Re: [Asterisk-Users] ZAP Line answer questio
On Thu, 3 Mar 2005, Eric Wieling aka ManxPower wrote: When you dialout using zap lines and sip phones, the sip connects to the zap channel and then dials the number, on the logs its shows sip = zap channel and when zap picks it up shows as answered but how can you really tell if the dialed number was answered or busy? If you are using analog ports then the call status information is not available. This is not an Asterisk thing, this is an analog thing. The call status can be avialable - if your telco provides that information via polarity changes. There were patches floating around to add support for this. I don't know if the full patch was added (support for answer supervision and disconnect supervision). The messages on the bug tracker messages seem to indicate that the complete stateful patch was not added. Going digital is better since you get a lot more information, especially from isdn. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] defold usernames in asterisk@home version 6
Dear Users, It is not my intention to install a working asterisk for real work. The intention is to learn about it just by playing aroung with it for a wile. I will have a good professional installing an * in my firm. The best way to find out what is possible with the * is to play with it for a wile. So a few starting points could help me a lot. From there I can learn more. I downloaded the iso. Installed it, and then opened the romote GUI on an other computer. I can not open the different parts becouse it asks the user name and the password. The passwords I have changed, that is not a problem at all, but I do not find the usernames documented somewhere. They are installed from the cd that is made of the above mentioned iso. I used the commands from help-aah and they work well. So, a little help here might be in place till I am started, then I only want to learn to configure the different config files, that's all. Thank you, Do not be angry at me! Willy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Friday, March 04, 2005 12:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] defold passwords in [EMAIL PROTECTED] version 6 Well, actually I guess it is help-aah not aah-help but I think you saw the response by mmiranda which will probably cover the sentiments of most here. He is correct to that if you are a linux noob, you need to go get a book and figure out the basics of linux before you embark on this project. Otherwise, you are just in for a world of pain as you try and work on a OS that you have no clue about and which is VASTLY different from Windows. This is assuming you even have the level of skill to teach yourself about Linux which hopefully you do. Dig deep, read long, and google you tail off. Then when you come back for help (which you will, we all do) at least you will be ready for what the gurus here (myself only being a humble jr. * user) can offer in assistance. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, March 03, 2005 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] defold passwords in [EMAIL PROTECTED] version 6 This is probably the worst forum to beg for help or be a noob. There is a strong sentiment that we should each do as much as possible to help ourselves before we come to the community for assistance. Not trying to be mean but you should just know that about this list. In your case, I am wondering where you downloaded your copy of [EMAIL PROTECTED] I got mine here and instructions for basic settings are there on the site. http://asteriskathome.sourceforge.net Password change options are available by typeing aah-help at the linux command prompt. You should actually see an announcement saying this if you log into the linux box with your root login and password. If you are looking for Asterisk docs, go here... http://www.voip-info.org/tiki-index.php?page=Asterisk If you are looking for docs on AMP then google Asterisk Management Portal. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Satchid Sent: Thursday, March 03, 2005 3:57 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] defold passwords in [EMAIL PROTECTED] version 6 Dear Users, I am begging for help. I just installed [EMAIL PROTECTED] This went amazingly well, this program is made for beginners like me. I do not know a thing of Linux and related programs and installation went very easy . Now I am so far that I can login to the GUI on a remote computer, but now I am stuk on the deffold user names. Where can I find the user names that are used in this program. Where can I find a user manual for [EMAIL PROTECTED] Thank you all, Willy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended Transfer (ATXFER) with CVS asterisk r 1_
Patch your chan_capi with this and you will be able to compile CVS HEAD http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 Jason On Thu, 03 Mar 2005 18:13:19 +0100, Massimo [EMAIL PROTECTED] wrote: Hi, I successfully installed asterisk 1.0 with Capi 0.35. In my pbx system I would like to use the atxfer function but is not included in the stable asterisk. Is there a way to include it in my version of asterisk: I did no used the last cvs because I can't compile the chan_capi .in it. :( Bye ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] notes: www.voicematch.cc speex 1.1.7, unrelated
Dear All, Two notes, completely unrelated. 1. only as thought food, www.voicematch.cc is doing voice authentication 2. speex 1.1.7 released (www.speex.org) Peace. Jason Sjobeck ICQ 5579183 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialing from a website. How to start...?
Hi all! We use a PHP-portal for management of our projects contacts. Now I would like to make it possible to dial contacts directly from the portal. Since users have to log in, I can use that to determine which office phone the call should originate from. And the number-to-be-dialed is of course also listed. How do I commence here? I'm pretty sure others have done this already, so I was wondering whether there's someone who can point me in the right direction... :-) (Preferable in PHP, since that's the flavor of choice of our portal) Regards, Evert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP hard phones choice
Hi everybody , I'm quite new in asterisk users, but really enjoy it ! I'd like to have non technical opinions about choosing SIP hard phones for asterisk. I'm studying VoIP implementation for a little company. I need to buy about 10 Phones. The most thing i need is a very good compatibility with Asterisk PBX, i must be sure the phones are going to register to it easily with no problems. And this for lowest cost possible of course:) If someone would have a comparative of different sip phones or something like this, I would really apreciate. Thanks in advance for your feedbacks and help. Nicolas Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bristuff e RealTime: STABLE vs. CVS-HEAD
Hi all! Was anybody able to install kapejod's zaphfc drivers together with RealTime application? I'm in big trouble because bristuff relay on STABLE version, while RealTime is included in the CVS-HEAD. I found this hint, Installing zaphfc with CVS-Head at http://voip-info.org/wiki-Asterisk+zaphfc+install, but it was written many months ago: may it be still useful? TIA, Alex ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting phpconfig to work?
On Thu, Mar 03, 2005 at 05:44:54PM +0300, Julius Kidubuka wrote: Hi, I have managed to re-install apache and php. I tried to install mod_php but it failed and returned the error below; === mod_php4-4.3.10_1,1 conflicts with installed package(s): php4-4.3.10_1 php4 is also the CGI version. If performance is not that an issue there is a resonable chance it will work as well. Anyway, didn't the freebasd packager make the common task of adding php support to apache simpler? -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CVS-HEAD change: queue/agent persistence
For anyone using CVS HEAD, if you are using queue member persistence or agent persistence, your next update will cause the persistence to break. The storage format for these elements has been changed so that it can be more easily extended in the future, but this required breaking compatibility. This should be the last time these features will be broken by an upgrade :-) Thanks for these informations. But what does it mean exactly update will cause the persistence to break. Which actions are required to maintain this feature after updating? Regards, Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bristuff e RealTime: STABLE vs. CVS-HEAD
Hi, -Original Message- Was anybody able to install kapejod's zaphfc drivers together with RealTime application? I'm in big trouble because bristuff relay on STABLE version, while RealTime is included in the CVS-HEAD. I found this hint, Installing zaphfc with CVS-Head at http://voip-info.org/wiki-Asterisk+zaphfc+install, but it was written many months ago: may it be still useful? Very doubtfull. In the mean time there have been a number of very radical changes to CVS-HEAD, which are not available in STABLE, therefore, not compatible with BRIstuff. You would be better off trying to backport RealTime into STABLE, I think... Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@home 0.6 + mISDN
Hi, I've a billion isdn0 card but i suppose than i cant patch kernel to use mISDN support because i've kernel 2.4 and patch on CVS i4l is for = 2.6.8 How do it ? help me please Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * intergation with Panasonic D500 and strange echo
Hi, all ! I have a situation like this: [SIP Terminals] - [*] -ISDN-PRI- [Panasonic D500] - Telecom (conn to Telecom is with second PRI card in Panasonic and 16 POTS lines). Panasonic has 2 ISDN PRI cards (one to Telco, and second to Asterisk), 16 POTS lines to telco and 32 (advanced hybrid telephone type) extensions. Idea is to have possibility to have some users on SIP terminals (Cisco 7912 SIP for now), and to have some users on panasonic extensions. Everything is working OK (SIP to/from extension (SIP or Panasonic's), SIP to outside line (it goes out on Panasonics PRI to telco)), except certain type of incoming calls from telco. When a call arrives through the PRI connected to telco and panasonic routes the call using DID to SIP ext. through the Asterisk, everything works fine (i.e no echo), but if a call arrives to Panasonic either throuhg PRI or through POTS lines and gets picked up by attendant and then transfered to my SIP phone, I (called party) have an very loud echo of my voice. Calling party has no problems with echo. If I plug asterisk's PRI into telco - no problems at all, but this situation doesn't suite my needs. Tried with all possible combinations of echocancel, echocancelwhenbridged, echotraining, different compile options in zconfig.h (MMX enabled/disabled, diffrent echo cancellers, aggressive suppresor on/off and etc. , noapic and nolapic in kernel parameters, but with no success at all. Little about configuration: Intel 865 chipset board, P4 Celleron 2.8 Ghz processor, and 1Gbyte of RAM, TE110P in PCI slot with its own IRQ. Linux is Debian with debian 2.6.8 - 2.6.10 kernels tested (no SMP, multithreading support since I cannot unload wct11xp module if SMP is enabled in kernel - freezes machine). Asterisks tested are 1.0.3 - 1.0.6 STABLE and couple of HEADs from CVSs all with same symptoms. ISDN-PRI (both * to Panasonic and Panasonc to telco) are EuroISDN, hdb3, ccs and there are NO alarms/errors on Telco, Panasonic or *. Anyone with similar configuration and problems, or ideas how to solve this ? Thank you very much. Regards, Nenad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home 0.6 + mISDN
Am Freitag 04 März 2005 11:27 schrieb Giovanni Miano: Hi, I've a billion isdn0 card but i suppose than i cant patch kernel to use mISDN support because i've kernel 2.4 and patch on CVS i4l is for = 2.6.8 use chan_capi Jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Answering Machine Detection with app_machinedetect.c
Hello, Is there any documentation available on how to use app_machinedetect.c to detect answering machine? Or is there anyone who can give me some pointers? We have compiled * with app_machinedetect.c, but not able to use it correctly in our configuration. Thanks, Aram Ter-Martirosyan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap channels intermittently bridging with SNOM190
Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GS102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked up with the SNOM is that sometimes when there are two active incoming calls viaZap channelsand the firstcaller hangs up while on hold the Zap channel doesn't detect the hangup correctly. What I end up with aftersome time is the twoactive Zap channels being bridged forever, or until I restart Asterisk. I think it's because the operator is not manually canceling a finished call and something in my dialplan is causing the channels to bridge when the calls are finished. I must have something wrong somewhere ? I've checked with the operator and she's said that she's been disconnecting any 'idle' calls i.e. when the remote user hangs up but yet the problem still occurs every now and again. This is what I end up with when I run a 'show channels': Channel (Context Extension Pri ) State Appl. Data Zap/1-1 (default 1 ) Up Bridged Call Zap/2-1 Zap/2-1 (default 2009 1 ) Up Dial SIP/switchboard|30|tr For reference call transferring on the SNOM is being done via the 'consultation transfer' method as set out in the SNOM manual. Perhaps there is a way in Asterisk to prevent/disallow bridging of specific Zap channels ? Has anyone else come across this phenomenon before ? Thanks in advance Kindest regardsDavid Wilson___D c D a t aTel +27 33 342 7003Fax +27 33 345 4155Cell +27 82 4147413http://www.dcdata.co.za[EMAIL PROTECTED]Powered by Linux, driven by passion ! ___ "Computers are not intelligent. They only think they are." ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialing from a website. How to start...?
