNEL})
exten = s,n,Set(AgentChannel=${CUT(AgentChannel,-,-2)})
exten = s,n,Set(AgentUser=${CUT(AgentChannel,/,2)})
exten = s,n,NoOp(${AgentUser})
; tax-queue agents
exten = s,n,GotoIf($[${AgentUser} = 2488-tessmanl]?:macdonap)
exten = s,n,Set(AgentPenalty=1)
exten = s,n,Goto(agent-login,s,1)
exten = s,n(macdonap),GotoIf($[${AgentUser} =
2488-macdonap]?:chengb)
exten = s,n,Goto(agent-login,s,1)
exten = s,n(chengb),GotoIf($[${AgentUser} = 2488-chengb]?:listhael)
exten = s,n,Set(AgentPenalty=2)
exten = s,n,Goto(agent-login,s,1)
exten = s,n(listhael),GotoIf($[${AgentUser} =
2488-listhael]?:nguyent)
exten = s,n,Set(AgentPenalty=3)
exten = s,n,Goto(agent-login,s,1)
exten = s,n(nguyent),GotoIf($[${AgentUser} =
2488-nguyent]?:NonAgentStart)
exten = s,n,Set(AgentPenalty=4)
exten = s,n,Goto(agent-login,s,1)
exten = s,n(NonAgentStart),BackGround(call-processors/2488)
Hope this helps.
CP
On Aug 30, 2006, at 8:55 AM, Artifex Maximus wrote:
Hello,
I'm looking for an agent managing dialplan/software/agi/whatever that
independent from asterisk queue management. I already tried this
http://www.voip-info.org/wiki/view/Agents+without+agent+channel
with no success but a lot of warning. I'm using asterisk 1.2.10 and
the dialplan above made for 1.0 might that cause the trouble.
So I'm looking for an agent management that not need agents.conf like
id and password for login. Instead if someone dial an extension from
his phone that agent (extension actually) login. If dial an another
extension he logout. If a logged in agent don't answer for a duration
automatically logoff. If no free agent on incoming call just play a
sound and hangup. This time I don't need queues just 'plain' agents
whos dynamically login/logout.
For example:
I dial 8301 and I log in with my phone (Zap, SIP, whatever). If I dial
8302 then I log off. If I don't answer for an incoming within 15 secs
asterisk automatically log me out.
If asterisk's queue managent can do this by default that would be much
better but as I see that only know the id/password solution.
bye,
Zsolt
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quadBRI beronet card: how to specify which ISDN channel to use to make calls
Giorgio, I believe the syntax for mISDN is mISDN/port:channel/number. In other words, replace your - with a :. On 8/25/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, I have a quadBRI beronet ISDN card. Is there anybody who knows how to choose the channel to make calls? I tried with Dial(mISDN/1-1/) to choose channel 1 of port 1 but without success. TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk speaks Russian!
Westany speaks biz CP On Aug 30, 2006, at 9:50 AM, Stuart wrote: Westany, the Asterisk voice experts, announce their first Russian voice for ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 and Asterisk
Francisco Seratti wrote:
Hi pals, im trying to save some money in cellphones calls, so i bought a
GSM gateway and a Sipura SPA3000 gateway.
The GSM gw is currently working, and now im trying to configure the SPA,
but every call i send, i get a 503 service unavailable.
Im using this extension to match cell calls and sendthem to the spa:
exten = _15.,1,Dial(SIP/300/${EXTEN}) ; cellphones are 15 and
300 is the spa3000 extension, registered OK
exten = _15.,2,Hangup
In the dialplan section of the sipura, i ve tried many different options
like xx.:@gw0, (xx.) and many others.
I cannot find a formal configuration doc for this device, so if you
giveme a hand to configure it or tellme where to start, or where
is the problem i would be very pleased.
This may be a stupid question, but are you sure you are using the
registration for the FXO port (PSTN Line) and not the FXS port (line 1)
? I usually don't give my trunk lines extension numbers is why I ask.
-Dave
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speex Problemz
Ninneman, Tj wrote: Hi all, I've compiled/installed both * and Speex but I'm getting an error upon * startup: ...Aug 30 11:23:34 WARNING[27652]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/codec_speex.so: undefined symbol: speex_preprocess_ctl Aug 30 11:23:34 WARNING[27652]: loader.c:554 load_modules: Loading module codec_speex.so failed! Just wondering if anybody else has come across this error. Thanks! I seem to remember seeing an error like this when I had mis-matched versions of speex and codec_speex on the same system. I had compiled asterisk against speex version 1.1.12 but had version 1.0.5 installed on the box. If you have compiled them both on the same machine I'd make sure the old speex libs aren't hiding around somewhere. -Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk speaks Russian!