Evert, The best way to do this is have your PHP code put a control file in the outgoing directory of Asterisk. This then invokes an Asterisk macro that calls the user, then transfers them to the contact. The format of the file is at: http://www.voip-info.org/wiki-Asterisk+auto-dial+out I run a consulting firm doing (amongst other things) Asterisk work. If you're interested, we can install Asterisk, configure it to talk to your telephone system, set up the click to dial, and integrate it with your PHP - email me off list. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Evert Meulie wrote: Hi all! We use a PHP-portal for management of our projects contacts. Now I would like to make it possible to dial contacts directly from the portal. Since users have to log in, I can use that to determine which office phone the call should originate from. And the number-to-be-dialed is of course also listed. How do I commence here? I'm pretty sure others have done this already, so I was wondering whether there's someone who can point me in the right direction... :-) (Preferable in PHP, since that's the flavor of choice of our portal) Regards, Evert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Beginning with Asterisk
Luz, Yes, this is possible with Asterisk. You probably already have a database system with predictive numbers to call, so you'll need some development to talk to that. For the Asterisk servers, this depends on quite a few factors such as what phones you're using, what database you're talking to, etc. You're probably looking a several modern server PCs, but you do need to plan your requirements in more detail first. You could use digium cards, but you'd need 13 TDM04B cards, which is not very practical. You're probably better of using E1/T1 cards in the Asterisk servers, then a channel bank to split this out to analogue lines. If you really want to do it right, get rid of the analogue lines to the outside world, and use E1s or T1s. For a project of this size, you definitely need professional help. The amount you spend will be much less than the cost of 50 operators sitting idle if the system breaks. My company does Asterisk consulting - drop me an email off list if you're interested. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Luz Lopez wrote: Hi All. I am beginning a project of Call center and predictive diales, my call center have 50 operators, I have 50 analog phone line with the company PTT in my country. I have the following questions: 1- Can I to work this project with Asterisk? 2- What caracteristic of hardware need for my servers? 3- For 50 analog phone line what tipe of card digium I need? Thanks in advanced, Regards. _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering Machine Detection with app_machinedetect.c
Aram jan tun es ? ---(RE) MSG (Start)--- Hello, Is there any documentation available on how to use app_machinedetect.c to detect answering machine? Or is there anyone who can give me some pointers? We have compiled * with app_machinedetect.c, but not able to use it correctly in our configuration. Thanks, Aram Ter-Martirosyan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---(RE) MSG (End)- Best regards, Tigran -- - \\\|||/// \ Tigran Petrossian \ http://www.hi-teck.com \ \ ~ ~ / \ Network, System Administrator \ | @ @ |\ mailto:[EMAIL PROTECTED] \ oOo---(_)---oOo--\--- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: dialing from a website. How to start...?
Thanks for the info! That's exactly the pointed I needed! ;-) (but I'll implement it myself. Cheaper...) ;-) ;-) Greetings, Evert Alistair Cunningham wrote: Evert, The best way to do this is have your PHP code put a control file in the outgoing directory of Asterisk. This then invokes an Asterisk macro that calls the user, then transfers them to the contact. The format of the file is at: http://www.voip-info.org/wiki-Asterisk+auto-dial+out I run a consulting firm doing (amongst other things) Asterisk work. If you're interested, we can install Asterisk, configure it to talk to your telephone system, set up the click to dial, and integrate it with your PHP - email me off list. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Evert Meulie wrote: Hi all! We use a PHP-portal for management of our projects contacts. Now I would like to make it possible to dial contacts directly from the portal. Since users have to log in, I can use that to determine which office phone the call should originate from. And the number-to-be-dialed is of course also listed. How do I commence here? I'm pretty sure others have done this already, so I was wondering whether there's someone who can point me in the right direction... :-) (Preferable in PHP, since that's the flavor of choice of our portal) Regards, Evert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Development help?
(Answering in private and in the list) On Thu, Mar 03, 2005 at 02:20:01PM -0800, Dan Austin wrote: I have a couple of quick questions that I do not see answered on the wiki or the Asterisk archives. I dropped an note to the Dev list and got an awating moderator Which basically means that you should subscribe to that list before posting there. http://lists.digium.com/mailman/listinfo/asterisk-dev -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bristuff e RealTime: STABLE vs. CVS-HEAD
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Florian Overkamp Sent: Friday, March 04, 2005 11:21 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Bristuff e RealTime: STABLE vs. CVS-HEAD Hi, Very doubtfull. In the mean time there have been a number of very radical changes to CVS-HEAD, which are not available in STABLE, therefore, not compatible with BRIstuff. You would be better off trying to backport RealTime into STABLE, I think... Florian Hi, thanks for your replay! Backporting RealTime into STABLE version sounds quite difficult: I'm not so skilled in Linux, but I could try. I've no idea about where to start. Do you have any link to suggest me, please? To have RealTime working, what about using chan_capi instead of bristuff? I read that chan_capi supports latest CVS-HEAD, but it is not completely clear to me whether it supports HFC based cards or not. Thanks again, Alex ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Why ${EXTEN} variable changes after Goto ?
In article [EMAIL PROTECTED],
Robert Rozman [EMAIL PROTECTED] wrote:
exten = 42,1,SetVar(SAVED_EXTEN=${EXTEN})
exten = 42,2,Goto(marvin,27,1)
thanks for help. I'd just like to be sure what happens if there is more than
one concurrent calls. Is variable set up for each of them or is necessary to
make variable that is somehow unique to each call ???
http://www.voip-info.org/wiki-Asterisk+cmd+SetVar tells you the answer is that
each call gets its own variable space.
There is also a global variable space which you can write to using SetGlobalVar.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Problems with g729 codec
Hello, I´m trying the g729 codec for testing pourpose. Whe I try to make a SIP call from a phone using g729 codec to another phone using another codec, when the destination phone answer, the call hangs up. this happend in both ways. In the asterisk console I get. Mar 4 13:11:35 NOTICE[24572]: channel.c:1724 ast_set_write_format: Unable to find a path from gsm to g729 What does it mean? Could this occur cause I am using the g729 without licence? If i buy a licence could solve my problem? Thanks. Ismael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialing from a website. How to start...?
Alistair, I may be confused, but I thought this was a users list, not really for advertising business activity? The last 2 posts of yours have been blatant adverts for your business..could you not take them off list? I may be fairly new here and may havegot the wrong impression, if so I'm sure I will be corrected. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alistair Cunningham Sent: 04 March 2005 11:19 To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dialing from a website. How to start...? Evert, The best way to do this is have your PHP code put a control file in the outgoing directory of Asterisk. This then invokes an Asterisk macro that calls the user, then transfers them to the contact. The format of the file is at: http://www.voip-info.org/wiki-Asterisk+auto-dial+out I run a consulting firm doing (amongst other things) Asterisk work. If you're interested, we can install Asterisk, configure it to talk to your telephone system, set up the click to dial, and integrate it with your PHP - email me off list. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Evert Meulie wrote: Hi all! We use a PHP-portal for management of our projects contacts. Now I would like to make it possible to dial contacts directly from the portal. Since users have to log in, I can use that to determine which office phone the call should originate from. And the number-to-be-dialed is of course also listed. How do I commence here? I'm pretty sure others have done this already, so I was wondering whether there's someone who can point me in the right direction... :-) (Preferable in PHP, since that's the flavor of choice of our portal) Regards, Evert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mISDN not initialising properly my Fritz cards
Hi all! I have two Fritz!PCI cards on a Debian with Kernel 2.6.9. I recompiled the Kernel to support mISDN and all went OK. Usually I initialize CAPI with this : alias /dev/capi20 avmfritz alias char-major-68-0 avmfritz install avmfritz /sbin/modprobe capi; \ /sbin/modprobe mISDN_core; \ /sbin/modprobe mISDN_l1; \ /sbin/modprobe mISDN_l2; \ /sbin/modprobe l3udss1; \ /sbin/modprobe mISDN_capi; \ /sbin/modprobe mISDN_x25dte; \ /sbin/modprobe --ignore-install avmfritz remove avmfritz /sbin/modprobe -r --ignore-remove avmfritz; \ /sbin/modprobe -r mISDN_x25dte; \ /sbin/modprobe -r mISDN_capi; \ /sbin/modprobe -r l3udss1; \ /sbin/modprobe -r mISDN_l2; \ /sbin/modprobe -r mISDN_l1; \ /sbin/modprobe -r mISDN_core; \ /sbin/modprobe -r capi ...and CAPIINFO starts reporting both cards ok. However, asterisk with chan_capi reports both cards (capi info) but never manages to uses them, OR, asterisk with chan_misdn never even gets to start saying No Upper ID port:1 / init_stack: No such device So I tried capiinit reload and it says : rootKABox:~# capiinit reload 1 mISDN detected mISDN1 - 2 mISDN detected mISDN2 - ERROR: missing config entry for controller 1 driver mISDN name mISDN1 ERROR: missing config entry for controller 2 driver mISDN name mISDN2 What the heck can be wrong ?!? Thanks for helping a (not so) newbie (anymore) M.G. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi with patch compilation error
Hi, I'm trying to make work chan_capi with last asterisk CVS. I installed last zaptel,libpri,last cvs ana patched chan_capi 0.35 with the patch kindly suggested me by Jason Williams: http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 First I received error 127 that I resolved commenting the line CC=gcc-2.95 but now I have this error: chan_capi.c: In function `load_module': chan_capi.c:2843: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_capi.c:2843: too many arguments to function `ast_channel_register' chan_capi.c: In function `unload_module': chan_capi.c:2863: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type make: *** [chan_capi.o] Error 1 Someone can help me ? Bye ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] timing/clock problem - SOLVED
Hi everybody, After some advice from you, we changed the order of my spans in the card. It was in slot 5 in this order. T1 (channelbank) T1 (channelbank) E1 (empty, with a loopback for testing porposes) E1 (Telco - PRI/ISDN) Now it is this way E1 (Telco - PRI/ISDN) E1 (empty, with a loopback for testing porposes) T1 (channelbank) T1 (channelbank) But when I chaged the order, I got too many lost interrupts. So I begin to chage slots and it gets OK in slot 2. I can sync to telco's clock as we hope! So, if you are having some problem - I think any kind of problem like interruptions, clock slips, HDLC (Abort and Bad FCS) - try EVERY configuration set as possible. I mean fisicaly configurations. try slot 1, 2, 3... every thing you can. Ok. It sounds like a Mandrake solution :-), but it can solve your problem. Thanks to everybody how helped me. -- Alex G Robertson NOC - Microlink ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SRV lookups
Anyone have comments on this? ty.. -Original Message- From: Matt Schulte Found this on the wiki, is this still true? If so then what's the alternative? Default srvlookup=yes If srvlookup is turned on, Asterisk supports DNS SRV lookups partially. Currently, Asterisk only reads the first SRV entry without bothering with priorities and weights. This option is turned on by default. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE110P module woes
Hi, I have been using asterisk for a couple of months now and for thee most part, I love it. However, I'm having a problem with the drivers of the Digium TE110P. I have tried both the Debian package and the CVS. I have tried several kernels, and am now at 2.6.11. This has been working before (with 2.6.8.1), but after a reboot it stopped working and I am not able to consistently make it work or fail. I have make clean, make and make install, no complains from make. The zaptel module loads fine and says so: Zapata Telephony Interface Registered on major 196 But the module for the TE110P fails. If I only modprobe it, it loads silently; but the moment I execute ztcfg, I get: ZT_SPANCONFIG failed on span 1: No such device or address (6) If I ask for more verbose errors, I get: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) Any ideas? -- Alfredo Sola ASP5-RIPE ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Audio pausing over IAX trunk
Hi, -Original Message- I am having a problem with periodic breaks in audio over an IAX trunk. The interruption only happens in one direction, and (I think) only with clients built on the open source libiax. Codec is irrelevant, and jitterbuffer on/off seems to make no difference either. The pause happens every few seconds, and is regular. If I disable trunking, audio is perfect. I am running CVS HEAD as of 1st March. Can anyone shed any light on this? No idea if this is the same as what you are describing but I'm seeing this effect on multihomed boxes using IAX2 (between 2 asterisk boxes) without trunking on recent STABLE... Weird. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with inbound call quality.