On Wed, Aug 30, 2006 at 05:50:53PM +0100, Stuart wrote: Westany, the Asterisk voice experts, announce their [ snip product description, that ommited a price tag of 124$ ] There¹s simply no substitute for knowledge and experience. Reading list descriptions also helps. This list is not asterisk-biz . -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk = Master and Slave ?
Noc Phibee wrote:
Hi
a small question:
I have one Asterisk Server with:
VoIP Provider gateway for incomming/outgoing call
5 VoIP Phone
(i name it Master)
i want add a another Asterisk server but only connected to:
5 new VoIP Phone
To the master for incoming/outgoing call (in g729)
It's possible ?
anyone have a sample of config ?
thanks
___
It's fairly simple, you can do it by setting up a pair of freinds (in
either IAX or SIP) and directing calls with the dialplan.
eg. for a setup where Master has extensions 1001 to 1005 as defined in
sip.conf as SIP/Voiphone1,SIP/Voipfone2 etc. and Slave has extensions
1006 to 1010 as defined in sip.conf as SIP/Voipfone6, SIP/Voipfone7 etc.
Master
==
in iax.conf.
[Slave]
type=friend
username=slave
host=ip.of.slave
auth=rsa {If they are communicating on an insecure network,
inkeys= rsa keyname {you should use RSA authentication and generate
outkey= rsa keyname2{keys in /var/lib/asterisk/keys as appropriate.
context=incoming context name
peercontext=outgoing context name
disallow=all
allow=g729
in extensions.conf
exten = 1001,1,Dial(SIP/Voiphone1) {change as appropriate
for options {and repeat for all 5
phones
exten = _100[6-9],1,Dial(IAX2/Master:[rsakeyname2[EMAIL PROTECTED]/{EXTEN})
{change as appropriate, and square
{brackets not used if the password
{is a plain test one rather than
{rsa keyname.
exten = 1010,1,Dial(IAX2/Master:[rsakeyname2[EMAIL PROTECTED])
pile of other rules to forward calls to voip provider
in iax.conf
[Master]
type=friend
username=master {probably unneccesary
host=ip.of.master
auth=rsa {If they are communicating on an insecure network,
inkeys= rsa keyname2 {you should use RSA authentication and
generateoutkey= rsa keyname {keys as appropriate.
context=incoming context name
peercontext=outgoing context name
disallow=all
allow=g729
in extensions.conf
exten = _100[1-5],1,Dial(IAX2/Slave:[rsakeyname[EMAIL PROTECTED]/{EXTEN})
{ above note
exten = 1006,1,Dial(SIP/Voiphone6) {repeat for extensions 1006-1010
exten = _X.,1,Dial(IAX2/Slave:[rsakeyname[EMAIL PROTECTED]/{EXTEN})
{ forward all other calls to Master
If you use voicemail you will probably need to decide where they will be
locally stored and setup an extension and add lines to the telephone
extension definitions as appropriate.
If you are not using rsa authentication, they you will need a secret
defined in each channel and the dialstring altered to match.
This is all off the top of my head, so may contain ommissions or typos.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you simultaniously dial multiple MSNs on one ISDN BRI b-channel?