I wonder where I should start looking for problems when my symptoms are: * Good quality on outbound (X-lite - asterisk - PSTN via X100P) calls. * Bad quality, very low volume and some distortion, on outbound (PSTN - asterisk - X-lite via X100P) calls. Regards, Jakob ### Detta mail har blivit skannat av F-Secure Anti-virus for Microsoft Exchange. For mer information, ga till http://www.F-Secure.se This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.F-Secure.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ZAP Line answer questio
I agree with you. So the best choice would be to get a partial E1/T1 and when needed, a full E1. Im in Mexico, so E1's here :) R2 signalling. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Viernes, 04 de Marzo de 2005 02:04 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ZAP Line answer questio On Thu, 3 Mar 2005, Eric Wieling aka ManxPower wrote: When you dialout using zap lines and sip phones, the sip connects to the zap channel and then dials the number, on the logs its shows sip = zap channel and when zap picks it up shows as answered but how can you really tell if the dialed number was answered or busy? If you are using analog ports then the call status information is not available. This is not an Asterisk thing, this is an analog thing. The call status can be avialable - if your telco provides that information via polarity changes. There were patches floating around to add support for this. I don't know if the full patch was added (support for answer supervision and disconnect supervision). The messages on the bug tracker messages seem to indicate that the complete stateful patch was not added. Going digital is better since you get a lot more information, especially from isdn. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi with patch compilation error
The HEAD version was changed last night to be incompatible with the patch I provided, My C skills are not good enough to fix this so you need to checkout from cvs yesterdays code cvs checkout -D 03/03/05 asterisk Jason On Fri, 04 Mar 2005 13:51:51 +0100, Massimo [EMAIL PROTECTED] wrote: Hi, I'm trying to make work chan_capi with last asterisk CVS. I installed last zaptel,libpri,last cvs ana patched chan_capi 0.35 with the patch kindly suggested me by Jason Williams: http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 First I received error 127 that I resolved commenting the line CC=gcc-2.95 but now I have this error: chan_capi.c: In function `load_module': chan_capi.c:2843: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_capi.c:2843: too many arguments to function `ast_channel_register' chan_capi.c: In function `unload_module': chan_capi.c:2863: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type make: *** [chan_capi.o] Error 1 Someone can help me ? Bye ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P module woes
Just to confirm that you also powered down and up? I've no experience with the TE110, but this is a known problem with the TE405 and TE410. They apparently can get locked up, and only a power cycle will clear it. Regards Scott Stingel www.evtmedia.com Alfredo Sola wrote: Hi, I have been using asterisk for a couple of months now and for thee most part, I love it. However, I'm having a problem with the drivers of the Digium TE110P. I have tried both the Debian package and the CVS. I have tried several kernels, and am now at 2.6.11. This has been working before (with 2.6.8.1), but after a reboot it stopped working and I am not able to consistently make it work or fail. I have make clean, make and make install, no complains from make. The zaptel module loads fine and says so: Zapata Telephony Interface Registered on major 196 But the module for the TE110P fails. If I only modprobe it, it loads silently; but the moment I execute ztcfg, I get: ZT_SPANCONFIG failed on span 1: No such device or address (6) If I ask for more verbose errors, I get: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] DyDNS + externip
So is this where SER comes into play? Letting ppl register thru SER and use asterisk behind it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connection time of Transferred Calls
This probably has nothing to do with Asterisk, but I'm hoping someone can point me in the right direction My senario is Phone A is a hardware SIP phone Phone B is a hardware SIP phone Phone C is a Microsoft RTC API SDK endpoint (http://msdn.microsoft.com/library/default.asp?url=/downloads/list/clientapi.asp) All three are registered with the latest release version of Asterisk. A phones B. B phones C. B hangs up, A is connected to C. This all works fine, however there is a wait of about 10 seconds between B handing up, and A being connected to C. I've compared traces between C being an RTC endpoint, and C being a SIP phone. The SIP phone seems to send more ACKs than the RTC endpoint. However the trouble with the RTC is that all the SIP commands have been abstracted out, so I can't just send my own ACKs. Then again, these ACKs can't be essential as the transfers do work with the RTC endpoint, it's just that I get this 10 second delay. Any ideas are very welcome. And before you say it, the project I'm working on has insisted I use the Microsoft RTC, so no changing library 'm afraid. :-( Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio pausing over IAX trunk
I suspect that what you are hearing is the VAD/silence suppression kicking in and out. Unfortunately, I have the same complaints from my users and I have been unable to determine a way to disable silence suppression. VAD=no seems to have no effect in IAX. Also running CVS head from about 2 weeks ago. --- Rod Bacon Thu, 03 Mar 2005 17:05:20 -0800 I have looked through the archives, and can only find old references to this problem that appear to be no longer relevant, so I thought I'd ask again. I am having a problem with periodic breaks in audio over an IAX trunk. The interruption only happens in one direction, and (I think) only with clients built on the open source libiax. Codec is irrelevant, and jitterbuffer on/off seems to make no difference either. The pause happens every few seconds, and is regular. If I disable trunking, audio is perfect. I am running CVS HEAD as of 1st March. Can anyone shed any light on this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Question
We have also been looking at various GUI's for Asterisk... ([EMAIL PROTECTED] being one)... can anyone recommend one that would be ideal for a business user in a basic small / medium office environment? Depends on what you mean by GUI Simple GUI to edit/view Asterisk's config : - phpconfig : available from digium's CVS, instructions to install http://www.voip-info.org/tiki-index.php?page=Asterisk+gui+phpconfig. Let you configure Asterisk thru a browser - AsWeAdTo (Asterisk Web Admin Tool) : similar to phpconfig, I coded this before I knew phpconfig : http://www.marccharbonneau.com/asterisk/asweadto_v_0_0_2.tar advantage over phpconfig is that you can reload SIP/extensions/all and it let you restart asterisk (phpconfig only let you do a reload) More advanced GUI to edit/view Asterisk's config : - AMP (Asterisk Management Portal) : http://amp.coalescentsystems.ca/ Comes pre-installed with [EMAIL PROTECTED], easy to setup things in it, but some peoples will tell you that it gets in the way when you want to make more complicated stuff. Other : - Flash Operator Panel : http://www.asternic.org/ Comes pre-installed with [EMAIL PROTECTED] This is a must for any asterisk install, I think. Go check the demo on the site I must be forgetting something, I just woke up and didn't finish my first coffee yet. But I'm confident somebody else will fill in the gap ;) hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with g729 codec
On Fri, 2005-03-04 at 13:29 +0100, [EMAIL PROTECTED] wrote: Hello, I´m trying the g729 codec for testing pourpose. Whe I try to make a SIP call from a phone using g729 codec to another phone using another codec, when the destination phone answer, the call hangs up. this happend in both ways. In the asterisk console I get. Mar 4 13:11:35 NOTICE[24572]: channel.c:1724 ast_set_write_format: Unable to find a path from gsm to g729 What does it mean? Could this occur cause I am using the g729 without licence? If i buy a licence could solve my problem? G729 will not work without a license. The error message above told you that asterisk couldn't find a valid path to convert from gsm audio to g729 audio data. Seems that should have been very obvious from the error. It is well documented had you even decided to search. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering Machine Detection withapp_machinedetect.c
This is an english speaking listserve. Please speak english so others can know what you are talking about. -Matthew - Original Message - From: Tigran Petrossian [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 5:34 AM Subject: Re: [Asterisk-Users] Answering Machine Detection withapp_machinedetect.c Aram jan tun es ? ---(RE) MSG (Start)--- Hello, Is there any documentation available on how to use app_machinedetect.c to detect answering machine? Or is there anyone who can give me some pointers? We have compiled * with app_machinedetect.c, but not able to use it correctly in our configuration. Thanks, Aram Ter-Martirosyan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---(RE) MSG (End)- Best regards, Tigran -- - \\\|||/// \ Tigran Petrossian \ http://www.hi-teck.com \ \ ~ ~ / \ Network, System Administrator \ | @ @ |\ mailto:[EMAIL PROTECTED] \ oOo---(_)---oOo--\--- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk ---Toshiba
I set up a TE405P to go T1---*---Toshiba.
I have the channels configured, and can place calls from the Toshiba,
through * to the t1. Incoming calls work great to *, but if they go to
the Toshiba, I get a hangup. I think the * is sending the call to the
wrong span. I have 2 spans, span 1 from the T1, span 2 to the Toshiba.
The bchannels show as 0/1 through 0/23 on both spans in * when it
starts. Should the span 2 be different, since the channels are 25-47?
Ztcfg -vv show no errors, and shows all channels. Group 0 is the T1,
group 1 is the Toshiba.
Here is what I am getting, and my confs...
Executing Ringing(Zap/1-1, ) in new stack
-- Accepting call from '8016947881' to '6018' on channel 0/1, span 1
-- Executing Dial(Zap/1-1, ZAP/g1/8016196000) in new stack
-- Called g1/8016196000
-- Zap/25-1 is making progress passing it to Zap/1-1
-- Zap/25-1 is ringing
-- Zap/25-1 answered Zap/1-1
-- Hungup 'Zap/25-1'
I also see this error
Mar 4 06:07:21 WARNING[13946]: chan_zap.c:7069 pri_fixup_principle:
Call specified, but not found?
Mar 4 06:07:21 WARNING[13946]: chan_zap.c:7711 pri_dchannel: Ringing
requested on channel 0/1 not in use on span 1
Extensions.conf
exten = _6XXX,1,Answer
exten = _6XXX,2,Dial(ZAP/g1/${EXTEN})
exten = _6XXX,3,congestion()
Zapata.conf
[channels]
switchtype=national
context=from-pstn
signalling=pri_cpe
pridialplan=unknown
usecallerid=asreceived
echocancel=no
echocancelwhenbridged=no
echotraining=400
overlapdial=yes
immediate=no
group=0
channel = 1-23
context=from-ctx
switchtype=national
pridialplan=unknown
signalling=pri_net
usercallerid=asreceived
echocancel=yes
busydetect=yes
overlapdial=yes
immediate=no
echocancelwhenbridged=no
echotraining=400
group=1
channel = 25-46
Zaptel.conf
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48
Thanks guys!!
Dan
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Re: [Asterisk-Users] Problem getting Voice Contract script to work
On Fri, 2005-03-04 at 22:31 +0800, mechaman wrote: Hi, wondering if anyone can help me with my problem. I can't get the verify.agi script to work in Asterisk This script is available for download at http://www.sineapps.com/downloads.php The agi script works for recording and playback when accessing it directly at it's extension, but will not record anything when doing the flashhook procedure during a call. Recording is cut off after the flashhook What interfaces are you using? Did you realize that it is making a three way call to get the recorder in the mix? So do you have three-way calling enabled on whatever interfaces you are using? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS-HEAD change: queue/agent persistence
Guido Hecken wrote: But what does it mean exactly update will cause the persistence to break. Which actions are required to maintain this feature after updating? It will break one time only. There is no action required on your part; when you bring up the new version of Asterisk you will likely have none of your persistent members/agents restored. From then on, the persistence behavior will work again across future restarts. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] country/city codes
Mine will be shortly -- http://voiprates.us/rateengine It'll return the ISO country code (if available), the country's dial-prefix and a short descriptor of the kind of call made (city, area, type of service). It will also return the two or three best IAX termination rates to the dialed number. I'm working on integrating this rate-engine into my * install for a low-maintenance LCR option. -Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: Friday, March 04, 2005 8:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] country/city codes but I'd rather trust my business to a database lookup. I agree. Is there one somewhere that is publicly available? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why ${EXTEN} variable changes after Goto ?
On Friday 04 March 2005 02:00 am, Robert Rozman wrote:
exten = 42,1,SetVar(SAVED_EXTEN=${EXTEN})
exten = 42,2,Goto(marvin,27,1)
thanks for help. I'd just like to be sure what happens if there is more
than one concurrent calls. Is variable set up for each of them or is
necessary to make variable that is somehow unique to each call ???
It should be pretty easy to try it and see. I believe SetVar is unique to the
call.
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[Asterisk-Users] Broadvoice + incoming call works only for ~2 minutes
Hi, all. The asterisk setup is working fine, receiving calls via broadvoice initially. When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in after about 3 min, asterisk does not pick up the call any more. When I check the status of peers and registry from asterisk, it still says it's registered to broadvoice fine. However, call doesn't come through. Would you have any idea why? Any help will be much appreciated. Thanks Woojin Here's the excerpt from sip.conf [broadvoice] type=friend nat=no host=sip.broadvoice.com fromdomain=sip.broadvoice.com username=5083021402 fromuser=5083021402 secret=password-for-1st-BV-account dtmfmode=inband context=sip canreinvite=no insecure=very [broadvoice2] type=friend nat=no host=sip.broadvoice.com fromdomain=sip.broadvoice.com username=5083021425 fromuser=5083021425 secret=password-for-2nd-BV-account dtmfmode=inband context=sip canreinvite=no insecure=very and here's the output from sip show peers and sip show registry *CLI sip show peers Name/username HostDyn Nat ACL Mask Port Status grandstream1/grandstream1 (Unspecified)D 255.255.255.255 0 Unmonitored phone2/phone2 (Unspecified)D 255.255.255.255 0Unmonitored phone1/phone1 192.168.1.108D 255.255.255.255 5060 Unmonitored simpleconnect-sip/wlee179 63.218.92.199 255.255.255.255 5060 Unmonitored broadvoice2/5083021425 147.135.0.128 255.255.255.255 5060 Unmonitored broadvoice1/5083021402 147.135.0.128 255.255.255.255 5060 Unmonitored 6 sip peers [4 online , 2 offline] *CLI sip show registry HostUsernameRefresh State sip.broadvoice.com:5060 [EMAIL PROTECTED] 2038 Registered sip.broadvoice.com:5060 [EMAIL PROTECTED] 2038 Registered ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy and Private IP
This setup can cause any problems to the comunication process? I'm aware that the IAX2 protocol is NAT friendly so I think this will work, but to be sure I want to hear some oppinions. Should be no problems with this ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P in the UK - seems to short the dialtone
Hi A quicky has anyone had success with an X100P in the UK ? Im seeing some oddness - when I plug the wall socket into the card, I lose dialtone. If I unplug it, it comes back. Does this sound familiar to anyone ? Cheers Nigel Nigel Taylor Technology Director ITAzure Limited Dunn House Warren Park Way Enderby Leicestershire LE19 4SA 0116 286 3016 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP trunk: asterisk - callmanager
Yeah I know, its an old post, but I have just the OPPOSITE problem. I can
all out from my Cisco SIP phones across the SIP trunk (CCM - *) but not the
reverse. Any help would be greatly appreciated...
Sip.conf
[labcm33]
type=friend
host=1.2.3.4
context=incoming
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes
Extensions.conf --
[outgoing]
exten = _14XXX,1,ChanIsAvail(SIP/labcm33)
exten = _14XXX,2,Cut(AVAILCHAN=AVAILCHAN,,1)
exten = _14XXX,3,Dial(${AVAILCHAN},${ARG1})
exten = _14XXX,4,Hangup
exten = i,1,Congestion
I tried several variations in extensions.conf from an example taken from the
wiki: http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration
Ultimately, I guess I don't know the format of the URL CCM is expecting:
exten = _14XXX,3,Dial(${AVAILCHAN},${ARG1}) returns:
Sip read:
SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK290ae2a3;rport
From: Tim sip:[EMAIL PROTECTED];tag=as2fdd9958
To: sip:1.2.3.4;tag=16777270
Date: Fri, 04 Mar 2005 15:56:47 GMT
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
9 headers, 0 lines
-- Got SIP response 400 Bad Request - 'Malformed/Missing URL' back
from 1.2.3.4
Transmitting:
ACK sip:1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK290ae2a3;rport
From: Tim Connolly sip:[EMAIL PROTECTED];tag=as2fdd9958
To: sip:1.2.3.4;tag=16777270
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 1.2.3.4:5062
-- SIP/labcm33-69a9 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/6101-b03a, ) in new stack
== Spawn extension (default, 14001, 4) exited non-zero on 'SIP/6101-b03a'
Reliably Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
3.4.5.6:38887;branch=z9hG4bKac10dc3d6f354228841c1e07ff7f
From: Timsip:[EMAIL PROTECTED];tag=10699268716550
To: sip:[EMAIL PROTECTED];tag=as6b37c0b8
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
...snip...?
403...circuit busy.. blah!
Any ideas?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Kemp
Sent: Tuesday, October 19, 2004 9:22 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP trunk: asterisk - callmanager
Pavel, have you resolved the CCM issue?
I have the same problem, I can place calls from Asterisk to CCM but not the
other way, same zero tcpdump when going CCM - Asterisk? Think it is
something to do with the CCM Media Termination Point but all shows OK.
Reams of CCM logs don't really say what is going on. Any ideas???
Dave
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[Asterisk-Users] IAX on netweb EEZEE phone
I'm running asterisk stable 1.0.5 and I'm trying to get the netweb eezee phone version v1.37.008 to talk IAX to asterisk. The pages I saw in the wiki maybe didn't hold my hand quite enough and the information on the eezee phone website appears to be for a different firmware version. If anyone has done this recently and has a working situation I would appreciate some useful hints about how to fill out the phone's settings and if there are any changes necessary in the iax.conf file, like the wiki suggests. I did get the phone to register SIP with Asterisk but when I set it to IAX and change what I think are correct parms I don't even seem to get messages on the console about a registration attempt. -Nate ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [OT] - [Asterisk-Users] Why should I answer a Newbie questio n, therethick!
-Original Message- From: Ronald Wiplinger [mailto:[EMAIL PROTECTED] Sometimes it is not the if you make a search, often is for new comers what to aks for. If you do not know the specific term, than you need to ask somewhere, and I think the list is good for that. Sure. So say, I tried a Googling for X, but I didn't have any luck. Then I looked at pages X and Y in the Wiki, but couldn't find anything that related to my problem. People are a lot more sympathetic if you demonstrate you've made some effort to find the answer on your own. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [OT] - [Asterisk-Users] Why should I answer a Newbie questio n,therethick!