Hello, Nobody has replied on this message. Isn't there anybody that has any input? Best regards, Henrik Woffinden Henrik Woffinden wrote: Hello, I'm fairly new to Asterisk. Installation went fine, and things seem to work, but I have 1 problem. Hardware: 2 HFC ISDN cards (1 in TE mode and 1 in NT mode) 1 SIP On the inside (NT mode card) I have 3 ISDN phones. Everything is connected with all cables and extra resistors, and all 3 phones can dial and be dialled. When I try to dial all 3 phones simultaniously, with Dial(Zap/g2/10Zap/g2/11Zap/g2/12,60,m(default)t) then 2 phones ring and the last one is busy/congestion. I assume its cause I only have 2 b-channels. How do I make all 3 phones ring using only 1 channel? It can be done. I also have a hardware PBX (Elmeg C46) which does that now. Can anyone help me how to do it in Asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] upgrade problem on IP phone 9133i
I think this is a known problem that was fixed in v1.3. I think you need to do this upgrade using a 'put' install via tftp client rather than trying to configure it to 'get' from a tftp server. It's been awhile so my memory is a bit foggy. I used pumpKIN. http://kin.klever.net/pumpkin/ -Original Message- From: Jean-Louis curty [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 30, 2006 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] upgrade problem on IP phone 9133i hi everybody, I bought few units for evaluation but we were not able to upgrade the firmware to 1.4 , it's currently set at 1.2, when we go to the webadmin page, whether we try to change the IP of the tftp server or the firmware name and set values, the reply is always Invalid IP address Please try again if we change the value via the phone itself , the tftp ip is changed but the firmware does not come up we are sure of our tftp server since it's used to upgrade other phones from other brands any idea ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ascom Eurit 133 cordless ISDN phone
Hi, I have an Ascom Eurit 133 ISDN base station with 2 cordless handsets. I can receive calls excellent on these phones, but when I dial out Asterisk can't see what number I want to dial, and it routes me to the s extension. That rather unlucky for an outgoing call not to know the number you want to dial. If I put the cable directly in the NT box it works fine. I have 2 other kind of ISDN phones, and the work fine out through the same Asterisk. Anyone know what could be the trouble here? ISDN hardware in the Asterisk box is 2 ZAPHFC cards (1 in TE mode, and 1 in NT mode). Asterisk 1.2.10-BRIstuffed 0.3.0-PRE-1s -- Best regards, Henrik Woffinden ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 501 config questions
Hi, I have a few questions on the Polycom 501. I am using latest firmware. 1) When I press the "Call List" button (on the left row of buttons), I get the call lists (as expected). When I press the "Directory" button, I get the choice between Directory and Call lists. How can I make this button go to Directory immediately? 2) I have 2 extensions on my 501. (let's say 101 and 102). Because of my dialplan, it actually matters which one I dial out with. When I pick a contact out of the directory, it calls automatically using line 101. How can I make it call with 102? 3) In call lists, my numbers are listed as 555-555-. Yet my asterisk dial plan requires me (by design) to press 9 first. How can I make the phone put the 9 by itself? Thank you for any help you may give me, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you simultaniously dial multiple MSNs on one ISDN BRI b-channel?
Hi, I have not tested yet, but maybe Dial(Zap/g1) would work; Guess this would ring everthing on Group 1... Best regards, Martin Polainer Am Mittwoch, 30. August 2006 21:45 schrieb Henrik Woffinden: Hello, Nobody has replied on this message. Isn't there anybody that has any input? Best regards, Henrik Woffinden Henrik Woffinden wrote: Hello, I'm fairly new to Asterisk. Installation went fine, and things seem to work, but I have 1 problem. Hardware: 2 HFC ISDN cards (1 in TE mode and 1 in NT mode) 1 SIP On the inside (NT mode card) I have 3 ISDN phones. Everything is connected with all cables and extra resistors, and all 3 phones can dial and be dialled. When I try to dial all 3 phones simultaniously, with Dial(Zap/g2/10Zap/g2/11Zap/g2/12,60,m(default)t) then 2 phones ring and the last one is busy/congestion. I assume its cause I only have 2 b-channels. How do I make all 3 phones ring using only 1 channel? It can be done. I also have a hardware PBX (Elmeg C46) which does that now. Can anyone help me how to do it in Asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ bkdat.net - highspeed internet Ing. Willi Hambammer Hieflauer Straße 18 A-8790 Eisenerz Tel: +43 3848 60048-104 Fax: +43 3848 60048-150 Mob: +43 664 3834879 Web: http://www.bkdat.net/ E-mail: mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you simultaniously dial multiple MSNs on one ISDN BRI b-channel?