-Original Message- From: Paul Fielding [mailto:[EMAIL PROTECTED] Frankly, I agree. If you don't like the question, feel it's lame or dumb, or don't like that someone hasn't done their research, then delete the message. Well, sometimes that works. But I've been on a lot of lists where newbies who thought they were being ignored started flaming people for not responding to them, writing posts badmouthing the project, hijacking other threads, accusing people of being cliquish, etc. Sometimes you just can't win. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zombie SIP channels
Ok - I finally found out what was causing the ZOMBIE channels. Now follow me on this one :) It appears that if you are using a Cisco 7960 and are on a call and want to transfer the call to another extension - if you press more and Trnsfer and dial the extension and you hit the Trnsfer button again before the extension answers, a ZOMBIE channel is created. If you use BlindXfer, it does not create the ZOMBIE channel. I have now informed my client that if they want to do a Blind Transfer, to use the BlindXfer softkey instead of the Trnsfer softkey or just use the # key to do a blind transfer. Now, I am running Asterisk CVS-v1-0-11/12/04-15:32:45. I would be interested in knowing if later versions of asterisk exhibited this same behavior. Any feedback would be appreciated. Thanks, Pedro On Fri, 11 Feb 2005 08:32:43 +0100, Florian Overkamp [EMAIL PROTECTED] wrote: Hi, -Original Message- Ok this is odd - caught it again twice today. The more I thought about what has changed on the server I realized that I was not using a timing device before, but am now using ztdummy. I if that could be causing the zombies? http://bugs.digium.com/bug_view_page.php?bug_id=0002938 I don't think so, but who knows. The patch resolves a locking issue that may or may not be timing-source dependant. I've seen the issue occur after call transfers in scenario's where I used a few chan_local's. Do yourself a favour: - If you can, unload the ztdummy and test for a while. However, this may put the issue to sleep - but it won't solve it! - After that, load ztdummy again and apply the two lines in channel.c. Test again. Good chance the issue will be gone. Report results here :) Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice + incoming call works only for ~2 minutes
Hi, Roger I think that may done the trick! I've put the qualify=100 and asterisk answered the call after about 5 min! The asterisk box is on public ip, also running firewall, so I thought it wouldn't need any nat related stuff, but may be I was wrong... Thanks! Woojin On Friday 04 March 2005 10:47 am, Roger Gulbranson wrote: On Fri, 2005-03-04 at 10:38 -0500, Woojin Lee wrote: Hi, all. The asterisk setup is working fine, receiving calls via broadvoice initially. When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in after about 3 min, asterisk does not pick up the call any more. When I check the status of peers and registry from asterisk, it still says it's registered to broadvoice fine. However, call doesn't come through. Would you have any idea why? Any help will be much appreciated. Thanks Woojin Here's the excerpt from sip.conf [broadvoice] type=friend nat=no host=sip.broadvoice.com fromdomain=sip.broadvoice.com username=5083021402 fromuser=5083021402 secret=password-for-1st-BV-account dtmfmode=inband context=sip canreinvite=no insecure=very [broadvoice2] type=friend nat=no host=sip.broadvoice.com fromdomain=sip.broadvoice.com username=5083021425 fromuser=5083021425 secret=password-for-2nd-BV-account dtmfmode=inband context=sip canreinvite=no insecure=very and here's the output from sip show peers and sip show registry *CLI sip show peers Name/username HostDyn Nat ACL Mask Port Status grandstream1/grandstream1 (Unspecified)D 255.255.255.255 0Unmonitored phone2/phone2 (Unspecified)D 255.255.255.255 0 Unmonitored phone1/phone1 192.168.1.108D 255.255.255.255 5060 Unmonitored simpleconnect-sip/wlee179 63.218.92.199 255.255.255.255 5060 Unmonitored broadvoice2/5083021425 147.135.0.128 255.255.255.255 5060 Unmonitored broadvoice1/5083021402 147.135.0.128 255.255.255.255 5060 Unmonitored 6 sip peers [4 online , 2 offline] *CLI sip show registry HostUsernameRefresh State sip.broadvoice.com:5060 [EMAIL PROTECTED] 2038 Registered sip.broadvoice.com:5060 [EMAIL PROTECTED] 2038 Registered Sounds like you have a firewall of some sort that closes down after about 2 minutes. Try a qualify=100 (or similar number) to provide some keep-alives. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P in the UK - seems to short the dialtone
I'm running two in a box Nigel and they work ok for me. Maybe a faulty card you have? Is the extension/socket you plug it in to working ok for a phone? Mike On Fri, 4 Mar 2005 15:48:53 -, Nigel Taylor [EMAIL PROTECTED] wrote: Hi A quicky has anyone had success with an X100P in the UK ? I'm seeing some oddness - when I plug the wall socket into the card, I lose dialtone. If I unplug it, it comes back. Does this sound familiar to anyone ? Cheers Nigel Nigel Taylor Technology Director ITAzure Limited Dunn House Warren Park Way Enderby Leicestershire LE19 4SA 0116 286 3016 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] defold usernames in asterisk@home version 6
OK. So check out the Wiki here http://www.voip-info.org/tiki-index.php?page=Asterisk The archive of this list can be search via google by entering... site:lists.digium.com some parameter www.digium.com has a link to all the materials for getting started in the Documentation section of the website. Those are really quite good so I would start there. Most were written prior to [EMAIL PROTECTED] being released so they document the real nuts and bolts of how to build a system from ground up. [EMAIL PROTECTED] is really quite turnkey so you don't learn as much. As for you specific question of usernames and passwords Linux Command Line: root and whatever password you set AMP GUI - maint and password (the password is password) Web address book - No idea Web Meet Me - maint and password (can be reset via help-aah) Web Voicemail - ext. # plus the PIN you setup Hope this helps. Please read up at the above sites and youw ill do fine. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Satchid Sent: Friday, March 04, 2005 1:21 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] defold usernames in [EMAIL PROTECTED] version 6 Dear Users, It is not my intention to install a working asterisk for real work. The intention is to learn about it just by playing aroung with it for a wile. I will have a good professional installing an * in my firm. The best way to find out what is possible with the * is to play with it for a wile. So a few starting points could help me a lot. From there I can learn more. I downloaded the iso. Installed it, and then opened the romote GUI on an other computer. I can not open the different parts becouse it asks the user name and the password. The passwords I have changed, that is not a problem at all, but I do not find the usernames documented somewhere. They are installed from the cd that is made of the above mentioned iso. I used the commands from help-aah and they work well. So, a little help here might be in place till I am started, then I only want to learn to configure the different config files, that's all. Thank you, Do not be angry at me! Willy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Friday, March 04, 2005 12:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] defold passwords in [EMAIL PROTECTED] version 6 Well, actually I guess it is help-aah not aah-help but I think you saw the response by mmiranda which will probably cover the sentiments of most here. He is correct to that if you are a linux noob, you need to go get a book and figure out the basics of linux before you embark on this project. Otherwise, you are just in for a world of pain as you try and work on a OS that you have no clue about and which is VASTLY different from Windows. This is assuming you even have the level of skill to teach yourself about Linux which hopefully you do. Dig deep, read long, and google you tail off. Then when you come back for help (which you will, we all do) at least you will be ready for what the gurus here (myself only being a humble jr. * user) can offer in assistance. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, March 03, 2005 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] defold passwords in [EMAIL PROTECTED] version 6 This is probably the worst forum to beg for help or be a noob. There is a strong sentiment that we should each do as much as possible to help ourselves before we come to the community for assistance. Not trying to be mean but you should just know that about this list. In your case, I am wondering where you downloaded your copy of [EMAIL PROTECTED] I got mine here and instructions for basic settings are there on the site. http://asteriskathome.sourceforge.net Password change options are available by typeing aah-help at the linux command prompt. You should actually see an announcement saying this if you log into the linux box with your root login and password. If you are looking for Asterisk docs, go here... http://www.voip-info.org/tiki-index.php?page=Asterisk If you are looking for docs on AMP then google Asterisk Management Portal. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Satchid Sent: Thursday, March 03, 2005 3:57 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] defold passwords in [EMAIL PROTECTED] version 6 Dear Users, I am begging for help. I just installed [EMAIL PROTECTED] This went amazingly well, this program is made for beginners like me. I do not know a thing of Linux and related programs and installation went very easy . Now I am so far that I can login to the GUI on a remote computer, but now I am
Re: [Asterisk-Users] Problems with g729 codec
G729 will not work without a licensecan't G729 work in passthrough mode without license? if yes, how to configure it work in passthrough mode? On Fri, 04 Mar 2005 08:50:11 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Fri, 2005-03-04 at 13:29 +0100, [EMAIL PROTECTED] wrote: Hello, I´m trying the g729 codec for testing pourpose. Whe I try to make a SIP call from a phone using g729 codec to another phone using another codec, when the destination phone answer, the call hangs up. this happend in both ways. In the asterisk console I get. Mar 4 13:11:35 NOTICE[24572]: channel.c:1724 ast_set_write_format: Unable to find a path from gsm to g729 What does it mean? Could this occur cause I am using the g729 without licence? If i buy a licence could solve my problem? G729 will not work without a license. The error message above told you that asterisk couldn't find a valid path to convert from gsm audio to g729 audio data. Seems that should have been very obvious from the error. It is well documented had you even decided to search. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: TE110P module woes
I upgraded to Redhat Enterprise Linux 4.0 and ran into a similar problem. Try looking at udev. The device files are being created dynamically and some configuration changes need to be made. Read the README.udev file in zaptel src directly. It has a brief explanation. Gene -Original Message- 3. TE110P module woes (Alfredo Sola) -- Message: 3 Date: Fri, 04 Mar 2005 14:01:58 +0100 From: Alfredo Sola [EMAIL PROTECTED] Subject: [Asterisk-Users] TE110P module woes To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, I have been using asterisk for a couple of months now and for thee most part, I love it. However, I'm having a problem with the drivers of the Digium TE110P. I have tried both the Debian package and the CVS. I have tried several kernels, and am now at 2.6.11. This has been working before (with 2.6.8.1), but after a reboot it stopped working and I am not able to consistently make it work or fail. I have make clean, make and make install, no complains from make. The zaptel module loads fine and says so: Zapata Telephony Interface Registered on major 196 But the module for the TE110P fails. If I only modprobe it, it loads silently; but the moment I execute ztcfg, I get: ZT_SPANCONFIG failed on span 1: No such device or address (6) If I ask for more verbose errors, I get: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) Any ideas? -- Alfredo Sola ASP5-RIPE ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P in the UK - seems to short the dialtone
Mike Thanks for the prompt reply. The line is fine. If I connect an analog phone I get ringtone and can call out. If I leave the phone connected and connect another wire from the socket into the X100P, I immediately lose the ringtone. I've checked The voltages and when the line is connected to the X100P, the line voltage drops from around 50 volts to 1 ish. It all feels like the X100P is shorting the line and is faulty but I'm, trying to make sure I'm not missing something. Thanks again Nige; -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Dent Sent: 04 March 2005 16:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] X100P in the UK - seems to short the dialtone I'm running two in a box Nigel and they work ok for me. Maybe a faulty card you have? Is the extension/socket you plug it in to working ok for a phone? Mike On Fri, 4 Mar 2005 15:48:53 -, Nigel Taylor [EMAIL PROTECTED] wrote: Hi A quicky - has anyone had success with an X100P in the UK ? I'm seeing some oddness - when I plug the wall socket into the card, I lose dialtone. If I unplug it, it comes back. Does this sound familiar to anyone ? Cheers Nigel Nigel Taylor Technology Director ITAzure Limited Dunn House Warren Park Way Enderby Leicestershire LE19 4SA 0116 286 3016 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
Hasn't anyone noticed that LiveVoip seems to happily blame just about everything on Asterisk? FWIW, I have experienced the same type of problem on a Sprint cell phone and also using a residential VOIP account with Broadvox. Both were able to correct the problem at THEIR end. Since no one else on this list seems to be complaining about the problem using provider's other than LV, I would suggest sacking them and getting DIDs from some other place. Seems like that is always the first thing they suggest too so they must not be that interested in your business. -mark On Mar 2, 2005, at 11:06 PM, Ryan Laginski wrote: Hi, I am experiencing the same problem as you. Ringback works great with the pstn or any other voip provider, but not with livevoip. I've just upgraded to 1.0.6 to see if that resolves the problem, but it has not. Please post back if you find a solution, I'll do the same. Thanks, -Ryan On Wed, 2 Feb 2005 13:25:29 -0500, Brian Dingman [EMAIL PROTECTED] wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with g729 codec
On Fri, Mar 04, 2005 at 04:37:06PM +, Asterisk guy wrote: G729 will not work without a licensecan't G729 work in passthrough mode without license? if yes, how to configure it work in passthrough mode? Passthrough means that the codec going in is the same as the codec going out. If you configure all your phones to be g729 then you can use passthrough. If you want to use prompts stored in something else (like wav or gsm) you'll need a licence for that... You said the other phone was using a different codec, hence the problem... On Fri, 04 Mar 2005 08:50:11 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: G729 will not work without a license. The error message above told you that asterisk couldn't find a valid path to convert from gsm audio to g729 audio data. Seems that should have been very obvious from the error. It is well documented had you even decided to search. Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P in the UK - seems to short the dialtone
On Friday 04 March 2005 11:43 am, Nigel Taylor wrote: The line is fine. If I connect an analog phone I get ringtone and can call out. If I leave the phone connected and connect another wire from the socket into the X100P, I immediately lose the ringtone. I've checked The voltages and when the line is connected to the X100P, the line voltage drops from around 50 volts to 1 ish. It all feels like the X100P is shorting the line and is faulty but I'm, trying to make sure I'm not missing something. Sounds like your phone line uses a 3-wire interface (common in the UK). You need a 3-wire to 2-wire adapter from your telco. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMS in 1.0.6
This is a heads up, I heard nothing about this I suppose in part because few of you use SMS, but note that the SMS application has changed the directory where it stores messages and the format of the file they are stored in. If ythe author or someone could chime in with a valid URL that mentions this it might be handy :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with g729 codec
sorry to ask, but what does it mean in passthrough mode ? On Fri, 4 Mar 2005 16:37:06 +, Asterisk guy [EMAIL PROTECTED] wrote: G729 will not work without a licensecan't G729 work in passthrough mode without license? if yes, how to configure it work in passthrough mode? On Fri, 04 Mar 2005 08:50:11 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Fri, 2005-03-04 at 13:29 +0100, [EMAIL PROTECTED] wrote: Hello, I´m trying the g729 codec for testing pourpose. Whe I try to make a SIP call from a phone using g729 codec to another phone using another codec, when the destination phone answer, the call hangs up. this happend in both ways. In the asterisk console I get. Mar 4 13:11:35 NOTICE[24572]: channel.c:1724 ast_set_write_format: Unable to find a path from gsm to g729 What does it mean? Could this occur cause I am using the g729 without licence? If i buy a licence could solve my problem? G729 will not work without a license. The error message above told you that asterisk couldn't find a valid path to convert from gsm audio to g729 audio data. Seems that should have been very obvious from the error. It is well documented had you even decided to search. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with g729 codec
On Fri, 2005-03-04 at 12:02 -0500, Erick Perez wrote: sorry to ask, but what does it mean in passthrough mode ? data, in this case audio, passes from one side through to the other with no need for modification. A standard serial cable is a passthrough cable. Same for standard network patch cables. The software here behaves much the same way, it picks the audio data out of the packet and passes it through to the other side of the communication. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
Mark Eissler wrote: Hasn't anyone noticed that LiveVoip seems to happily blame just about everything on Asterisk? FWIW, I have experienced the same type of problem on a Sprint cell phone and also using a residential VOIP account with Broadvox. Both were able to correct the problem at THEIR end. Since no one else on this list seems to be complaining about the problem using provider's other than LV, I would suggest sacking them and getting DIDs from some other place. Seems like that is always the first thing they suggest too so they must not be that interested in your business. -mark On Mar 2, 2005, at 11:06 PM, Ryan Laginski wrote: Hi, I am experiencing the same problem as you. Ringback works great with the pstn or any other voip provider, but not with livevoip. I've just upgraded to 1.0.6 to see if that resolves the problem, but it has not. Please post back if you find a solution, I'll do the same. Thanks, -Ryan On Wed, 2 Feb 2005 13:25:29 -0500, Brian Dingman [EMAIL PROTECTED] wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users is this using their asterisk city, or just a straight sip account?? Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Options in Brazil
If you speck portuguese, visit AsteriskBrasil.org: http://www.asteriskbrasil.org Regards. Denis. Em Qui 03 Mar 2005 22:23, Paul Davidson escreveu: All- I am considering an Asterisk implementation in Brazil. Unfortunately, this presents something of a challenge to plan sitting in Chicago, USA. I know there is a large section of Brazillian Asterisk users who actively read this list- so I'd love to pump out a few questions- note, I'm not necessarily a newbie, having successfully implemented a few Asterisk boxes here in the US. My primary question revolves around connection hardware- I need to plug in 8 POTS lines (I've no idea what they'd be called there) to an Asterisk box. Is digium's TDM400 series availble down there? Recommended? Undesirable? ATA's? (Sipura, presumably) - channel banks? If anyone has any solid knowledge they can share- gotchas appreciated- feel free to contact me off list. Thanks, -pbd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. On Fri, 04 Mar 2005 12:23:45 -0500, Cirelle Internet Products [EMAIL PROTECTED] wrote: Mark Eissler wrote: Hasn't anyone noticed that LiveVoip seems to happily blame just about everything on Asterisk? FWIW, I have experienced the same type of problem on a Sprint cell phone and also using a residential VOIP account with Broadvox. Both were able to correct the problem at THEIR end. Since no one else on this list seems to be complaining about the problem using provider's other than LV, I would suggest sacking them and getting DIDs from some other place. Seems like that is always the first thing they suggest too so they must not be that interested in your business. -mark On Mar 2, 2005, at 11:06 PM, Ryan Laginski wrote: Hi, I am experiencing the same problem as you. Ringback works great with the pstn or any other voip provider, but not with livevoip. I've just upgraded to 1.0.6 to see if that resolves the problem, but it has not. Please post back if you find a solution, I'll do the same. Thanks, -Ryan On Wed, 2 Feb 2005 13:25:29 -0500, Brian Dingman [EMAIL PROTECTED] wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users is this using their asterisk city, or just a straight sip account?? Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1953 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio pausing over IAX trunk
Rod Bacon wrote: I have looked through the archives, and can only find old references to this problem that appear to be no longer relevant, so I thought I'd ask again. I am having a problem with periodic breaks in audio over an IAX trunk. The interruption only happens in one direction, and (I think) only with clients built on the open source libiax. Codec is irrelevant, and jitterbuffer on/off seems to make no difference either. The pause happens every few seconds, and is regular. If I disable trunking, audio is perfect. I am running CVS HEAD as of 1st March. Can anyone shed any light on this? Not unless you can describe the problem more clearly. Which direction does this happen in, what exactly are these clients you're talking about, and what is does the network look like between the endpoints. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] - [Asterisk-Users] Why should I answer a Newbie question, therethick!
Ronald Wiplinger wrote: *snipped Sometimes it is not the if you make a search, often is for new comers what to aks for. If you do not know the specific term, than you need to ask somewhere, and I think the list is good for that. *snipped no, if you don't know a 'term' you search for a glossary! http://www.google.com/search?hl=enq=%22telecom+glossary%22btnG=Google+Search 86,700 hits for telecom glossary, i think that you should be able to find one to your liking. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] budgetphone
Hi all,
I registered a SIP account at budgetphone.nl/talkin2ya.nl
Receiving calls works like a charm, I even redirected my
normal PSTN number to the number I got from them so
everything ends up in my * server.
Before I ask them to take over my normal phone number I
wanted to test all of it, so I ordered some calling minutes
to test. Now I cannot get outbound calling to work with
them. Anyone here knows how to set it up ?
Some more info:
Asterisk CVS-HEAD as of 15-02-2005
My sip.conf
[general]
context=from-sip
realm=vanbaak
port=5060
bindaddr=0.0.0.0
srvlookup=yes
maxexpirey=3600
defaultexpirey=120
musicclass=default
allow=all
language=en
relaxdtmf=yes
rtptimeout=60
rtpholdtimeout=300
;trustrpid = no
;progressinband=no
useragent=Asterisk
nat=no
externip=XXX.XXX.XXX.XXX
localnet=192.168.2.0/255.255.255.0
promiscredir = no
register = 7304502:[EMAIL PROTECTED]/7304502
register = 31557110304:[EMAIL PROTECTED]/557110304
register = mvanbaak:[EMAIL PROTECTED]
[7304502]
type=friend
context=from-sipgate
host=sipgate.de
username=7304502
secret=my_sipgate_pass
nat=yes
canreinvite=no
insecure=very
[31557110304]
type=friend
context=from-budgetphone
host=sip.budgetphone.nl
username=31557110304
secret=my_budgetphone_pass
qualify=yes
nat=yes
canreinvite=no
insecure=very
[nikotel]
secret=my_nikotel_pass
username=mvanbaak
fromuser=mvanbaak
type=peer
context=from-nikotel
host=calamar0.nikotel.com
canreinvite=no
nat=yes
...some more entries for sip phones/softphones follow, they
all work...
the dial statement in my extensions.conf
[outgoing-budgetphone]
exten = _0X,1,SetAccount(outgoing-budgetphone)
exten = _0X,2,SetCallerID(31557110304)
exten = _0X,3,Dial(SIP/31557110304/${EXTEN})
exten = _0X,4,Congestion
exten = _0X,104,Busy
And this is wat I get on the CLI when I call my cellphone:
-- Executing SetAccount(SIP/michiel-d5bd, outgoing-budgetphone) in new
stack
-- Executing SetCallerID(SIP/michiel-d5bd, 31557110304) in new stack
-- Executing Dial(SIP/michiel-d5bd, SIP/31557110304/06X) in new
stack
-- Called 31557110304/06X
Mar 4 18:51:11 WARNING[4529]: chan_sip.c:6830 handle_response: Forbidden -
wrong password on authentication for INVITE to '31557110304 sip:[EMAIL
PROTECTED];tag=as0ccbacfe'
-- SIP/31557110304-5857 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Busy(SIP/michiel-d5bd, ) in new stack
== Spawn extension (internal, 06X, 104) exited non-zero on
'SIP/michiel-d5bd'
-- Got SIP response 483 Too many hops back from 81.23.228.150
I tripple checked my password, and I am sure it is correct.
What to do ?
--
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Two of the most famous products of Berkeley are LSD and BSD. I don't think
that this is a coincidence.
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Re: [OT] - [Asterisk-Users] Why should I answer a Newbie question, therethick!
- Original Message - From: David Brodbeck [EMAIL PROTECTED] Sure. So say, I tried a Googling for X, but I didn't have any luck. Then I looked at pages X and Y in the Wiki, but couldn't find anything that related to my problem. People are a lot more sympathetic if you demonstrate you've made some effort to find the answer on your own. True, but sometimes a newbie doesn't know that people are looking for this, they're new to how lists work as well. So why not answer the question, nicely, and then say 'BTW, some people will be more symathetic if you research... yada yada...' The key thing being the term 'nicely'. Some people don't realize just how agressive their blunt approach can come across to a newbie... Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] - [Asterisk-Users] Why should I answer a Newbie question,therethick!