Hi, I've just tested that... And no, nothing on the channel rings. Henrik Woffinden Martin Polainer wrote: Hi, I have not tested yet, but maybe Dial(Zap/g1) would work; Guess this would ring everthing on Group 1... Best regards, Martin Polainer Am Mittwoch, 30. August 2006 21:45 schrieb Henrik Woffinden: Hello, Nobody has replied on this message. Isn't there anybody that has any input? Best regards, Henrik Woffinden Henrik Woffinden wrote: Hello, I'm fairly new to Asterisk. Installation went fine, and things seem to work, but I have 1 problem. Hardware: 2 HFC ISDN cards (1 in TE mode and 1 in NT mode) 1 SIP On the inside (NT mode card) I have 3 ISDN phones. Everything is connected with all cables and extra resistors, and all 3 phones can dial and be dialled. When I try to dial all 3 phones simultaniously, with Dial(Zap/g2/10Zap/g2/11Zap/g2/12,60,m(default)t) then 2 phones ring and the last one is busy/congestion. I assume its cause I only have 2 b-channels. How do I make all 3 phones ring using only 1 channel? It can be done. I also have a hardware PBX (Elmeg C46) which does that now. Can anyone help me how to do it in Asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 and Asterisk
Dave Fullerton escribi:
Francisco Seratti wrote:
Hi pals, im trying to save some money in
cellphones calls, so i bought a GSM gateway and a Sipura SPA3000
gateway.
The GSM gw is currently working, and now im trying to configure the
SPA, but every call i send, i get a 503 service unavailable.
Im using this extension to match cell calls and sendthem to the spa:
exten = _15.,1,Dial(SIP/300/${EXTEN}) ; cellphones are 15
and 300 is the spa3000 extension, registered OK
exten = _15.,2,Hangup
In the dialplan section of the sipura, i ve tried many different
options like xx.:@gw0, (xx.) and many others.
I cannot find a formal configuration doc for this device, so if you
giveme a hand to configure it or tellme where to start, or where
is the problem i would be very pleased.
This may be a stupid question, but are you sure you are using the
registration for the FXO port (PSTN Line) and not the FXS port (line 1)
? I usually don't give my trunk lines extension numbers is why I ask.
-Dave
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__ NOD32 1.1730 (20060829) Information __
This message was checked by NOD32 antivirus system.
http://www.eset.com
Dave, thanks for your time. Yes, im sure it was the FXO port that i
regitered in the Asterisk. Do you know in which cases a 503 "Service
Unavailable" is obtained?
I also configured the syslog for this line and im getting just before
the 503 response, the line: "151AUD:Stop PSTN Tone". I dont
know what is this, but maybe a clue. If you need an extra data or
config,, ask to me.
--
Francisco Seratti
Sunesys Telecomunicaciones
Bouchard 644. 5to A. Puerto Madero
[EMAIL PROTECTED]
Tel: (54) 011- 4311-9009 (Rotativas)
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] visual indication of temp. closed mode
I may have to do something like that to be able to setup some way to temporarily close our office. I haven't really found anything else that would have a visual indicator that the system is on temp. closed mode. I can manually set a database entry (which I already do), and I know I can add an extension to set this, but neither of those visually alerts our receptionist.
Right now, we still have a Cisco SCCP phone on the system. She hits the DND button, which the SCCP driver updates the database of this, and that's what I check.
I know this has come up several times, but so far, I haven't really seen any really good solutions. My receptionist system will more than likely be a Polycom IP601. I'm thinking of creating an extension that would be used only for this purpose.
Any thoughts or ideas?
Thanks!
On 8/28/06, Michiel van Baak [EMAIL PROTECTED] wrote:On 13:12, Mon 28 Aug 06, Michael Sampson wrote: This is what I have so far [app-set-mwi]
exten = *35,1,Answer exten = *35,n,Wait(1) exten = *35,n,Playback(please-enter-yourextension) exten = *35,n,Read(fromext,then-press-pound,,) exten = *35,n,Wait(1)
exten = *35,n,system(touch /var/spool/asterisk/voicemail/default/${fromext}/INBOX/msg0001.txt); exten = *35,n,Macro(hangupcall,) ; end of [app-set-mwi] For some reason I get a busy signal when I dial *35 from an ext.
I did some playing around and found that if I changed the heading to look like this [app-cf-busy-off] exten = *35,1,Answer exten = *35,n,Wait(1) exten = *35,n,Playback(please-enter-yourextension)
exten = *35,n,Read(fromext,then-press-pound,,) exten = *35,n,Wait(1) exten = *35,n,system(touch /var/spool/asterisk/voicemail/default/${fromext}/INBOX/msg0001.txt); exten = *35,n,Macro(hangupcall,)
; end of [app-cf-busy-off] It works fine. I'm pretty new to editing the extensions.conf files, why can't I make a new app and have it work?
did you doinclude = app-set-mwiin the context where the phone is?I guess not, so the phone wont know about *35--Michiel van Baak[EMAIL PROTECTED]
http://michiel.vanbaak.euGnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
___--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- Lacy MooreAspendora, Inc.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 and Asterisk
You will find here all the info that you need to
make the SPA3000 to work with Asterisk:
- Original Message -
From:
Francisco
Seratti
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, August 30, 2006 5:39
PM
Subject: Re: [asterisk-users] Sipura 3000
and Asterisk
Dave Fullerton escribió:
Francisco Seratti wrote:
Hi pals, im trying to save some money in
cellphones calls, so i bought a GSM gateway and a Sipura SPA3000 gateway.