- Original Message - From: David Brodbeck [EMAIL PROTECTED] Well, sometimes that works. But I've been on a lot of lists where newbies who thought they were being ignored started flaming people for not responding to them, writing posts badmouthing the project, hijacking other threads, accusing people of being cliquish, etc. Sometimes you just can't win. Sure. But if they flame, then they deserve to get hammered on. In that case, hammer away. However, I don't think it's fair to lump all newbies into that basket. Let them throw the first punch, rather than assume that all newbies who don't know better are out to wreak havoc How would the experts like it if everyone assumes they're a bunch of arrogant techies who only want to talk down to those less worthy, before they speak up and prove it to be true? :) Don't make the same assumptions in the reverse for the newbies... regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Options in Brazil
If you don't speak Portuguese, visit: http://translate.google.com/translate?u=http%3A%2F%2Fwww.asteriskbrasil.org%2Flangpair=pt%7Cenhl=enie=UTF-8oe=UTF-8prev=%2Flanguage_tools On Fri, 4 Mar 2005 14:28:41 -0300, Denis Galvão - iSolve [EMAIL PROTECTED] wrote: If you speck portuguese, visit AsteriskBrasil.org: http://www.asteriskbrasil.org Regards. Denis. Em Qui 03 Mar 2005 22:23, Paul Davidson escreveu: All- I am considering an Asterisk implementation in Brazil. Unfortunately, this presents something of a challenge to plan sitting in Chicago, USA. I know there is a large section of Brazillian Asterisk users who actively read this list- so I'd love to pump out a few questions- note, I'm not necessarily a newbie, having successfully implemented a few Asterisk boxes here in the US. My primary question revolves around connection hardware- I need to plug in 8 POTS lines (I've no idea what they'd be called there) to an Asterisk box. Is digium's TDM400 series availble down there? Recommended? Undesirable? ATA's? (Sipura, presumably) - channel banks? If anyone has any solid knowledge they can share- gotchas appreciated- feel free to contact me off list. Thanks, -pbd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1953 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
--On Friday, March 04, 2005 11:58 AM -0600 James Taylor [EMAIL PROTECTED] wrote: It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. There is clearly an issue between LiveVoip and Asterisk. The LiveVoip people claim that they have been ignored on the Asterisk List and they indeed blame Asterisk for everything from lost dtmf to other failures. That said, they are the only company I've found that offers inbound DIDs with multiple simultaneous calls, suitable for a call center or calling card application. Most others limit you to one, or a small few, inbound paths. They (Level 3, actually) also have the widest coverage for DIDs in the US. At the current level of service, LiveVoip is not going to get my business. If I can find anybody else to provide my inbound service, I'm very interested in talking to them. /edg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] defold usernames in asterisk@home version 6
web address book is user:admin pass:password --- Wiley Siler [EMAIL PROTECTED] wrote: OK. So check out the Wiki here http://www.voip-info.org/tiki-index.php?page=Asterisk The archive of this list can be search via google by entering... site:lists.digium.com some parameter www.digium.com has a link to all the materials for getting started in the Documentation section of the website. Those are really quite good so I would start there. Most were written prior to [EMAIL PROTECTED] being released so they document the real nuts and bolts of how to build a system from ground up. [EMAIL PROTECTED] is really quite turnkey so you don't learn as much. As for you specific question of usernames and passwords Linux Command Line: root and whatever password you set AMP GUI - maint and password (the password is password) Web address book - No idea Web Meet Me - maint and password (can be reset via help-aah) Web Voicemail - ext. # plus the PIN you setup Hope this helps. Please read up at the above sites and youw ill do fine. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Satchid Sent: Friday, March 04, 2005 1:21 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] defold usernames in [EMAIL PROTECTED] version 6 Dear Users, It is not my intention to install a working asterisk for real work. The intention is to learn about it just by playing aroung with it for a wile. I will have a good professional installing an * in my firm. The best way to find out what is possible with the * is to play with it for a wile. So a few starting points could help me a lot. From there I can learn more. I downloaded the iso. Installed it, and then opened the romote GUI on an other computer. I can not open the different parts becouse it asks the user name and the password. The passwords I have changed, that is not a problem at all, but I do not find the usernames documented somewhere. They are installed from the cd that is made of the above mentioned iso. I used the commands from help-aah and they work well. So, a little help here might be in place till I am started, then I only want to learn to configure the different config files, that's all. Thank you, Do not be angry at me! Willy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Friday, March 04, 2005 12:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] defold passwords in [EMAIL PROTECTED] version 6 Well, actually I guess it is help-aah not aah-help but I think you saw the response by mmiranda which will probably cover the sentiments of most here. He is correct to that if you are a linux noob, you need to go get a book and figure out the basics of linux before you embark on this project. Otherwise, you are just in for a world of pain as you try and work on a OS that you have no clue about and which is VASTLY different from Windows. This is assuming you even have the level of skill to teach yourself about Linux which hopefully you do. Dig deep, read long, and google you tail off. Then when you come back for help (which you will, we all do) at least you will be ready for what the gurus here (myself only being a humble jr. * user) can offer in assistance. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, March 03, 2005 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] defold passwords in [EMAIL PROTECTED] version 6 This is probably the worst forum to beg for help or be a noob. There is a strong sentiment that we should each do as much as possible to help ourselves before we come to the community for assistance. Not trying to be mean but you should just know that about this list. In your case, I am wondering where you downloaded your copy of [EMAIL PROTECTED] I got mine here and instructions for basic settings are there on the site. http://asteriskathome.sourceforge.net Password change options are available by typeing aah-help at the linux command prompt. You should actually see an announcement saying this if you log into the linux box with your root login and password. If you are looking for Asterisk docs, go here... http://www.voip-info.org/tiki-index.php?page=Asterisk If you are looking for docs on AMP then google Asterisk Management Portal. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Satchid Sent: Thursday, March 03, 2005 3:57 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] defold passwords in [EMAIL PROTECTED] version 6 Dear Users, I am begging
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
On Fri, 04 Mar 2005 10:12:05 -0800 Ed Greenberg [EMAIL PROTECTED] wrote: --On Friday, March 04, 2005 11:58 AM -0600 James Taylor [EMAIL PROTECTED] wrote: It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. There is clearly an issue between LiveVoip and Asterisk. The LiveVoip people claim that they have been ignored on the Asterisk List and they indeed blame Asterisk for everything from lost dtmf to other failures. That said, they are the only company I've found that offers inbound DIDs with multiple simultaneous calls, suitable for a call center or calling card application. Most others limit you to one, or a small few, inbound paths. They (Level 3, actually) also have the widest coverage for DIDs in the US. At the current level of service, LiveVoip is not going to get my business. If I can find anybody else to provide my inbound service, I'm very interested in talking to them. /edg Seems kind of starnge that they are the only ones having this problem. I am pulling an account from Voicepulse using IAX and not have a problem at all. Maybe they need to call Digium, or some other contractor, and pay someone to set it up for them correctly since it is obviously they cannot accomplish this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
Seems kind of starnge that they are the only ones having this problem. I am pulling an account from Voicepulse using IAX and not have a problem at all. Maybe they need to call Digium, or some other contractor, and pay someone to set it up for them correctly since it is obviously they cannot accomplish this. I had some issues with VoicePulse as well with IAX. Don't remember exactly what they were... but I believe it may had been an IAX trunking issue. -forrest ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ANNOUNCEMENT : Asterisk-Stat V2.0 - CDR Analyser
Dear ALL, As everybody seems to like very much Asterisk-Stat, I decided to make couples of improvements... so here we go with a new version :D FEATURES : - CDR report (monthly or daily) - monthly traffic reports (pie graph) - DAILY LOAD !!! - compare call load with previous days - many criterias to define the report - export CDR report to PDF - export CDR report to CSV - support MYSQL POSTGRESQL - etc... Better to check out the screenshot: http://areski.net/asterisk-stat-v2/about.php Waiting for your feedbacks! Enjoy and have a good weekend, Areski -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_ Belad Arezqui Web:http://areski.net/ Email: areski ($alt) gmail ($dot) com -_-_-_-_-_-_-_-_-_-_-_-_-_-_ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
Ok, time for me to ask my own newbie question. :) I've done some digging on ringback, and if I'm understanding it correctly, it's the ring tone that the caller hears when dialing another person. What exactly is it that people are finding now working with LiveVoip? Everyone says 'ringback isn't working', but nobody's really explained exactly what's happening. At least not that I've been able to find. I have a DID with them, and it works just fine. Dialing out works fine, when people call in it works fine. I'm interested in knowing what it is that isn't working, and if I can re-create it on my system... regards, Paul - Original Message - From: Ed Greenberg [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 11:12 AM Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip --On Friday, March 04, 2005 11:58 AM -0600 James Taylor [EMAIL PROTECTED] wrote: It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. There is clearly an issue between LiveVoip and Asterisk. The LiveVoip people claim that they have been ignored on the Asterisk List and they indeed blame Asterisk for everything from lost dtmf to other failures. That said, they are the only company I've found that offers inbound DIDs with multiple simultaneous calls, suitable for a call center or calling card application. Most others limit you to one, or a small few, inbound paths. They (Level 3, actually) also have the widest coverage for DIDs in the US. At the current level of service, LiveVoip is not going to get my business. If I can find anybody else to provide my inbound service, I'm very interested in talking to them. /edg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [OT] - [Asterisk-Users] Why should I answer a Newbie question,therethick!
Jeff Busch wrote and I modified: *** Asterisk is a Open Source community and supported by volunteers. Please do the following before asking one of these volunteers for help. 1. Before asking a question, do a Google search 2. After a general Google search, do a specific search on this group 3. After a Google search, look at http://www.voip-info.org/wiki-Asterisk the information contained in these pages will answer 95% of your startup questions. 4. If you have done 1, 2, 3 - feel free to email the list. 5. Please do not email the list asking people to hold your hand. That is not what the list is for, it is for help if you run into an implementation problem, not to teach you the basics by using 1, 2, 3. In addition: Output from the Asterisk console with -vvv and sip debug turned on is VERY helpful to diagnose your errors. * I like this idea if it is possible! I am trying to get my sales department to do something like this as well. I could write a hundred FAQs, but nobody reads them, just calls:) Shanon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
On Fri, 04 Mar 2005 11:35:55 -0700 Paul Fielding [EMAIL PROTECTED] wrote: Ok, time for me to ask my own newbie question. :) I've done some digging on ringback, and if I'm understanding it correctly, it's the ring tone that the caller hears when dialing another person. What exactly is it that people are finding now working with LiveVoip? Everyone says 'ringback isn't working', but nobody's really explained exactly what's happening. At least not that I've been able to find. I have a DID with them, and it works just fine. Dialing out works fine, when people call in it works fine. I'm interested in knowing what it is that isn't working, and if I can re-create it on my system... regards, Paul Setup your * box to not answer the call right away. Allow for say 5 seconds of ringing. Then call into it on one of your DID's. From the calling end all you will get is dead air. No ringing. At least this is the issue I am having.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
Ed Greenberg wrote: --On Friday, March 04, 2005 11:58 AM -0600 James Taylor [EMAIL PROTECTED] wrote: It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. There is clearly an issue between LiveVoip and Asterisk. The LiveVoip people claim that they have been ignored on the Asterisk List and they indeed blame Asterisk for everything from lost dtmf to other failures. That said, they are the only company I've found that offers inbound DIDs with multiple simultaneous calls, suitable for a call center or calling card application. Most others limit you to one, or a small few, inbound paths. They (Level 3, actually) also have the widest coverage for DIDs in the US. At the current level of service, LiveVoip is not going to get my business. If I can find anybody else to provide my inbound service, I'm very interested in talking to them. /edg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I believe LiveVOIP is a reseller of Level 3. From what I understand, you need to buy millions of minutes to get decent pricing at Level 3 as they are a mega wholesaler... I may be wrong, but that's what I got out of it. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bluetooth phone as SIP handset?
I know people are working on using a bluetooth phone as an extra line to send a receive calls through asterisk, but is anyone working on using a bluetooth phone as a handset - i.e. using it to dial calls and talk though asterisk? I would easily give upto $100 as a boounty for this functionality and I'm sure many others would too, as it would mean people wouldn't have to buy a hardware SIP phone or an ATA. Anyone know if this is possible? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [OT] - [Asterisk-Users] Why should I answer a Newbie questio n,therethick!