The GSM gw is currently working, and now im trying to configure the
SPA, but every call i send, i get a 503 service unavailable. Im using
this extension to match cell calls and sendthem to the spa: exten
= _15.,1,Dial(SIP/300/${EXTEN}) ; cellphones are
15 and 300 is the spa3000 extension, registered OK exten =
_15.,2,Hangup In the dialplan section of the sipura, i ve tried
many different options like xx.:@gw0, (xx.) and many others. I
cannot find a formal configuration doc for this device, so if you giveme a
hand to configure it or tellme where to start, or where is the problem
i would be very pleased. This may be a stupid
question, but are you sure you are using the registration for the FXO port
(PSTN Line) and not the FXS port (line 1) ? I usually don't give my
trunk lines extension numbers is why I ask. -Dave
___ --Bandwidth and
Colocation provided by Easynews.com -- asterisk-users mailing list
To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
__ NOD32 1.1730 (20060829) Information __
This message was checked by NOD32 antivirus system. http://www.eset.com
Dave, thanks for your time. Yes, im sure it was
the FXO port that i regitered in the Asterisk. Do you know in which cases a
503 "Service Unavailable" is obtained?I also configured the syslog for
this line and im getting just before the 503 response, the line:
"151AUD:Stop PSTN Tone". I dont know what is this, but maybe a clue.
If you need an extra data or config,, ask to me.
-- Francisco
SerattiSunesys TelecomunicacionesBouchard 644. 5to A. Puerto
Madero[EMAIL PROTECTED]Tel: (54)
011- 4311-9009 (Rotativas)
___--Bandwidth and
Colocation provided by Easynews.com --asterisk-users mailing
listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 and Asterisk
Sorry,
http://voxilla.com/PNphpBB2-viewforum-f-14.html
Cheers,
- Original Message -
From:
Francisco
Seratti
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, August 30, 2006 5:39
PM
Subject: Re: [asterisk-users] Sipura 3000
and Asterisk
Dave Fullerton escribió:
Francisco Seratti wrote:
Hi pals, im trying to save some money in
cellphones calls, so i bought a GSM gateway and a Sipura SPA3000 gateway.
The GSM gw is currently working, and now im trying to configure the
SPA, but every call i send, i get a 503 service unavailable. Im using
this extension to match cell calls and sendthem to the spa: exten
= _15.,1,Dial(SIP/300/${EXTEN}) ; cellphones are
15 and 300 is the spa3000 extension, registered OK exten =
_15.,2,Hangup In the dialplan section of the sipura, i ve tried
many different options like xx.:@gw0, (xx.) and many others. I
cannot find a formal configuration doc for this device, so if you giveme a
hand to configure it or tellme where to start, or where is the problem
i would be very pleased. This may be a stupid
question, but are you sure you are using the registration for the FXO port
(PSTN Line) and not the FXS port (line 1) ? I usually don't give my
trunk lines extension numbers is why I ask. -Dave
___ --Bandwidth and
Colocation provided by Easynews.com -- asterisk-users mailing list
To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
__ NOD32 1.1730 (20060829) Information __
This message was checked by NOD32 antivirus system. http://www.eset.com
Dave, thanks for your time. Yes, im sure it was
the FXO port that i regitered in the Asterisk. Do you know in which cases a
503 "Service Unavailable" is obtained?I also configured the syslog for
this line and im getting just before the 503 response, the line:
"151AUD:Stop PSTN Tone". I dont know what is this, but maybe a clue.
If you need an extra data or config,, ask to me.