1. Before asking a question, do a Google search 2. After a general Google search, do a specific search on this group 3. After a Google search, look at http://www.voip-info.org/wiki-Asterisk the information contained in these pages will answer 95% of your startup questions. 4. If you have done 1, 2, 3 - feel free to email the list. 5. Please do not email the list asking people to hold your hand. That is not what the list is for, it is for help if you run into an implementation problem, not to teach you the basics by using 1, 2, 3. In addition: Output from the Asterisk console with -vvv and sip debug turned on is VERY helpful to diagnose your errors. * I like this idea if it is possible! I am trying to get my sales department to do something like this as well. I could write a hundred FAQs, but nobody reads them, just calls:) Shanon A Google search of the lists using terms site:digium.com in Google is also very helpful in finding pertinent material. -Nate ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
On Fri, 04 Mar 2005 11:46:27 -0700 Paul Fielding [EMAIL PROTECTED] wrote: Hmmm. My server is currently set to let the line ring for 20 seconds, ringing several extensions internally. (I do not answer the line, it just rings the extensions). If I don't pick up after 20 seconds it then answers the line and sends to voicemail or to an auto-attendant, depending on the situation. Ringback seems to be working for me, I hear ringing on the calling end... *shrug*. Paul Ok,I have to retract my last statement and give an update. It has been a while since I had played with the DID I have from them. It is not an issue before the * box picks up. I set my incoming context to ring my VoIP phone for 20 seconds directly with using the IVR system and I had the ringing. But when I restored it to no background on hold music and issued a dial command of Dial(SIP/2001,15,r) instead of Dial(SIP/2001,15,m), after the IVR plays its intro, I got no ringing on the calling end. Just dead air from LiveVoIP. I then used this same test context by dialing in through a VP Connect account and after the initial greeting and moving to the Dial command, I got the ringing on the the calling end. Sorry for the incorrect info the first time, it had just been quite a while since I had played with the Live account. Robert - Original Message - From: Robert Webb [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Friday, March 04, 2005 11:42 AM Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip On Fri, 04 Mar 2005 11:35:55 -0700 Paul Fielding [EMAIL PROTECTED] wrote: Ok, time for me to ask my own newbie question. :) I've done some digging on ringback, and if I'm understanding it correctly, it's the ring tone that the caller hears when dialing another person. What exactly is it that people are finding now working with LiveVoip? Everyone says 'ringback isn't working', but nobody's really explained exactly what's happening. At least not that I've been able to find. I have a DID with them, and it works just fine. Dialing out works fine, when people call in it works fine. I'm interested in knowing what it is that isn't working, and if I can re-create it on my system... regards, Paul Setup your * box to not answer the call right away. Allow for say 5 seconds of ringing. Then call into it on one of your DID's. From the calling end all you will get is dead air. No ringing. At least this is the issue I am having.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bluetooth phone as SIP handset?
Chris, I will take your $100.00 bounty :-D I am using a bluetooth headset with firefly and my laptop right now. Their softphone works well with asterisk. All you have to do is pair the headset to your computer, and set in the options to use the bluetooth. Mine works well. -Linn Chris Birkinshaw wrote: I know people are working on using a bluetooth phone as an extra line to send a receive calls through asterisk, but is anyone working on using a bluetooth phone as a handset - i.e. using it to dial calls and talk though asterisk? I would easily give upto $100 as a boounty for this functionality and I'm sure many others would too, as it would mean people wouldn't have to buy a hardware SIP phone or an ATA. Anyone know if this is possible? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bluetooth phone as SIP handset?
Even better you can set your firefly softphone to auto answer so that you don't even need to be near the pc to answer. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Linn Boyd Sent: Friday, March 04, 2005 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bluetooth phone as SIP handset? Chris, I will take your $100.00 bounty :-D I am using a bluetooth headset with firefly and my laptop right now. Their softphone works well with asterisk. All you have to do is pair the headset to your computer, and set in the options to use the bluetooth. Mine works well. -Linn Chris Birkinshaw wrote: I know people are working on using a bluetooth phone as an extra line to send a receive calls through asterisk, but is anyone working on using a bluetooth phone as a handset - i.e. using it to dial calls and talk though asterisk? I would easily give upto $100 as a boounty for this functionality and I'm sure many others would too, as it would mean people wouldn't have to buy a hardware SIP phone or an ATA. Anyone know if this is possible? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bluetooth phone as SIP handset?
What head set are you using? We have the pro XTen and would like to be able to press a button on the BT device and pickup the call remotly. Just wear the BT on your ear as you walk about the office. You hear your softphone ring in your ear, press a button and Hello. -Matthew - Original Message - From: Linn Boyd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 1:11 PM Subject: Re: [Asterisk-Users] Bluetooth phone as SIP handset? Chris, I will take your $100.00 bounty :-D I am using a bluetooth headset with firefly and my laptop right now. Their softphone works well with asterisk. All you have to do is pair the headset to your computer, and set in the options to use the bluetooth. Mine works well. -Linn Chris Birkinshaw wrote: I know people are working on using a bluetooth phone as an extra line to send a receive calls through asterisk, but is anyone working on using a bluetooth phone as a handset - i.e. using it to dial calls and talk though asterisk? I would easily give upto $100 as a boounty for this functionality and I'm sure many others would too, as it would mean people wouldn't have to buy a hardware SIP phone or an ATA. Anyone know if this is possible? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web based tool asterisk real time
Is there a webbased tool to use with asterisk real time. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sip.c:6848 handle_response: Failed to authenticate on INVITE
Hi I am running Asterisk CVS-v1-0-03/04/05-18:54:35 and I see to get the error stated above. My setup is ser which takes in a call and rewrites to asterisk. ser.cfg : rewriteuri(sip:[EMAIL PROTECTED]:5090); This takes call to asterisk sip.conf [general] autocreatepeer=yes port=5090 defaultexpirey=3600 register = user:[EMAIL PROTECTED]/10 context=sip This creates a peer with the sip server. extensions.conf [sip] exten = 10,1,SetCIDName(Test Line) exten = 10,2,Dial(SIP/[EMAIL PROTECTED]) So from the command line in asterisk I try dial [EMAIL PROTECTED] this sends details to ser, ser picks them up, but asterisk shows this , I have obvioulsy missed something very simple here tks Iqbal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + SIP + NAT - seriously, what's the secret?
Stuart Ford wrote: Seriously, this has to be the simplest NAT problem there is with Asterisk. What's the secret? How do I learn the dark art? What am I missing? I'm guessing here, but the NAT'd grandstream does not have the correct external IP configured. The phones are trying to establish a direct SIP to SIP connection, after SIP to SIP call is established asterisk tries to get out of the middle of the conversation. This decreases latency and save processing on the asterisk box. canreinvite=no sometimes helps this problem when asterisk is a sip client... don't know if it will have an effect here. The thing to do is setup an extension with the Echo Application. Call that from each phone and see what happens. If it works for both phones you know the problem is a reinvite issue, if one phone or the other doesn't work it is a network or Nat config issue. No sense flailing about, try to reduce the problem space. If your familiar with ethereal it can be used to snoop on the SIP connection.. SIP is human readable, so you might be able to learn something interesting. But I really know almost nothing about this. Mark Farver ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP MWI and MySQL Realtime
I know that there are some patches being worked on to cache realtime users that might ultimately fix this problem, but until then, here is a little script that brings back the MWI when using the excellent mysql realtime architecture with sip: http://www.cheapnet.net/~mike/asterisk/send_mwi.txt This script relies on sipsak utility found at http://sipsak.berlios.de/ Download, rename to send_mwi.pl and chmod 755 it. See top of file for notes on usage and configuration. If you have any feedback, let me know. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi patch for the new cvs HEAD
I've patched the chan_capi to let it compile under the new CVS head Give it a try please You have to start from the original chan_capi http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz and then apply the patch http://www.c-net.it/chan_capi.diff.bz2 it also includes the fax patch from Frank Sautter Let me know please Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bluetooth phone as SIP handset?
I have a new HP IpaQ 6315. I run SJPhone on it with a bluetooth headset. Works great! Paul paul mahler www.signate.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Friday, March 04, 2005 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bluetooth phone as SIP handset? What head set are you using? We have the pro XTen and would like to be able to press a button on the BT device and pickup the call remotly. Just wear the BT on your ear as you walk about the office. You hear your softphone ring in your ear, press a button and Hello. -Matthew - Original Message - From: Linn Boyd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 1:11 PM Subject: Re: [Asterisk-Users] Bluetooth phone as SIP handset? Chris, I will take your $100.00 bounty :-D I am using a bluetooth headset with firefly and my laptop right now. Their softphone works well with asterisk. All you have to do is pair the headset to your computer, and set in the options to use the bluetooth. Mine works well. -Linn Chris Birkinshaw wrote: I know people are working on using a bluetooth phone as an extra line to send a receive calls through asterisk, but is anyone working on using a bluetooth phone as a handset - i.e. using it to dial calls and talk though asterisk? I would easily give upto $100 as a boounty for this functionality and I'm sure many others would too, as it would mean people wouldn't have to buy a hardware SIP phone or an ATA. Anyone know if this is possible? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.0 - Release Date: 3/2/2005 = Paul Mahler [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardphone deployment recommendation
I'm looking to purchase and deploy a bunch of hardphones for agent use. The phones will have to register with Asterisk and/or SER, depending on where the phones go. They need only one line, G729 codec, and no super fancy features. Preferrably something that is easy to provision. I would think the BudgeTone would be good, but then I've read so many people complaining about them, and some people seem to recommend the Sipura adapters. I'm looking to keep my cost down, and the BudgeTone is around $100 CDN, give or take. Let me know what you would purchase for about 100 users, 1 line each, G729, and why. We've had decent results with the BudgeTone phones I have already, but I only have about 4 of them, and I have about 5 Aastra/Sayson 480i phones which are a bit pricey for this application, and very featureful. The Sipura box I have is alright, and the IAXys work, but aren't an option for this application. I'm looking for SIP, not IAX. The main reason I ask to the mailing list instead of basing a large purchase decision on the phones I have here is that while these devices haven't failed on me yet (with the exception of one flaky 480i), I know that there are some of you who have experience with large deployments. Also, if you recommend an analog adapter, is there any recommendation for analog phone to go with it? I'm not sure if the users will want headsets or handsets, so either one is fine. Thanks for any advice and experiences. PS: If you're thinking you'll get a purchase contract out of me, you won't - the supplier decision isn't in my hands, so don't bother spamming me with your deals. -- Dana ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hardphone deployment recommendation
I would think the BudgeTone would be good, but then I've read so many people complaining about them, and some people seem to recommend the Sipura adapters. For agent use, the BudgeTone's lack of three-way calling would be an issue. Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk box and verizon calling it
I set up an asterisk box with a broadvoice sip connection for incoming connections it works great when I use a cell phone, vonage line, calling card to call the asterisk box, but when I try to call it from our verizon land line it is busy and asterisk logs do not show incoming call. Any ideas on what the issue is? Thanks! Randy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Placing a call from command line and passing it to an extension if connected - Is it possible?
Is it possible to dial number from the command line and passing the connection to one of my extension (or speakerphone) if the other party answers the call? I was thinking of implementing this sort of feature with and accounting application. The customer phone number is in the database, so clicking and icon asterisk would dial the number and connected to my speakephone when the connection goes through. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice over Frame Relay Asterisk
Has anyone done Voice Over Frame Relay with Asterisk. With Frame Relay work reliably with Asterisk? Any experiences? __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