-- Francisco
SerattiSunesys TelecomunicacionesBouchard 644. 5to A. Puerto
Madero[EMAIL PROTECTED]Tel: (54)
011- 4311-9009 (Rotativas)
___--Bandwidth and
Colocation provided by Easynews.com --asterisk-users mailing
listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] upgrade problem on IP phone 9133i
good idea! I tried but it doesn't work either...[08/30/06 23:12:33] UDP packet receive failed[08/30/06 23:12:33] Invalid opcode (0) during transfer received[08/30/06 23:12:34] Sending ' firmware.st' to '192.168.0.101'[08/30/06 23:12:34] UDP packet receive failed[08/30/06 23:12:34] Invalid opcode (12800) during transfer received[08/30/06 23:12:43] UDP packet receive failed [08/30/06 23:12:43] Invalid opcode (12800) during transfer received[08/30/06 23:12:44] UDP packet receive failed[08/30/06 23:12:44] Invalid opcode (12800) during transfer received[08/30/06 23:12:53] UDP packet receive failed [08/30/06 23:12:53] Invalid opcode (12800) during transfer received[08/30/06 23:12:54] UDP packet receive failed[08/30/06 23:12:54] Invalid opcode (12800) during transfer received 2006/8/30, shadowym [EMAIL PROTECTED]: I think this is a known problem that was fixed in v1.3.I think you need to do this upgrade using a 'put' install via tftp clientrather than trying to configure it to 'get' from a tftp server.It's beenawhile so my memory is a bit foggy.I used pumpKIN. http://kin.klever.net/pumpkin/-Original Message-From: Jean-Louis curty [mailto:[EMAIL PROTECTED]]Sent: Wednesday, August 30, 2006 9:07 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] upgrade problem on IP phone 9133ihi everybody,I bought few units for evaluation but we were not able to upgrade the firmware to 1.4 , it's currently set at 1.2, when we go to the webadminpage,whether we try to change the IP of the tftp server or the firmware name andset values, the reply is alwaysInvalid IP address Please try againif we change the value via the phone itself , the tftp ip is changed but thefirmware does not come up we are sure of our tftp server since it's used to upgrade other phones from other brands any idea ?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 and Asterisk
Franciso, can you make a call to the outside world, from the FXS port
and going out the FXO port?
I mean, without Asterisk in between. (The SPA300 can be configured that way)
I'm asking because I remember having trouble with the SPA recognizing
that the FXO line was alive when I plugged in a Panasonic PBX line.
But when I connected it directly to the phone company it recognized the
voltage or whatever.
If I remember correctly, the 503 error message is exactly what I was
getting. But this was almost 2 years ago.
BarZ
Francisco Seratti wrote:
Dave Fullerton escribió:
Francisco Seratti wrote:
Hi pals, im trying to save some money in cellphones calls, so i
bought a GSM gateway and a Sipura SPA3000 gateway.
The GSM gw is currently working, and now im trying to configure the
SPA, but every call i send, i get a 503 service unavailable.
Im using this extension to match cell calls and sendthem to the spa:
exten = _15.,1,Dial(SIP/300/${EXTEN}) ; cellphones are 15
and 300 is the spa3000 extension, registered OK
exten = _15.,2,Hangup
In the dialplan section of the sipura, i ve tried many different
options like xx.:@gw0, (xx.) and many others.
I cannot find a formal configuration doc for this device, so if you
giveme a hand to configure it or tellme where to start, or where
is the problem i would be very pleased.
This may be a stupid question, but are you sure you are using the
registration for the FXO port (PSTN Line) and not the FXS port (line 1)
? I usually don't give my trunk lines extension numbers is why I ask.
-Dave
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__ NOD32 1.1730 (20060829) Information __
This message was checked by NOD32 antivirus system.
http://www.eset.com
Dave, thanks for your time. Yes, im sure it was the FXO port that i
regitered in the Asterisk. Do you know in which cases a 503 Service
Unavailable is obtained?
I also configured the syslog for this line and im getting just before
the 503 response, the line: 151AUD:Stop PSTN Tone. I dont know
what is this, but maybe a clue. If you need an extra data or config,,
ask to me.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk with PABX
At 10:20 AM 8/30/2006, you wrote: Well, I started writing a tutorial for programming dial plans and sent it to two people who claimed interest, never heard back from either so I stopped. It's hard to know if what I write would be useful to anyone, so I don't want to just post it without feedback. I'd be happy to keep going if it seemed there was a point. I see the world quite different from the average person and it's not always clear how I explain thing is useful to others. I'd hate to post something that just ended up confusing people. Ira How about creating some documentation? be to just start using *. And again, I'm not trying to knock TrixBox, but it's a box and if you fit in it, it's probably the best thing there is. Personally, when I started, it seemed too small and limiting so I learned to program my own dial plans. It's not trivial as the language we've been given is not really well thought out and poorly documented at best, but in the end it's done everything I've asked of it. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] oddity with TDM400P / Asterisk setup
Hi List,I am working with an Asterisk server running on Fedora Core 4. It has two TDM400P cards installed. There are 6 trunk ports and 2 (unused) analog line ports. There are 5 Polycom SoundPoint 501 SIP phones connected to the server, and a Linksys 24-port powered switch connecting everything. The * version running is 1.2.7.1. All of the ports on the switch with voice devices, including the server, have a service class of 5, while non-voice devices are connected to other ports that have a service class of best effort.The problem, which began this morning, is very elusive. Calls-in-progress from zap-to-sip or sip-to-zap or sip-to-Asterisk will drop at odd times during the call, anywhere from 2 minutes to 15 minutes into the call. At the same time the call drops, my SSH session to the server will hang. After 10 to 15 seconds, the output and input from ssh session appears on my terminal and I am able to resume working in the shell. Zap-to-Asterisk doens't seem to cause the problem. Only when I dial through to a SIP device does it seem to hang.Top reveals nothing out the ordinary, utilization wise, the disk has plenty of free space, and the arp cache doesn't ever indicate a duplicate IP address with the server's NIC, which I thought might have been the problem. I also attempted to move the server to another port on the switch. No improvement. Anybody have a problem like this?--Ted WallingfordBest Technology Strategy LLC440-864-6084 phone440-815-2083 fax[EMAIL PROTECTED]http://www.btstrategy.com --Ted WallingfordBest Technology Strategy LLC440-864-6084 phone440-815-2083 fax[EMAIL PROTECTED]http://www.btstrategy.com On Jul 13, 2006, at 3:02 PM, Warren (mailing lists) wrote:Ronald Wiplinger wrote: Kevin P. Fleming wrote: Can we please keep the discussions about carriers, money, jobs, work,etc. off of this list? This is not the place to discuss yourexperiences with _any_ company, it's a place to talk about Asteriskand using Asterisk.Please move flamewars and similar discussions to some other forum. I agree with you!Which place is in your opinion the right place?As long there is no other place, such messages will always pop up. How about the Asterisk-biz list?W___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk presence (from manager API)
Hello, Google is your friend: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState not today. I mentioned in my original mail, that ExtensionState is unrealiable too. Sometimes I quit my softphone and I see extension as Idle (status 0), sometimes I log in and the extension is shown as unavailable. Juraj. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown CLI output
OK, changed the register interval for the Linksys PAP2T to 10 times longer and the output described earlier on the CLI also appears to follow the same schedule. I guess I'll have to check a Linksys list to see what could be causing this and if I should expect things to get worse. On Aug 29, 2006, at 8:21 PM, Carlos Leal wrote: I'm wondering if anyone can tell me what the following output, repeated about once per minute on my verbose=5 CLI , means. -- Contact header: transport -- Contact header: q -- Contact header: transport -- Contact header: q I'm on the latest version, 1.2.11, and am recovering from a too- near lighting strike that caused damage here and there. Asterisk is back up, minus a clone FXO card that the phone company said was causing a short in the phone line. SIP and IAX lines seem to work normally again except for this message that pops up about once a minute. Could it be a PAP2T that refreshes registration every 60 seconds? If so, what's changed? Thanks.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 config questions
On Aug 30, 2006, at 2:58 PM, Mike wrote: Hi, I have a few questions on the Polycom 501. I am using latest firmware. 1) When I press the Call List button (on the left row of buttons), I get the call lists (as expected). When I press the Directory button, I get the choice between Directory and Call lists. How can I make this button go to Directory immediately? 2) I have 2 extensions on my 501. (let's say 101 and 102). Because of my dialplan, it actually matters which one I dial out with. When I pick a contact out of the directory, it calls automatically using line 101. How can I make it call with 102? Pick up 102, then select contact 3) In call lists, my numbers are listed as 555-555-. Yet my asterisk dial plan requires me (by design) to press 9 first. How can I make the phone put the 9 by itself? It will not. either add to your contact entries, or alternatively have your dial plan add 9 to any exten longer than say 3 digits Thank you for any help you may give me, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText Queue Notification
Hi, we have few cisco's...is there a way to push the queue information to the phone ?thanks in advance,jean-louis2006/8/24, Brodie Macleod [EMAIL PROTECTED]:I know this isn't answering your question, but what I did for queue notification was use softkeys on the phones that call a PHP script on the *box that'll output XML for the phone to parse and display the queue stats ondemand. Of course your phone would need to have an XML parser or some other type of minibrowser.For sending SIP messages to my Snom phones I use Sipsakto display agent login info and their associated queue(s) so that it's easyfor agents to know what their status is.-Brodie On Thursday 24 August 2006 10:33 am, John D. Coleman wrote: I was wondering if anyone was able to execute custom commands on a channel once a caller connects to an agent after being in a queue.The reason I ask, is because I would like to use SendText to send a message to the agent receiving the call to let the agent know how many calls are waiting in the queue.I tried using ChanSpy, but then SendText will send messages only to and from the caller who initiated the ChanSpy. One way I could get around this is if I found out how to use SendText from the commandline, like smsq. I'm pretty sure that's not possible because of the nature of SIP MESSAGE but I figured I'd ask. Thanks, John Coleman ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with PABX
Just post it. be sure to wear asbestos. someone is sure to take offense. someone else just as surely will silently find it useful Ira wrote: At 10:20 AM 8/30/2006, you wrote: Well, I started writing a tutorial for programming dial plans and sent it to two people who claimed interest, never heard back from either so I stopped. It's hard to know if what I write would be useful to anyone, so I don't want to just post it without feedback. I'd be happy to keep going if it seemed there was a point. I see the world quite different from the average person and it's not always clear how I explain thing is useful to others. I'd hate to post something that just ended up confusing people. Ira How about creating some documentation? be to just start using *. And again, I'm not trying to knock TrixBox, but it's a box and if you fit in it, it's probably the best thing there is. Personally, when I started, it seemed too small and limiting so I learned to program my own dial plans. It's not trivial as the language we've been given is not really well thought out and poorly documented at best, but in the end it's done everything I've asked of it. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- One day at a time, one second if that's what it takes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER Dispatcher Load Balance How-To?
Douglas Garstang wrote: -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 29, 2006 11:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To? Well, it really depends on what he's using the asterisk servers for. If it's for voicemail or apps, he'll just have to make sure that certain apps land on certain servers, and voicemail can be distributed for various things. If ser can do what I've heard/read it can do, it can handle all the basic call functions (i.e. forwarding) for plenty of calls. Also, if the asterisk servers are just acting as gateways (i.e. t1, e1, etc), then they will have no problem handling a load balanced configuration. To do that, you'd need to use the avpops module in OpenSER. You think Asterisk documentation is bad? Wait until you try and get that stuff to work. Douglas, I have found the OpenSER documentation to be quite good. While they often don't give a view of the whole picture, they are very good at explaining the purpose, parameters, and functions of each module. Example configurations of how each module interacts with other modules can be hard to find, but once you understand a few dozen main functions in the ~20 or so very useful modules, you can do just about anything. I will say that avpops is by far the most confusing! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] w as pause dialing issue
OK, so I had an issue where I needed to add a w when dialing out my POTS line. But now when the calls go out my VoIP providers the w makes the call fail. I am using freePBX and the only place I found to change this was in the extensions.conf which makes it global. Am I missing something where I can add this while using freePBX? W does not appear to be a valid entry on the trunk prefix or outbound dialing entries. I tried to find a freePBX forum from Google but the only thing that looked promising came up as page cannot be displayed for the past hour. Does anyone have a link to a freePBX forum? I would think this would be a nice feature to add so you can add your pause. I saw where you could add a ticket to the Trac but I would rather discuss it on a list before calling it a needed feature or open ticket. Has anyone experienced this? If so how did you overcome it? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can anyone recommend a large button sip phone for the elderly.
Not in this case -- this is simply the phone doing something configurable when it receives a plain old ring on the line. We're not necessarily talking about the old phones in which the changed voltage on the line is actually shaking the bell around -- the phone would be smart enough to see the ring and do whatever the heck you wanted it to :) Chuck Bunn wrote: Hi, So am I to understand that the visual indicator responds the same way a ring would and thus if Asterisk tells a phone to ring the visual indicator uses that signal and does not require a separate signal? I guess I am use to seeing visual indicators in hotels that blink when there is a message waiting and other stuff like that and in that case I would assume that the visual indicator has multiple uses and it some how addressable? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:500,44f36ee6156882068143078! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
